Hi,
-Original Message-
i have extensions in locations across a number of telco area codes.
when someone in seattle picks up and dials 91234567, it would
be nice to transform it to 92061234567. i would prefer not
to have an extension context per area code. it would be cool
to be
I'm having the exact same problems here - won't start with safe_asterisk.
I'm running a slightly dated CVS head (CVS-02/24/04-15:39:13) however I have
two machines running this date CVS, the other already has G.729 installed
and works fine - however it registered automatically with the voiceage
MessageHi,
-Original Message-
From: Aaron Martin
Sent: Tuesday, May 18, 2004 8:55 PM
Does anyone have any recomendations for a free Windows softphone, SIP or
IAX that supports the following features:
* Message Waiting Indicator
* Consultative Transfers
* Speed Dials
DIAX can do
Hi,
-Original Message-
- Sometimes a call won't go through, dialtone stays on after
keypresses.
Strange. A powercycle of the iaxy usually helps
Are you changing which phone is plugged into the iaxy before
this happens? The changing of resistance levels I have
noticed to cause
Hello Vit,
just try the indications from the UK. That worked fine in Germany.
Bye
Felix
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dudlik
Sent: Monday, May 17, 2004 9:20 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] indications.conf
FYI:
insmod does not resolve module dependencies...
modprobe does...
In this case zaptel isn't loaded yet.
So... copy qozap.(o|ko) to /lib/modules/`uname -r`/misc/
redo the module dependencies..
modprobe qozap
or
make load in the qozap subdir in bristuff...
(Beware this executes ztcfg with the
At 1:12 PM +0100 on 5/18/04, Tony Hoyle wrote:
Manuel Wenger wrote:
Hi Tony,
Try adding fromuser=x, maybe username= isn't enough...
Just a guess, it already solved a few problems for me.
I've tried fromuser=, username= and some fromdomain= combinations -
unfortunately I'm not 100% sure
In article [EMAIL PROTECTED],
Dave Packham [EMAIL PROTECTED] wrote:
has anyone done caller announce in MeetMe's before?
I'm working on some modifications that should make this possible,
amongst other things.
However, I have some questions about specific details of the operation
of app_meetme,
At 5:58 PM -0400 5/18/04, James H. Cloos Jr. wrote:
Randy == Randy Bush [EMAIL PROTECTED] writes:
Randy how does a call to sip://[EMAIL PROTECTED] come in to asterisk so i
Randy can route it?
I beleive it comes in to extension user in the default context:
[default]
exten = user,1,whatever...
At 12:10 AM -0600 on 5/19/04, brian k. west wrote:
Just as a better example of how to use accountcode for this ... maybe we
could add areacode= to sip.conf and pull that into a var instead of wasting
the accountcode for this hrm...
[1234567]
username=whatever
secret=whatever
context=phones
John Todd wrote:
For what it's worth, I haven't been able to make the realm= setting
do diddly-squat. I think it's broken, but I don't have time to test
enough to put useful/valid data into the bugtracker.
For me, it's changing the realm in the auth header. Need to re-check later
with the
Yes. Give away your LJs to some university for research...
They are not for business... and don't buy X100P. Buy TDM400P.
It has the same price of a LJ and have 4 FXOs instead of only one.
Isamar
On Wed, 19 May 2004, Jer wrote:
Dear all
I read on the list back in 2003 that * does not
I want use iLBC and have following in mind, please help me is it possible ?
ISDN -(ALAW)- * -(ALAW)- SNOM
SIP -(iLBC)- * -(ALAW)- SNOM
1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC
because a lack of codec).
2. SIP incoming codec should
This is what i got ... and i could not find the problem by my own
/sbin/depmod -a
depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/qozap.o
depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/tor2.o
depmod: *** Unresolved symbols in
I've been lurking on the list for quite a while now, and I hadn't seen
much from people using FreeBSD, so I figured I'd post something.
Yes, the driver in the ports tree for the digium card works!
I'd heard reports that it crashes the box/etc, but I have yet to encounter
that.
I've had all the
put iLBC at the top of your allowed list of codecs in the general section of
sip.conf.
- Original Message -
From: nicolas [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, May 19, 2004 9:47 AM
Subject: [Asterisk-Users] using iLBC
I want use iLBC and have following in mind,
That is the way it works over one zap channel, to keep * in the call it
would need to dial out on anoher line and that would then use an additional
zap interface and tie it up for the duration of the call.
Jason
At 20:30 18/05/2004 -0300, you wrote:
Yes, I've tried with SendDTMF, and it works,
Stefan Tichy wrote:
registration.
Excuse me, but where do You find the "Subscribe for MWI" flag? What
firmware version?
"Subscribe for MWI" flag in the GS config page, and it stopped losing
1.0.4.55
No, do not send SUBSCRIBE for Message Waiting
Hi!
Can you use wildcards on the caller id? Eg certain area codes get a
certain ring type others get a different ring type?
Wildcards don't appear to be working (tried only * and ?), but partial
leading matches do. So 12 for ringtone 1 will match 1234 - unless you
have a more specific
Hi there,
I have an asterisk server with cvs-code from May 13,2004 and a Quad-PRI
Card.
1 Port of the Quad-Pri is connected to the Telekom-PSTN, the other
Port is connected to an Alcatel PBX.
What I want is to make Asterisk Bridge the Calls from PBX to PSTN
and to add some VoIP Functionality
You know, I'm not so sure this is limited to chan_capi. I have two
asterisk boxes running, with one connected to my PSTN gateway (also
using Asterisk). 1.0 stable works fine with my Cisco phone. CVS head
works if I comment out the offending lines. Without commenting them out,
the cisco
Hello!
Does anybody know of a way to access the Request URI in a SIP message?
I've got the following problem/scenario:
We have a SIP Proxy (SER) wich forwards SIP-messages for non-IP
destinations to our Asterisk. There is no authentication done between
Asterisk and SER. I've configured Asterisk
Have you installed the zaptel module (zaptel.o) in your modules ?
Try an :
insmod zaptel
If it does not work, it means it has not been installed.
- Original Message -
From: Michael Devenijn [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, May 19, 2004 10:45 AM
Subject:
yes i installed it ...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Frederic
Olivie
Sent: woensdag 19 mei 2004 12:44
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problem compiling zaptel with BRIstuff
0.0.2
Have you installed the zaptel module
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 19 May 2004 03:45, Vic Cross wrote:
Where are you ethereal traces so I can look over them.
I appreciate that.
http://veejoe.com.au/isdnbadcall.gz
http://veejoe.com.au/isdnbadcall2.gz
I took a look at those as well and it looks a lot
Do an :
insmod qozap
You'll have the list of unresolved symbols.
You can post a subset here, it will help determine what's missing.
- Original Message -
From: Michael Devenijn [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, May 19, 2004 12:56 PM
Subject: RE: [Asterisk-Users]
brian k. west wrote:
You know, I'm not so sure this is limited to chan_capi. I have two
asterisk boxes running, with one connected to my PSTN gateway (also
using Asterisk). 1.0 stable works fine with my Cisco phone. CVS head
works if I comment out the offending lines. Without commenting them out,
Christoph Adomeit wrote:
May 14 09:16:21 VERBOSE[1180010432]: -- Extension '9149' in context
'dtagpri' from '02166458729' does not exist. Rejecting call on channel
10, span 1
I am sure all these callers called more Numbers than 9149, they
might have
called 9149-0 or 9149-xx but not the 9149
Are there any variables or structure elements unique to a call that stay
till the end of a call -- including when caller enters a queue and then
bridged with agent. I am trying to get some variables about the caller in
an AGI script when the agent's phone is ringing, and I'm finding not even
the
this is an old problem.
there's a kernel patch for that.
search the mailing lists archives.
Matteo.
Il mer, 2004-05-19 alle 13:42, Michael Ljtnant ha scritto:
Hi,
I have a problem that I just can't figure out how to solve.
I start *, dial it using a ISDN phone over PSTM, to a Hisax card
Hi,
I have a Junghanns.net quadBRI PCI Card with Telefonica ISDN BRI line, and
we have in zapata.conf usecallerid=yes
and hidecallerid=no, but we have not the caller ID.
Can I make some configuration to solve this?
Thanks,
Pedro
___
Asterisk-Users
A bit about my experience with the TDM-04 FXO. Only saw a few post on
this subject, thought I would contribute a little about my experience to
save others the hassle.
and I might add...
a. if any analog phones bridged on the pstn line go off-hook then on-hook,
the tdm card senses it and
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lars Boegild
Thomsen
Sent: Tuesday, May 18, 2004 11:23 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] * and Cisco routers
Well - I would assume that most Asterisk instances run on Linux boxes, so
any one have succeed to run the ./register binary. Im having problem with
glibc.
[EMAIL PROTECTED] asterisk]# ./register my-license
./register: /lib/libc.so.6: version `GLIBC_2.3' not found (required by
./register)
[EMAIL PROTECTED] asterisk]#
I would be so pleasure if any one can help me!
Hi Alf,
Have you got a Junghanns.net quadBRI PCI Card ?
If yes, Have you received CallerID number ? How you have got configured
zaptel and zapata ?
Im collapssed at this point, thaks in advance,
Pedro
PD: maybe... around your question, are you using cdr in csv or in mysql?, if
you are using
John Vogel wrote:
I upgraded to the latest stable version of 1.0 today and am still
seeing the *8 problem where the phone that was originally dialed keeps
on ringing even after another phone picks up.
Are other people also seeing this? Has somebody figured out how to
make this go away?
Hi Pedro,
please do a bri debug span X on the span with the BRI line. Look at
the SETUP message, the incoming caller ID is in the Calling Party IE.
If you do not see the caller ID in that IE then your telco is not
providing incoming caller ID on your line, some telcos like to charge
extra for
Hi,
I've gotten hold of a few SIP videophones and would love to be able to
transfer the video-feed from RTP through IAX2 which is used between two
asterisk boxes:
Videophone (SIP) - * (IAX2) - * (SIP) - Videophone
I did get nice video between the two phones when they were registered to the
same
From what I understand, the Stable code has not yet been fixed for the
iax problem and that most certainly was a major bug. Fair number of
folks seem to using stable code as well.
I'm going to state this again for those not paying close attention.
CVS-HEAD is just as stable if NOT MORE stable
From what I understand, the Stable code has not yet been fixed for the
iax problem and that most certainly was a major bug. Fair number of
folks seem to using stable code as well.
I'm going to state this again for those not paying close attention.
CVS-HEAD is just as stable if NOT MORE
http://www.eweek.com/article2/0,1759,1592801,00.asp
Seems Ellen has done such a fine job smacking Jim Louderback up side the
head for us! :) Thought you might get a kick out of this one.
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
You're missing the point, Brian. Those comments were in response to your
statement that essentially said there isn't a problem because your system
is working fine. And based on your comment, your primary (only?) iax link
is to Nufone.
No I'm getting it loud and clear. You have some IAX
We as users of lots of other service providers and systems other then
Nufone don't have the choice of forcing those systems to either Head
or Stable. That's purely irrelevant. Since we can't force others to
upgrade to anything, we're stuck with either throwing away the Cisco
phones or
I pumped up the verbosity to freetds and found this:
pid: 30035:Received packet @ 11:02:14.897325
aa 82 00 e9 03 00 00 01-0f 37 00 4c 00 69 00 6e | .7.L.i.n|
0010 00 65 00 20 00 31 00 3a-00 20 00 4c 00 65 00 6e |.e. .1.: . .L.e.n|
0020 00 67 00 74 00 68 00 20-00 6f 00 72 00 20 00 70
In our last exciting episode, Tom (UnitedLayer) ([EMAIL PROTECTED]) said:
Yes, the driver in the ports tree for the digium card works!
I'd heard reports that it crashes the box/etc, but I have yet to encounter
that.
What is the name of the port? I don't see it and I refresh my ports tree
on my
Hekuran Doli wrote:
any one have succeed to run the ./register binary. Im having problem with
glibc.
[EMAIL PROTECTED] asterisk]# ./register my-license
./register: /lib/libc.so.6: version `GLIBC_2.3' not found (required by
./register)
[EMAIL PROTECTED] asterisk]#
I would be so pleasure if any one
The fact is the provider is running broken code. They should fix it.
That's the true bottom line.
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
brian wrote:
You're missing the point, Brian. Those comments were in response to your
statement that essentially said there isn't a problem because your system
is working fine. And based on your comment, your primary (only?) iax link
is to Nufone.
No I'm getting it loud and clear. You have
How can someone try the new beta G.729 codec with the key of the
Voiceage's codec? I tried to execute the registration program
but I get a 'XXX...' is not a valid key! message.
Michael.
Hekuran Doli wrote:
any one have succeed to run the ./register binary. Im having problem with
glibc.
[EMAIL
Quoting brian [EMAIL PROTECTED]:
You're missing the point, Brian. Those comments were in response to your
statement that essentially said there isn't a problem because your system
is working fine. And based on your comment, your primary (only?) iax link
is to Nufone.
No I'm getting it
It works fine here.
Proxy-Authorization: Digest
username=10,realm=bkw.org,uri=sip:x.x.x.x,response=xxx,nonce
=4a9f244e,algorithm=md5
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Olle E. Johansson
Sent: Wednesday, May
Luis Vazquez [EMAIL PROTECTED] wrote:
I upgraded to the latest stable version of 1.0 today and am still
seeing the *8 problem where the phone that was originally dialed keeps
on ringing even after another phone picks up.
Are other people also seeing this? Has somebody figured out how to
We as users of lots of other service providers and systems other then
Nufone don't have the choice of forcing those systems to either Head
or Stable. That's purely irrelevant. Since we can't force others to
upgrade to anything, we're stuck with either throwing away the Cisco
phones or
Peter Corlett [EMAIL PROTECTED] wrote:
The pricing makes rather a mockery of most VoIP providers! :)
This isn't entirely difficult, unfortunately.
VoIP isn't really about price any more; it's more about flexibility,
and choice. If you use OneTel, you can make free calls anywhere in
the UK
The fact is the provider is running broken code. They should fix it.
That's the true bottom line.
Agreed but Rich needs a workaround. I think what I suggested will work and is
cheap -- a spare PC with a pair of ethernet cards in it, and a second
ethernet card for his existing * server.
-A.
/usr/ports/misc/zaptel
--
Arnold Cavazos, Jr. abcjr at abcjr . net
On Wed, May 19, 2004 at 10:06:00AM -0500, Jason T. Nelson wrote:
In our last exciting episode, Tom (UnitedLayer) ([EMAIL PROTECTED]) said:
Yes, the driver in the ports tree for the digium card works!
I'd heard
port misc/zaptel
In our last exciting episode, Tom (UnitedLayer) ([EMAIL PROTECTED]) said:
Yes, the driver in the ports tree for the digium card works!
I'd heard reports that it crashes the box/etc, but I have yet to encounter
that.
What is the name of the port? I don't see it and I
this email was for the digium support becouse they compiled the codec with
glib2.3, and what should I do now If Im using redhat7.3 for years and It
was so stable while redhat7.3 does not support glibc2.3?
Best Regards
Hekuran Doli
Hekuran Doli wrote:
any one have succeed to run the ./register
I commented the two lines in rtp.c, took 20 seconds, rebuilt the source, and
off we go. Fixes the iax issue just fine.
I think the rtp.c 'hack' is the way to go. Bad 'ole 'hack'.
Just my 2c.
- Chris
Netlabz, Inc.
- Original Message -
From: Ray Burkholder [EMAIL PROTECTED]
To: [EMAIL
I have contact the digium support and they said that it should work!
Best Regards
Hekuran Doli
How can someone try the new beta G.729 codec with the key of the
Voiceage's codec? I tried to execute the registration program
but I get a 'XXX...' is not a valid key! message.
Michael.
Hekuran
Brian Cuthie [EMAIL PROTECTED] wrote:
Also on a side note if Kapejod isn't wanting keep chan_capi up to date
then someone needs to ask him if he will disclaim it so digium can
include it and help maintain it.
As I understand it, chan_capi has been released under the GPL. That
being the
Shaun Ewing wrote:
I'm not seeing this - using stable CVS from 14-05-2004.
Phone types are Grandstream Budgetone 101 using firmware 1.0.4.68, and Cisco
7940 using SIP 6.2.
-Shaun
Just to give more info.
I just made a testing using stable CVS from 24-04-2004 and 3 softphone
clients registered
This is great! Go Ellen!
brian
http://www.eweek.com/article2/0,1759,1592801,00.asp
Seems Ellen has done such a fine job smacking Jim Louderback up side the
head for us! :) Thought you might get a kick out of this one.
bkw
___
Two * boxes does not fix the problem as the timestamps will trickle through
each. The root of the problem is two fold: a) iax conversations where pkts
originate from an * system that is either older then about 30 days ago (or
a system based on stable code), and, b) cisco 7960 issue with
On Wed, May 19, 2004 at 11:16:39AM +0200, Tomas Prybil wrote:
Please use plain text instead of text/html.
Excuse me, but where do You find the Subscribe for MWI flag? What
firmware version?
It is 1.0.4.55
Did You find anything out with the tests?
Swiching it on did not cause problems
have you tried the #asterisk-dev IRC room? thats the best place
Dave P
[EMAIL PROTECTED] 5/19/2004 2:12:10 AM
In article [EMAIL PROTECTED],
Dave Packham [EMAIL PROTECTED] wrote:
has anyone done caller announce in MeetMe's before?
I'm working on some modifications that should make this
You can't consider CVS_HEAD dev anymore as its been allowed to stabilize in
the past month and will be 1.1
Bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, May 19, 2004 11:09 AM
To: [EMAIL PROTECTED]
What's wrong with compiling glibc2.3 and installing it on your box? That's
how things have been done for years... before those evil things called RPM's
came along.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Hekuran Doli
On Wed, 2004-05-19 at 10:12, Brian Cuthie wrote:
Asterisk is currently a rapidly moving target, as this very issue
demonstrates. Once a 1.0 or 1.1 is released you can bet everyone will
upgrade. Until then, all of us should probably keep our expectations in
check.
Actually CVS -head is the
Yes they MUST disclaim the code as digium has a dual lic. so digium must
have permission to add it to CVS that is why no GPL code can touch asterisk.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin Walsh
Sent: Wednesday, May
On Wed, 2004-05-19 at 10:38, Kevin Walsh wrote:
As I understand it, chan_capi has been released under the GPL. That
being the case, the author doesn't need to sign over his copyright
or release it as no-license public domain code, and the Asterisk
maintainers are free to include it in the CVS
Thanks to all for the comments even if they don't agree!
I think this issue is significant and I would really like it to be fixed in
the 1.0 release.
Does anybody know how to get the same functionality without using *8?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
JT,
I've not yet tested (played) with setting the realm yet, but I plan
to. However, one would think that realm should be a per-entry setting
versus a global setting, or perhaps both.
The problem that really arises is when you try to use asterisk as a local
UA proxy. I don't mean proxy in
Kevin Walsh wrote:
Brian Cuthie [EMAIL PROTECTED] wrote:
Also on a side note if Kapejod isn't wanting keep chan_capi up to date
then someone needs to ask him if he will disclaim it so digium can
include it and help maintain it.
I'm almost certain I didn't say this. Please be careful with
The fact is the provider is running broken code. They should fix it.
That's the true bottom line.
Agreed but Rich needs a workaround. I think what I suggested will work and is
cheap -- a spare PC with a pair of ethernet cards in it, and a second
ethernet card for his existing *
Hi,
When I installed my X100P card there was a two ring
delaybefore the phone started to ring.
I managed to solve that by stopping the Caller Line
ID being passed to the phone. I'm in the UK so that doesn't matter
anyway.
The problem that I have now is that when a call
comes in on the
What's the story on the enhanced voicemail features ? When will they be
committed ? They don't appear to be in CVS-HEAD-05/19/04.
Thanks,
Chris
Netlabz, Inc.
- Original Message -
From: brian [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, May 19, 2004 12:04 PM
Subject: RE:
It's in /usr/ports/misc/zaptel. For everyone else, there were some updates
to the driver recently by Maxim Sobolev (the maintainer) that can be
obtained by cvsup (see /usr/share/examples/cvsup/ports-supfile).
It works great with a single x100p card here as well!
Cheers,
Rich
-Original
Hi,
I have the setup of my xp100 plugging into my dlink gateway that i use
with a voip provider. I notice that when someone calls my pstn # that
goes to the asterisk box it works but when they hang up asterisk does
not recognize the hangup. What needs to be done to make it work with a
dlink
FYI I see it only on 1 in about 10-20 pickups...
On Wed, 19 May 2004, Luis Vazquez wrote:
Shaun Ewing wrote:
I'm not seeing this - using stable CVS from 14-05-2004.
Phone types are Grandstream Budgetone 101 using firmware 1.0.4.68, and Cisco
7940 using SIP 6.2.
-Shaun
Just to
Chris,
Isn't that what fromuser and fromdomain are for? These effect the Digest line. I use these in my friend entry to authenticate with FWD as well as other providers.
Bill Doll Jr
Chris A. Icide [EMAIL PROTECTED]
Chris A. Icide [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
Andrew Kohlsmith wrote:
The fact is the provider is running broken code. They should fix it.
That's the true bottom line.
Agreed but Rich needs a workaround. I think what I suggested will work and is
cheap -- a spare PC with a pair of ethernet cards in it, and a second
ethernet card for his
Ok,
but if i check codec with sip show channels * tell me it is the alaw codec
in use.
nicolas
Chris Stenton wrote:
put iLBC at the top of your allowed list of codecs in the general section
of sip.conf.
- Original Message -
From: nicolas [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
They are there.. read the configs/voicemail.conf.sample
Bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chris Clifton
Sent: Wednesday, May 19, 2004 11:43 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] AArgh, * and the
How does this have anything to do with this thread: Re: [Asterisk-Users] AArgh, * and the 7960.
Bill Doll Jr
Chris Clifton [EMAIL PROTECTED]
Chris Clifton [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
05/19/2004 09:42 AM
Please respond to
[EMAIL PROTECTED]
To
[EMAIL PROTECTED]
On Wed, 2004-05-19 at 11:37, Paul Tyreman wrote:
Hi,
When I installed my X100P card there was a two ring delay before the
phone started to ring.
I managed to solve that by stopping the Caller Line ID being passed to
the phone. I'm in the UK so that doesn't matter anyway.
The
Paul Tyreman [EMAIL PROTECTED] wrote:
When I installed my X100P card there was a two ring delay before the
phone started to ring.
I managed to solve that by stopping the Caller Line ID being passed to
the phone. I'm in the UK so that doesn't matter anyway.
The problem that I have now
On Wed, 2004-05-19 at 11:37, Paul Tyreman wrote:
The problem that I have now is that when a call comes in on the
external phone line and it is answered by a standard phone (not
connected to Asterisk), the IP Phones that are connected to it ring
for another two rings.
This is very irritating
On Friday 07 May 2004 09:55 pm, John Todd wrote:
OK, here's a configuration challenge: I want to have certain line
appearances able to be interrupted by various other line apperances
elsewhere in the office. This is harder to describe than it is to
demonstrate, so I'll do that:
Let's assume
David H Hickman wrote:
I have it working on an industrial single board pc. :)
Could you post some more info about your setup? Like board brand/model,
what kind of interfaces are you using and even some photos :-)
Seems a very interesting project... is there anybody else running a
small/compact
On Friday 07 May 2004 10:57 pm, Kyle Hagan wrote:
When I put someone on hold audio doesnt play and i get
mpg123: unknown option mono,
Any ideas. I searched wifi and archives.
Are the music file you are playing monophonic or stereophonic?
Anon
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On Wed, 19 May 2004, Mike Sturdee waxed:
Are there any variables or structure elements unique to a call that stay
till the end of a call -- including when caller enters a queue and then
bridged with agent. I am trying to get some variables about the caller in
I think the account code sticks
Hi-
I'm totally stumped configuring my TDM400P with one FXS and one FXO
module. Before I got the FXO module, I used to have an X101P, and
everything was working very well. Now * doesn't seem to recognize the
FXO channel. I've searched the wiki and the list archives. Stock Debian
3.0 stable
I have the setup of my xp100 plugging into my dlink gateway that i use
with a voip provider. I notice that when someone calls my pstn # that
goes to the asterisk box it works but when they hang up asterisk does
not recognize the hangup. What needs to be done to make it work with a
dlink
On Tuesday 18 May 2004 04:50, Manuel Wenger wrote:
I've just checked out the latest CVS from the 1.0-stable branch,
but DateTime() seems somewhat buggy. It says something like:
Tuesday May 18 11:46 AM 2004
instead of
Tuesday May 18th 2004 at 11:46 AM
(notice the wrong order of the words
Hello everybody
How can I get * timming in a server without usb and not rtc ?
Note: the server don't have USB port too
The server is remote and plans to be and serve as hosted *
for services
Regards
JMFA
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I am trying to get a building that has 3 company's on one asterisk server.
I need to make the IVR via DID take them to there right menu. So far I have
everything working except when they goto via standard_marco to an extension
and are sent to voicemail they are dropped off in the first menu and
Andrew Kohlsmith wrote:
The fact is the provider is running broken code. They should fix it.
That's the true bottom line.
I did not write that, Brian did.
Agreed but Rich needs a workaround. I think what I suggested will work
and is cheap -- a spare PC with a pair of ethernet cards in
On Sunday 09 May 2004 12:59 pm, Ed Mansouri wrote:
Hello,
I manage a small office and we have a 4-year old legacy analog PBX
manufactured by Iwatsu. We have four incoming analog lines that terminate
to 7 different desktop phones.
The interface to Iwatsu requires Windows and the Iwatsu
On Sunday 09 May 2004 02:02 pm, leonardo wrote:
I've been trying to play the default music on hold file, but no luck yet.
here is my configuration:
extensions.conf
[incoming]
exten = s,1,Dial,Zap/2|10
exten = s,2,Voicemail,u34
exten = s,102,Voicemail,b34
exten =
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