The T option is for the # transfer which is handled by Asterisk, in your
case the phone has a transfer button and is able to send SIP messages
telling Asterisk that the call should be transferred.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
Brian West wrote:
Also I must point out that your NAPTR record is a bit wrong:
wrong:(bind9)
!+(.*)!iax2:foofone/1!
Read again, Brian.
The text clearly states that the shell eats up one of the slashes, so we have to
double-quote.
The only part I would add is the advice to make sure that
I have an interesting problem now. I use asterisk to connect
to both FWD and a sip provider here in sweden. suddenly, (i know
my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try
to make a call using this provider. FWD still works fine, and I can call directly
Olle,
This is the complete extensions.conf. I wasnt getting the SIPDOMAIN right.
Rest of your script/configuration works only if ${SIPDOMAIN} works
Am I missing anything in this? I had the latest CVS checkout this morning,
i.e., 1st Dec. 12.00 Noon GMT +5.30.
Also, I see them on eBay all the time for around $35 US.
--
Josh Roberson
Indigent Networks
1.877.677.9647 x1
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lists
Sent: Sunday, November 30,
That is their new price
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josh
Roberson
Sent: Monday, 1 December 2003 7:00 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] cisco 7960 power suplies?
Also, I see them on eBay all the
ranga wrote:
This is the complete extensions.conf. I wasnt getting the SIPDOMAIN right.
Rest of your script/configuration works only if ${SIPDOMAIN} works
Am I missing anything in this? I had the latest CVS checkout this morning,
i.e., 1st Dec. 12.00 Noon GMT +5.30.
Ranga,
I agree, seems like the
Here it goes
Sip read: CLI
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Content-Length: 116
Contact: sip:192.168.68.12
Call-ID: [EMAIL PROTECTED]
Content-Type: application/sdp
Max-Forwards: 70
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
CSeq: 1 INVITE
To: sip:[EMAIL PROTECTED]
Via:
Joe Dennick wrote:
I think you need to better define your Queue Environment in
extensions.conf. Below is what I've got in mine, and it seems to work
quite well:
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
put the core file into gdb, backtrace it
and then we'll have some useful information:
# gdb asterisk corefile
and issue bt on gdb console
or run asterisk directly into gdb :
# gdb --args asterisk -vvvgc
play with it and when it seg faults, issue a 'bt'
command
matteo.
Il lun, 2003-12-01 alle
David M. Wilson schrieb:
Hi there!
I'm currently considering various PBX solutions for our office telephone
network, and would very much like to use Asterisk. Currently, my
research is incomplete. I have been recommended to use the above cards,
but it is unclear from my Googling whether my
Anton Yurchenko wrote:
Joe Dennick wrote:
I think you need to better define your Queue Environment in
extensions.conf. Below is what I've got in mine, and it seems to work
quite well:
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten =
I'm pretty sure that is incorrect. The inside_net is the ip address of
the asterisk server, and the inside_mask is the subnet mask. At least
that is how I have mine setup in my sip.conf, and it works.
inside_mask for the internal mask would make more sense to me as well :)
--
Leif
hallo all.
how i must setting extension.conf or other conf, if i want do this. If asterisk
receive this error: Failed to authenticate on INVITE asterisk still giving normaly
call, normaly signal dont fast busy. I want - fast busy, and disconnect all
connection. (in sip.conf - have:
Michael Devenijn wrote:
count me in
I'm coming to Paris Dec 19. I was wondering if there was any interest in
having an Asterisk get together in Paris sometime near there. Any one out
there interested? Anyone in Paris who could help organize something like
that? :)
Mark
I'll be there :)
Thomas Dingermann wrote:
David M. Wilson schrieb:
Hi there!
I'm currently considering various PBX solutions for our office telephone
network, and would very much like to use Asterisk. Currently, my
research is incomplete. I have been recommended to use the above cards,
but it is unclear from my
Mark,
We're happy to host something in London if you were dropping round these
sides.
Tan
Telappliant.com
Voiptalk.org
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: 30 November 2003 20:45
To: [EMAIL PROTECTED]
Subject: Re:
And while you are in Europe, why not also do Brussels ? ;)
zoa.
At 11:16 1/12/2003 +, you wrote:
Mark,
We're happy to host something in London if you were dropping round these
sides.
Tan
Telappliant.com
Voiptalk.org
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Michael Bielicki wrote:
Thomas Dingermann wrote:
David M. Wilson schrieb:
Hi there!
I'm currently considering various PBX solutions for our office
telephone
network, and would very much like to use Asterisk. Currently, my
research is incomplete. I have been recommended to use the above cards,
Paul Liew wrote:
- Original Message -
From: Anton Yurchenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 29, 2003 3:34 AM
Subject: Re: [Asterisk-Users] call waiting disable in sip
what would happend if all operators are busy? would app_queue exit?
would it
And while you are in Europe, why not also do Brussels ? ;)
zoa.
Hey, surprise! Just discovered it on the web:
http://graphics.cs.uni-sb.de/~rainer/tour.jpg
Mark is going on tour!
SCNR,
Rainer
--
http://graphics.cs.uni-sb.de/VoIP/
pgp0.pgp
Description: PGP
On Mon, Dec 01, 2003 at 03:33:50PM +0200, Anton Yurchenko wrote:
I have a problem, when caller is in Queue and the operator is busy
answering other call he/she still hears the call waiting signal.
I have the latest cvs and incominglimit is set to 1. But here is what *
shows when the operator
zoa [EMAIL PROTECTED] said:
And while you are in Europe, why not also do Brussels ? ;)
Amsterdam!!
--
Cees de Groot http://www.tric.nl [EMAIL PROTECTED]
tric, the new way helpdesk/ticketing software, VoIP/CTI,
web applications, custom
Second that !
-Original Message-
From: Cees de Groot [mailto:[EMAIL PROTECTED]
Sent: Monday, December 01, 2003 2:35 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in
Paris
zoa [EMAIL PROTECTED] said:
And while you are in Europe, why not
Sofia (Bulgaria) !!! :)))
Cees de Groot wrote:
zoa [EMAIL PROTECTED] said:
And while you are in Europe, why not also do Brussels ? ;)
Amsterdam!!
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[EMAIL PROTECTED]
zoa [EMAIL PROTECTED] said:
And while you are in Europe, why not also do Brussels ? ;)
Amsterdam!!
Oslo!
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Low, Adam wrote:
Second that !
-Original Message-
From: Cees de Groot [mailto:[EMAIL PROTECTED]
Sent: Monday, December 01, 2003 2:35 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in
Paris
zoa [EMAIL PROTECTED] said:
And while you are
Cees de Groot wrote:
zoa [EMAIL PROTECTED] said:
And while you are in Europe, why not also do Brussels ? ;)
Amsterdam!!
Feel forced to add STOCKHOLM!
/O ;-)
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On Fri, 28 Nov 2003 18:15:38 +0100, Peer Oliver schmidt wrote
David M. Wilson wrote:
Hi there!
I'm currently considering various PBX solutions for our office
telephone
network, and would very much like to use Asterisk. Currently, my
research is incomplete. I have been recommended to
Amsterdam!!
I had my laptop and suitcase stolen in Amsterdam the one time I went
there, after hearing someone talk about how safe a city it was over
dinner. Most importantly, also stolen was my (apparently irreplacable)
copyleft shirt (yellow/gold with large blue backwards (C) symbol on front
Why not Sardinia, in Italy?
good food, nice people :)
and real italian pizza coffee..
matteo
Il lun, 2003-12-01 alle 14:35, Cees de Groot ha scritto:
zoa [EMAIL PROTECTED] said:
And while you are in Europe, why not also do Brussels ? ;)
Amsterdam!!
--
Brancaleoni Matteo [EMAIL
On Mon, 2003-12-01 at 15:17, Olle E. Johansson wrote:
Cees de Groot wrote:
zoa [EMAIL PROTECTED] said:
And while you are in Europe, why not also do Brussels ? ;)
Amsterdam!!
Feel forced to add STOCKHOLM!
Well, in this case, I have to add BARCELONA !!! ;)
/O ;-)
Hello,
Can somebody help mw with set up Announcment while phone is ringing ?
Is it suppouse to be like this:
Dial(SIP/[EMAIL PROTECTED],A(test)) ?
Bart
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[EMAIL PROTECTED]
I had my laptop and suitcase stolen in Amsterdam the one time I went
there, after hearing someone talk about how safe a city it was over
dinner. Most importantly, also stolen was my (apparently irreplacable)
copyleft shirt (yellow/gold with large blue backwards (C) symbol on front
and GPL
For a door release, I have a cheap radio-shack device which is supposed
to light up a lamp when a phone rings. Basically, it has a contact
which is activated by the ring signal on a telephone line.
I wired this up to the door release in the office, and have it hooked
up to our (non asterisk)
Mark Spencer wrote:
Amsterdam!!
I had my laptop and suitcase stolen in Amsterdam the one time I went
there, after hearing someone talk about how safe a city it was over
dinner. Most importantly, also stolen was my (apparently irreplacable)
copyleft shirt (yellow/gold with large blue
Ranga,
I'm sorry, I can't find the error in this configuration. I called on IP address myself,
and my Asterisk picked out the IP address into the domain part and dialed out.
I'm stuck. Anyone else that see the problem?
/O
ranga wrote:
Here it goes
Sip read: CLI
INVITE sip:[EMAIL PROTECTED]
no, the A option is used to play an announce to the
called party as soon as he answers.
matteo.
Scrive Bartosz Jozwiak [EMAIL PROTECTED]:
Hello,
Can somebody help mw with set up Announcment while phone is ringing ?
Is it suppouse to be like this:
Dial(SIP/[EMAIL PROTECTED],A(test)) ?
Those things generally happen in Amsterdam. And in Kristiania in
Copenhagen. The usual problem: Smoking too much pot
Actually we just had dinner and had left our things in his car which
(according to the police inspector) was entered through the trunk using a
half a tennis ball.
Mark
In my extensions.conf I have two 2 contexts (sip and pstn) with two
extensions each one, but the command show dialplan, on the CLI, show me
only the 2 contexts without any extensions and more the config in
parking.conf.
I can't do any call !
What I can do correct it ?
Best regards,
Miguel
That's an interesting solution.
Caveats :
- To have a door phone and door release would require 2 ports on the pbx
unless you don't want to be able to call the doorphone, or release the
door while the caller is still on the line.
- if the doorphone is a regular phone and uses this method,
On Sat, Nov 29, 2003 at 11:28:56PM -0600, Mark Spencer wrote:
I'm coming to Paris Dec 19. I was wondering if there was any interest in
having an Asterisk get together in Paris sometime near there. Any one out
there interested? Anyone in Paris who could help organize something like
that? :)
Hey, surprise! Just discovered it on the web:
http://graphics.cs.uni-sb.de/~rainer/tour.jpg
Mark is going on tour!
Not sure if this is real info or just a JPG that someone created.
Is Stuttgart a definate date on the 30th? If so, where in Stuttgart??
Robert
Friedrichshafen
At 08:33 1-12-2003 -0600, you wrote:
Amsterdam!!
I had my laptop and suitcase stolen in Amsterdam the one time I went
there, after hearing someone talk about how safe a city it was over
dinner. Most importantly, also stolen was my (apparently irreplacable)
copyleft shirt (yellow/gold with large
Why stick with moving from city to city in Europe? Why not just rent
the whole nation of Liechtenstein and have an * party?
http://www.rentastate.com/en/flash5.html
This (almost) unreal extension of capitalistic excess brings up an
interesting point: if this whole nation is small enough to
Perhaps it's because the Contact: field does not have an extension in
it, just an IP address? This is a guess without really thinking
about it too much.
JT
Ranga,
I'm sorry, I can't find the error in this configuration. I called on
IP address myself,
and my Asterisk picked out the IP
Amsterdam!!
I had my laptop and suitcase stolen in Amsterdam the one time I went
there, after hearing someone talk about how safe a city it was over
dinner. Most importantly, also stolen was my (apparently irreplacable)
copyleft shirt (yellow/gold with large blue backwards (C) symbol on
Those things generally happen in Amsterdam. And in Kristiania in
Copenhagen. The usual problem: Smoking too much pot
Actually we just had dinner and had left our things in his car which
(according to the police inspector) was entered through the trunk using a
half a tennis ball.
Mark
Yep
Anyone have any thoughts on this since last week?
I am having issues with Privacy Manager and Zapateller.
If I set callerid= on a sip user zapateller sends the tones
If I set callerid=Anonymous 8475551212 zapateller doesn't send the
tones
If I call from a phone after dialing *67
Those things generally happen in Amsterdam. And in Kristiania in
Copenhagen. The usual problem: Smoking too much pot
I have to object to that, as a rule of thumb the Dutch only rob tourists who
are dressed like tourists and act like tourists, that's what we all agreed
to here and live by --
Walker Haddock wrote:
On Mon, Dec 01, 2003 at 03:33:50PM +0200, Anton Yurchenko wrote:
I have a problem, when caller is in Queue and the operator is busy
answering other call he/she still hears the call waiting signal.
I have the latest cvs and incominglimit is set to 1. But here is what *
rnc Info Lists wrote:
Hey, surprise! Just discovered it on the web:
http://graphics.cs.uni-sb.de/~rainer/tour.jpg
Mark is going on tour!
Not sure if this is real info or just a JPG that someone created.
Is Stuttgart a definate date on the 30th? If so, where in Stuttgart??
Robert
Not sure if this is real info or just a JPG that someone created.
Is Stuttgart a definate date on the 30th? If so, where in Stuttgart??
These dates were just made up bye Rainer and me.
--
Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP
pgp0.pgp
Description: PGP signature
Why not create a listing of the Asterisk resellers. Have a link off the
main Digium page and post what asterisk services that particular
reseller offers. This way people who are just getting into asterisk
know where they can go for commercial support. Maybe the reseller could
offer some sort of
Hello:
I need to prepare some detailed stats from asterisk, and I'm asked to show
data I don't know how to obtain it: It's the 'final' number (don't know
what's its name)
In the stats I have to show the caller_id (I have it), the called_id (I have
it) and the final number that actually
What does it mean ??
WARNING[265236]: File dsp.c, Line 1198
(ast_dsp_process): Unable to detect process 2 framesWARNING[265236]: File
dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2
framesWARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect process 2
All,
I would like to play an announcement to the user on what external line a
call came in, right before this call get bridged to this user. How would I
go about implementing this in * ?
Regards,
Hans
--
The contents of this e-mail are intended for the named addressee only. It
contains
Il lun, 2003-12-01 alle 15:36, Brancaleoni Matteo ha scritto:
Why not Sardinia, in Italy?
good food, nice people :)
Since this thread has already grown way larger than it should, may I add
Venice? :)
--
Emanuele
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John Todd wrote:
Perhaps it's because the Contact: field does not have an extension in
it, just an IP address? This is a guess without really thinking about
it too much.
The Contact: field sure looks weird, but the SIPDOMAIN comes from
the INVITE - or?
Ranga, please check your debug log in
Hans although your somewhat right I don't think its fare to ask all
tourists to leave their clothes at customs and to don clogs and ride a
battered old bike around the city. I also must say that from my experience
its very rarely (I've never heard of it) the native Dutch that perform
these
Don't use dtmfmode=inband on GSM codec it'll only work on G711.
Martin
On Mon, 1 Dec 2003, Bartosz Jozwiak wrote:
What does it mean ??
WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2
frames
WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process):
Hmmm... what size was that T shirt ? (c;
Large.
Mark
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On Mon, 01 Dec 2003 00:46:22 +0100, Brancaleoni Matteo [EMAIL PROTECTED]
wrote:
Hi.
Isn't possible to have a statically linked version for linux?
[EMAIL PROTECTED] iaxcomm]$ ./iaxcomm
./iaxcomm: error while loading shared libraries: libwx_gtk_xrc-2.4.so: cannot open
shared object file: No
Mark Spencer [EMAIL PROTECTED] said:
Most importantly, also stolen was my (apparently irreplacable)
copyleft shirt (yellow/gold with large blue backwards (C) symbol on front
and GPL preamble on back) which no amount of effort has managed to find a
replacement for and it's *that* part i've never
Yes I would like it too !
- Original Message -
From: Vledder, Hans [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 01, 2003 1:50 PM
Subject: [Asterisk-Users] Call Announcement - How To ...
All,
I would like to play an announcement to the user on what external line a
We have an h.323 based IVR platform. When we make a call to it using an
h.323 phone, it can see the callers number (ANI), but when we make a call to
it via asterisk, the call goes through OK, but we don't get the number. How
can I make this work?
h323.conf
===
[general]
port = 1720
bindaddr
You might need to edit the code of chan_zap.c You need two things to fix:
outgoing calls and incoming calls. Outgoing you should be able to find
pri_call call and do chan-1 for chans16. For incoming calls you need to
find the handling of PRI_EVENT_RING or something like that and do chan+1
for
That's a good idea. There is already a resellers list on the Digium site,
but perhaps a line or two about specialities could be added.
Cheers
Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
URL:www.evtmedia.com
At 16:46 1-12-2003 +0100, you wrote:
Well the Aussie's recently announced an additional travel warning for The
Netherlands due to the increased level of petty crime although I feel it
was a little extreme. The petty crime problem is very much specific to
Amsterdam and foreign crims come into
--- Vledder, Hans [EMAIL PROTECTED] wrote:
I would like to play an announcement to the user on what external line a call came
in, right before this call get bridged to this user. How would I go about
implementing this in * ?
Use the A option to the Dial application:
'A(x)' -- play an
I would like to release prepaid
application.
But I have a small problem, we are using their
Cisco prompts (nice lady voice)
And I do not know if it is ok to release
it.
Bart
Outlook Express mangled my message before, so I've reattached it...
Hopefully, it'll go this time...
- Original Message -
From: Areski [EMAIL PROTECTED]
To: Asterisk-Users Mailing-list [EMAIL PROTECTED]
Sent: Tuesday, November 25, 2003 12:13 PM
Subject: [Asterisk-Users] Outgoing-call
I have had several cases where the message waiting indicator was stuck
in the on state
with Cisco 7960 SIP phones. Here are the two cases:
1. Single extension that mapped to a single voice mailbox. Restarting
Asterisk or getting a
new voicemail then clearing it fixed the problem.
2. Three
Hi!
I have to know who answered the call, how can I do this?
I'm currently looking for a Dial as the last command and getting the data
for that command, but doesn't seem a solid solution.
Very good question - I've also run into this problem. I do think that the
CDR could use some
On Mon, 2003-12-01 at 05:52, Darren McIntosh wrote:
In my configuration I have internal SIP clients registering from
192.168.0.0/28 and my * address is at 192.168.0.100. Using the host address
of the * box as the inside_net variable the audio from 192.168.0.0/28 was
sent to the outside_addr
Anyone able to confirm whether the T400P (or any other Zap device) works
with the 2.4.23 kernels?
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Mini meeting next week lubomir ?
i'll be there starting on monday :)
zoa.
At 16:00 1/12/2003 +0200, you wrote:
Sofia (Bulgaria) !!! :)))
Cees de Groot wrote:
zoa [EMAIL PROTECTED] said:
And while you are in Europe, why not also do Brussels ? ;)
Amsterdam!!
--- Bartosz Jozwiak [EMAIL PROTECTED] wrote:
I would like to release prepaid application.
But I have a small problem, we are using their Cisco prompts (nice lady voice)
And I do not know if it is ok to release it.
Why don't you release it without the prompts then? It would probably be nice if
Bartosz Jozwiak wrote:
I would like to release prepaid application.
But I have a small problem, we are using their Cisco prompts (nice lady
voice)
And I do not know if it is ok to release it.
I don't know about the rights to the Cisco prompts, so be sure
to remove them and then release it. User
Probably a good idea to re-record the prompts, to avoid intellectual
property issues later on. Plus, you'll likely need to add more prompts in
the future, and so you can have the voice match what you've already
recorded.
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California
I would love to send it to couple of peoples so thay can write some docs and
clean th code.
OK ?
Who would like to do it ?
Bart
- Original Message -
From: Scott Stingel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 01, 2003 3:57 PM
Subject: RE: [Asterisk-Users] PREPAID
I would like to release prepaid application.
But I have a small problem, we are using their Cisco prompts (nice lady voice)
And I do not know if it is ok to release it.
Bart
I will agree with the comments of others on this topic.
You should _not_ include the prompts from Cisco. That is almost
Speaking of voice prompts, could anyone tell me why the pre-recorded
prompts sometimes sound garbled, but the voicemail messages themselves
sound fine? Is it the format of the prompts?
Stephen
I would like to release prepaid application.
But I have a small problem, we are using their Cisco
Hi All,
I hope this is the right list for this sort of request.
I'm wondering if you all could recommend (or are) an asterisk
integrator. I've been following the lists, etc, and have played with the
software, but just don't have the time to really figure it out, nor to
deliver a solution in a
Yeah, the cities in the Netherlands suck. That's of course nothing too
surprising, but someone telling you that Amsterdam is safe, especially
regarding theft, it just plain stupid.
I've heard that a Canadian not even living in Amsterdam told Mark that
it was safe to put that stuff in the
you are welcome zoa :)
I'll be happy if we make a little * party here ;)))
Lubo
zoa wrote:
Mini meeting next week lubomir ?
i'll be there starting on monday :)
zoa.
At 16:00 1/12/2003 +0200, you wrote:
Sofia (Bulgaria) !!! :)))
Cees de Groot wrote:
zoa [EMAIL PROTECTED] said:
And while
Cees de Groot wrote:
Mark Spencer [EMAIL PROTECTED] said:
Yeah, the cities in the Netherlands suck. That's of course nothing too
surprising, but someone telling you that Amsterdam is safe, especially
regarding theft, it just plain stupid.
well. they'd be a lot better if there weren't all those
Andrew Thompson wrote:
Outlook Express mangled my message before, so I've reattached it...
Hopefully, it'll go this time...
mebbe you should switch to a better mailer, like mozilla for instance ;)
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Anyone have tried * with kerio SIPPS softphone?
It registers ok with *, but
I get missing sdp body message when dialing any extension.
Thanks.
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Hi!
I've heard that a Canadian not even living in Amsterdam told Mark that
it was safe to put that stuff in the trunk. I am sure the junkie wasn't
scoping you guys doing that and only was counting the cobblestones in
the street :)
During first visit to Amsterdam by car (with a German number
what would be nice is to get this on MeetMe app. so that you can announce someone
joining the conf call
Dave
[EMAIL PROTECTED] 12/1/2003 11:11:49 AM
--- Vledder, Hans [EMAIL PROTECTED] wrote:
I would like to play an announcement to the user on what external line a call came
in, right before
uh-oh :)
22:30:13: can't read from file descriptor 4 (error 21: Is a directory)
22:30:13: Failed to read PID from lock file.
never used iaxcomm on that box :)
Matteo.
Il lun, 2003-12-01 alle 18:14, Michael Van Donselaar ha scritto:
On Mon, 01 Dec 2003 00:46:22 +0100, Brancaleoni Matteo [EMAIL
http://bugs.digium.com
It is appreciated if you submit your own code; otherwise I doubt
anything will be done. On the Grandstream phones I think the call would
be dropped if the transfer fails by disabling it in asterisk.
-Original Message-
From: [EMAIL PROTECTED]
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nicolas Gudino wrote:
| If the terminating tail circuit has cancelable echos and if the echo
| canceler is enabled, you will hear echo for the first few utterances and
| then it will die away. After a few seconds of speech, the echo should be
| gone or
I would like to release prepaid application.
But I have a small problem, we are using their Cisco prompts (nice
lady voice)
And I do not know if it is ok to release it.
Bart
I will agree with the comments of others on this topic.
You should _not_ include the prompts from Cisco. That is almost
Hello.
Do you have your linux starting in Graphical mode ( init mode 5 ) ? I
also had a problem with audio on my sip phones and it was generated
because of the frame buffering that my video drivers use ( I have *
installed in my personal computer ), so I changed the startup mode to 3
and only
Sure I agree so I've already removed the cisco prompts.
And will record something eals.
Bart
Quoting John Todd [EMAIL PROTECTED]:
I would like to release prepaid application.
But I have a small problem, we are using their Cisco prompts (nice
lady voice)
And I do not know if it is ok to
Hi,
VoiceGlo is comercial version of Asterisk? :)))
loo
Take a loock on http://www.voiceglo.com/
The softphone is IAX :)
Best regards,
Chris HARIGA
Techselesta Inc.
http://www.techselesta.com/
Erik,
I have just finished a job in Toronto and will be in the area for a few
days, I could arrange to stop by on my way home.
Contact me off list - [EMAIL PROTECTED], 512-789-5214
Robert J Rae
Softprofit Solutions
- Original Message -
From: Erik LaBianca [EMAIL PROTECTED]
To:
This is a resend - customer complains
thatsome phone systems they call don't respond to key tones, please
advise.
Thanks Rob.
- Original Message -
From:
Softprofit
Solutions
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 7:34
PM
Subject: tone
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