You have not shown us ANY example yet for which this
facility is *NEEDED*.
Well, I have users that get an account on my PBX.
I really don't care how many phones they want to use, hardware phones on
their desktop or soft phones on their laptop while travelling. It's still a user
with one account. W
Kannaiyan Natesan wrote:
* No, there's no quick fix for a 100 USD bounty
How much you estimate on quick fix?
I apologize for my Swenglish language...
I don't believe there's a quick fix at all.
If you want a quote for a fix, contact me off-list. But remember, that I believe
that fixing this is chan
Paul Mahler wrote:
Well, this is certainly getting exciting.
Yes, it is. Sorry for coming in late to this debate...
Andy, I took your advice and re-read the RFP.
It's actually RFC, not RFP. (teasing :-)
> So, gentlemen, help me out here. The spec says:
"The Address of record is the "SIP address
Registration to Astricon - the first Asterisk user's and developer's conference -
is now open. Astricon is taking place at the Atlanta Marriot September 22-24.
Digium is our Diamond partner in arranging this conference.
The web site is updated with information on hotel, prices and speakers for
the
3) Can anyone explain the meaning of "peer", "friend",
"user" in more details? For each case, what is the
role of Asterisk in SIP world, a UA, a proxy, or
others?
In some diagrams, Asterisk take's the role of a SIP Proxy, but it is *not*
a SIP proxy by design. Asterisk answers SIP calls and origin
Andrew Thompson wrote:
Peer: A connection that sends calls to asterisk.
User: A connection that asterisk sends calls out to.
Friend: an attempt at a combination of both, to simplify set up of phones
that send and receive calls. (There are several people here who will tell
you friend is evil.)
Actua
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--
Olle E. Johansson, Edvina.net AB, [EMAIL PROTECTED]
- Phone +46 8 594 788 10, Cell phone: +46 70 593 68 51
- IP phone: sip:[EMAIL PROTECTED]
- Address: Runbovägen 10, S
Sunday news is today published on a monday. Yesterday was fourth of
july, and I used that as an excuse for being off line yesterday.
(Sweden's national day is June 6th - and it's not yet a public holiday,
btw). Most of my Asterisk time lately have been used for producing
the registration site for A
Lenny Tropiano / asterisk.org Mailing list wrote:
We're doing some SIP development and have a question on "additional parameters"
supplied to the register (in this case maddr= and the non-standard clport= in
our example below).
What we're experiencing is the INVITE doesn't included these parameters
chouck wrote:
Im having a trouble understanding the config files setup even with some
documentation ive read such as the handbook, maybe im just illiterate.
Anyway do you think some one can point me to some examples of real
config files. Such as IAX, Extensions, and Sip. I just cant grasp the
Andres wrote:
Ernest W. Lessenger wrote:
Can anyone tell me how (and for how long) asterisk remembers the IP
address
for an IAX2 peer? Voicepulse has been going up and down for me, and it
seems
to have something to do with the IP address changing. Is there a way to
force asterisk to run gethostby
Eric Wieling wrote:
On Tue, 2004-06-29 at 09:53, Isamar Maia wrote:
I'm trying to do the following:
exten => i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
At that pint ${EXTEN} is "i". Try using ${INVALID_EXTEN}
Eric,
Thank you, I've added th
Welcome to the Asterisk users community!
Asterisk.org is a fast moving project. New code is added every day.
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Our community is also growing fas
Steven Critchfield wrote:
On Mon, 2004-06-14 at 11:13, Peter Mitchell wrote:
I can't seem to find the link to examples of asterisk installations for
different sized sites. I'm not after specific configuration of the conf
files, just an overview on what hardware/chassis cards people are
running and
Due to the dismissal of the stable-1.0 cvs source code, I've changed policy of the
Asterisk
Wiki - we now document CVS head. I would like all contributors to document which
version of
Asterisk (date) an addition was applied to, so readers can find out if a new function
works
with their version o
TC wrote:
Does Asterisk have a test plan for releases? It seems like if there was
a plan for testing that people could carry out (in a distributed
manner), our releases might not have so many quirky bugs. Most testing
can be automated; only some would have to be done in an interactive
manner with
Nicholas Bachmann wrote:
Olle E. Johansson wrote:
The decision is to base the future 1.0-release on the CVS head tree.
The current "stable-1.0" tree will be released as something intermediary,
maybe 0.91, and at that point it will be considered end-of-life.
At some point when we have c
Thank you very much for all feedback on Asterisk Sunday News!
This is the last issue for June. This week I'll go on holiday
and will be back with more news in early July.
My kids are getting summer leave this week and we'll be
visiting the south of England for a while. Another part of
Europe that s
Doug Kennedy wrote:
Hello,
I have modified the VoiceMailMain application to satisfy the request of
the "local users", i.e., my wife. She lost patience with too many
options (we have one mailbox, so we don't need to forward messages, or
reply to messages, or file them in 6 different folders...)
Senad Jordanovic wrote:
brian wrote:
That's the only way to make it work.
Devices behind nat, on same network, can call each other ONLY if
"canreinvite" is set to no? Is that what you are saying?
Canreinvite=yes *only* works if all devices are on the same side of the NAT, the
outside or the inside.
List,
Sorry for sending a private e-mail to the list. Tired...
While speaking about Astricon, we are looking to fill the last holes in the
tutorial agenda.
We agreed on two topics that we feel are missing:
* Dialplan tips and tricks
* Agent and call queues
If you are interested in teaching one of t
Rich,
I'm working on the tutorial agenda today. I've schedule a tutorial for you. Will need
a photograph
as well as some text for the web page that describes our tutorials.
Please read
http://www.astricon.net/astricon2004/tutorials.shtml
And you'll see what I need from you.
If you have any questio
Time for Duane to start implementing DNS SRV, since it's from now on is turned on
by default in CVS head.
Thank you, Mark!
/O ;-)
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Duane wrote:
If they want a simple method
of allowing calls they should use enum, least then it's obvious that it
isn't a email address and that they would possibly need to enable a few
things to make it work.
Enum doesn't replace SRV records at all.
Enum records point to a SIP URI.
To resolve t
This week, I've been really busy with the launch of a new Swedish Voip provider,
www.bbtele.se, so I haven't been able to follow the Asterisk community and haven't
been very responsive either. My apologies if you've tried to contact me and I did
not reply quickly or at all.
So to cover up (can't re
Duane wrote:
Olle E. Johansson wrote:
If you do not enable SRV records, you can't phone me. There's no
SIP proxy on edvina.net ;-)
Exactly my point, by ***DEFAULT*** Asterisk won't use SRV records, even
if it did, it doesn't support SRV correctly (as you pointed out), and
sip://[EMAIL PROTECTED] works perfectly well...
There are many benefits in stability when you use SRV records to find a
SIP proxy. However this requires that you have some sort of load balancing
between the servers.
It was a long time ago anyone mailed [EMAIL PROTECTED] Let's hope
we can prove th
Bruce Marler wrote:
All,
I have been beating my head against a wall trying to figure out how I would
implement a separate moderator code and participant code for the same
conference using meetme, the deal is I dont want the participants to be able
to join until the moderator is in the conference.
I
XISCOAIR wrote:
Hi everybody,
I'm trying to develop a web application for controlling if SIP users
are registered in * or not, and show it in a web.
My problem is that I don't now if it's possible to do a Shell Script to
control this:
1. Connect to console.
2. Execute command.
3. Obtain users re
Eric Wieling wrote:
On Mon, 2004-05-31 at 10:16, Duane wrote:
Andy Powell wrote:
Anything that's added to * that breaks how protocols work should be by default OFF not ON,
but that's just IMO...
I agree 100%, this has been very frustrating trying to work out why
Asterisk suddenly stopped accepti
Philipp von Klitzing wrote:
more stable/usable (eg, fewer bugs) then is Stable. As previously
stated, its time to lock down Head, fix the few remaining bugs, and
make "it" the well-overdue & historic v1.0 production release.
Might indeed by the better approach. However, will this same problem
oc
Spring is back in the Stockholm area. After a few day's worth winter re-runs,
the sun is back and night-time temperature is at least 5 degrees celsius. Time to
move out all my annual flowers and prepare the garden for summer.
Sweden is famous for our annual five week holidays - by law. From june to
Rich Adamson wrote:
It's a known fact that bugs are not being fixed in Stable, and even Mark
has suggested no one should be running Stable in a production environment.
On the other hand, there's not many bugs open in the bug tracker. Feature
requests and patches, but not bugs.
If you are aware of b
Ignace CARIA wrote:
I know I know this subject have been The most written subject about VoIP
:-)
If Asterisk is on a Public IP Address and a softphone behind the nat,
sip.conf must contains for this phone: nat=yes
And in most cases qualify=yes
The "nat=yes" makes asterisk don't trust the phon
Lars Boegild Thomsen wrote:
Hi Everybody
Any significant changes to CVS HEAD over the last couple of days. I've got
two asterisk boxes - both on public IP but one is dynamic. The one on
dynamic IP registers at the other one - that part is fine.
Calls going from the one with dynamic to the static
Graham Turner wrote:
was wondering if someone could give any indication of the messages that are
appearing on the console of an Asterisk PBX
WARNING[1116941120]: chan_sip.c:532 retrans_pkt: Maximum retries exceeded on
call [EMAIL PROTECTED] for seqno 103 (non-critical request)
192.168.90.1 is a 794
http://bugs.digium.com/bug_view_page.php?bug_id=0001411
We have a coder that are eagerly waiting for response and advice on how to
*finally* solve the problem with recursive mutexes on FreeBSD. If you are
running FreeBSD and know this kind of coding - log in to bugs and add
your comment while he's
Fabio Donaggio wrote:
Hi to all!!
I'm successful to connect Asterisk to PostgreSQL database...
If it's possible, can anyone learn me how to store sip user in
PostgreSQL database and how to configure voicemail??
The standard SIP channel in CVS only supports MySQL. However,
there might be script
Karl Brose wrote:
This is also closely related to Asterisk SIP's lack of proper [user
section] authentication/recognition for incoming calls. We've seen a lot
of posts here where new users have problems with this, but the real
problem is usually not acknowledged.
So tell me what's wrong with th
Welcome to the Asterisk users community!
Asterisk.org is a fast moving project. New code is added every day.
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Our community is also growing fas
Karl Brose wrote:
If the response to an OPTIONS is generated by a proxy server, the
proxy returns a 200 (OK), listing the capabilities of the server.
The response does not contain a message body.
Allow, Accept, Accept-Encoding, Accept-Language, and Supported header
fields SHOULD be presen
Karl Brose wrote:
Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or
not, Asterisk doesn't do it correctly either.
The host should respond with 200/OK if the call >could< succeed
theoretically if it were an INVITE or else it should send a
404 or maybe a 487(? hmm, have to look)
Here in Sweden, it's supposed to be springtime. A wonderful time of the year,
with sunny skies and wonderful weather. Almost summer. Today, it's not.
It's winter all over again with rain and only 3 degrees celsius outside.
Better to stay inside and write a weekly Asterisk newsletter :-)
This week's
Randy Bush wrote:
[foo]
type=friend
I do not beleive that will work for type=friend. If you use separate
type=peer and type=user blocks in sip.conf it may work. Expecially
if you also specify a port in the Dial().
Else, use the hostname (or a const).
hmmm. then, how do i let it be dynamic if it
Andres wrote:
Another problem with the SIPURA is the lack of a working STUN solution.
Even Grandstream works better with NAT today.
/O
I disagree. We have hundreds of Sipura customers using STUN with our
SER Solution. The are the most stable SIP UA we have ever tested. We
had to dump loads of
John Todd wrote:
At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote:
[snip]
Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk
can handled the NAT traversal all by itself with Qualify (as John points
out) disabling the NOTIFY will not change anything.
The NOTIFY will in no
John Todd wrote:
At 7:18 AM -0700 on 5/22/04, Bruce Komito wrote:
When I enable NOTIFY messages in my SIP device (Sipura), Asterisk
reports:
handle_request: Unknown SIP command 'NOTIFY' from 'xxx.xxx.xxx.xxx'
When I disable NOTIFY messages, * reports the device UNREACHABLE,
followed
by REACHABLE
Chris Stenton wrote:
Olle,
I looked at this a month or so ago and what you outline worked for anonymous
sip phones but it did not work for anonymous * sip connections. It would not
go to the context in general, you had to have a valid extension for some
reason.
Add a match-all extension in the co
Chad Brown wrote:
Does anyone have experience setting up * to accept anonymous sip UAs and
the dumping the call into IVR? I’m thinking this would be a good way to
have customers call us without creating an extension. So for my tests
have been focused on providing internal functionality.
Just con
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 p
Juan J. Sierralta P. wrote:
On Mon, 2004-05-17 at 09:57, Olle E. Johansson wrote:
Ignace CARIA wrote:
Perhaps stupid question but, is Asterisk a statefull or stateless proxy?
Asterisk is a no-SIP-proxy-at-all :-)
Again: Asterisk is not a SIP proxy. A SIP proxy doesn't handle media stream
During Astricon 2004, we'll have the first Asterisk developer's meeting.
The Asterisk developer's meeting is a one day meeting with discussions, brainstorms
and tests.
For each session, we need a white paper produced that outlines the topic to be
discussed.
If controversial, several whitepapers o
Ignace CARIA wrote:
Perhaps stupid question but, is Asterisk a statefull or stateless proxy?
Asterisk is a no-SIP-proxy-at-all :-)
Again: Asterisk is not a SIP proxy. A SIP proxy doesn't handle media streams and
doesn't terminate or originate calls. Asterisk does.
Asterisk is a stateful SIP UAS (Us
Another Asterisk week has gone by. A lot of changes has been submitted into
the CVS head, only a few to CVS stable.
CVS stable only changes for bug fixes now.
* Using MGCP? Please step forward!!
---
There are a number of MGCP bugs and fixes in the bug tracker that ne
tmpm wrote:
The ones that come to mind are en_ca, where every sentence has an ..eh?
ending.
"Our Weasels are eating doughnuts, and drinking beer ..eh?"
much like Aussie en_au which prepends a "G'day mate.. and ends with the
..eh?"
"G'day mate, our weasels be puttin our phone system on the barbie
Mike Machado wrote:
I have done a little work on asterisk and database integration. Below is
a link to app_dbmysql, modeled after Brian's app_dbodbc but for pure
MySQL.
I also ported the mysql-vm-routines.h to ODBC in case anyone is
interested.
COuld you please add your apps and changes to the bug
Could we do it like this:
[_][-]
Meaning
: ISO two letter language code
: ISO two letter country code
: up to 8 letter code for choosing a set of files within the lang code
directory
se-förför: Would look for files in the se/förför directory, use se syntax
en_au-male: Would look for files in the
Guan Yang wrote:
How about having a number of default language (locale) names, but allow
the user to define his own. For example some people might want to offer
the option of male and female voices for English.
You've always had that option. What we're sorting out is not the actual files,
but the
Robinson Tim-W10277 wrote:
And let's also spell things properly! Like 'internationalisation' ...'Weasels have got into your phone system' instead of 'gotten into your phone system...'
And 'please press the hash key..' instead of 'pound key'
There should probably be en_uk, en_us, en_ca, en_za, e
Stephan Wik wrote:
Splendid. Do we get a
us - American English
en - English
We certainly need to look into country variants.
Since we have learned quite a lot, I think we're ready to go for
en_us and en_uk
Also to let the friends on the other side of the Baltic sea add
se_fi :-)
Let's investigate
http://bugs.digium.com/bug_view_page.php?bug_id=0001485
After spending a lot of time saying numbers and dates, the Asterisk I18N project now
targets
voicemail. The voicemail prompts are very much based on english language syntax, which
works
for some languages and doesn't work for a lot of langu
Ignace CARIA wrote:
Is the SDP that negociate the codec to establish a voice communication?
Yes. The SDP, session description protocol, attachment to INVITE/ACK/OK
messages is the basis of selection of codecs. In Asterisk you either configure
codecs in the general section of SIP.conf or per user/p
Andy Powell wrote:
On 13/05/2004 at 14:57 Paul Crick wrote:
I've had a quick look through the mail list and wiki but haven't yet
resorted to looking at the meetme source code.. I see references to a
background agi script that can run if you're using Zap channels. Am I right
in saying that that s
The internationalization of Asterisk moves on. Now, there are two patches in
bugs.digium.com that needs your test report and feedback.
http://bugs.digium.com/bug_view_page.php?bug_id=0001600
http://bugs.digium.com/bug_view_page.php?bug_id=0001599
These will add support for Taiwanese signals and ind
Welcome to the Asterisk users community!
Asterisk.org is a fast moving project. New code is added every day.
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Our community is also growing fas
Steve Towlson wrote:
In order to try and keep some standardisation in Bridged Lines implementation on Asterisk using SIP,
you should look at the following IETF draft.
http://www.ietf.org/internet-drafts/draft-anil-sipping-bla-01.txt
There is work in progress within the SIP community on Bridged
Mark Elkins wrote:
I'd like to think you've got the same problem as me - something in the
new CVS head of the last day or so has stopped DTMF detection (eg - VM
menus don't work). Since I reported "DTMF Broken", there have been a few
updates to CVS as well - maybe its fixed? - perhaps try a 'cvs u
Mark,
Could you please add a SIP debug message with the SIP INFO?
/O
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* Read the config sample files! (even if you're an Asterisk guru)
-
For those of you that have a working installation that you keep using, this is a
reminder to check into the configs/ directory of the Asterisk source tree, regardless
In cvs head version of chan_sip, there's two new CLI commands:
* sip show peer
Show details of peer - configuration, registration status etc
* sip show subscriptions
List active SIP subscriptions to extension state changes in Asterisk
/Olle
___
Asteris
Philipp von Klitzing wrote:
Hi folks,
this does look like a bug to me: Asterisk replaces the @63.214.186.6 by
@context which obviously leads to a failure. Any comments, do I have a
configuration issue on my side that I missed?
We don't support 302 redirects to other hosts/domains now.
It's a b
James Sizemore wrote:
I am going to change my Digest realm to match my DNS SVR record.
I dug through the code in chan_sip.c and on line 2748 I found it hard
coded :
snprintf(tmp, sizeof(tmp), "Digest realm=\"asterisk\", nonce=\"%s\"",
r\anddata);
I'm going to change this to :
snprintf(tmp, si
AJ Grinnell wrote:
What is the best free stun server out there? The one that I have looked at
from vovida requires two NICs. Is this neccessary?
I don't know which server is the best one, I'm using the one from IPtel.org.
You need two IP addresses for STUN to work, with a proper implementation
yo
This looks like something is not working as it should be.
Please open a bug report, add a full SIP debug copied from an asterisk started with
'-dr' including the text between the packets.
Also add your extensions.conf and sip.conf
State your platform (O/S) and the version of Asterisk
http://bugs.digium.com/bug_view_page.php?bug_id=0001019
"This patch allows to bind RTP flows to a specific interface, additionally the SDP session descriptor get's coherent with the same address
that is used for RTP traffic, this includes sip<->sip and sip<->voicemail and others(not tested, but s
http://bugs.digium.com/bug_view_page.php?bug_id=0001476
If you're calling voicemail from IAX clients and want voicemail or other IVR
prompts to be in some other language than english, this is a patch that you
need to test.
This patch allows you to set the default language for a user/peer so that
w
http://bugs.digium.com/bug_view_page.php?bug_id=0001429
Saying numbers is not always easy, especially if you want one software to be
able to do it in many different languages with different syntaxes for
how to construct numbers like "one-hundred-twenty-four" or
"fem-hundra-tjugo-åtta" or "quatre-v
still - no dice when it comes to registering this puppy. I used the web
interface to specify the username/password but still nothing.
Any ideas or docs I could look at to get this Polycom phone setup?
--
Olle E. Johansson, Edvina.net AB, [EMAIL PROTECTED]
- Phone +46 8 594 788 10
http://bugs.digium.com/bug_view_page.php?bug_id=0001474
If you're from NZ and need this, please test if this is the correct setup.
Add your comments, positive or negative, to the bug tracker. We need
confirmations from the community to move ahead.
Thank you!
/O
Geert Nijpels wrote:
Ian White wrote:
On recent releases of the snom200 firmware, the MWI indicator will
turn on, but won't turn off when the message has been checked. It
works on firmware 2.03o, but not in 2.04g or newer. I filed a bug
report with snom, but they're claiming it is an asterisk i
http://bugs.digium.com/bug_view_page.php?bug_id=0001471
Indications for Russia
Please check if this works for you - if you know how it should sound in
Russia and other countries that have the same phone system. If it works
or if you object, please confirm or add your comments to the bug tracker.
Welcome to the Asterisk users community!
These are exiting times for Asterisk.org. We're getting close to a
1.0 release, working hard to fix all reported bugs in Asterisk.
At the same time, the community is growing and we're having a lot
of interaction, on t
Sören!
Tusen Tack :-)
I'll add your input and will see what I can do to fix this. Does the other danish
people agree?
For the rest of you - please add your input to the bugtracker. For those
of you who have earlier contributed with patches, answer my e-mails!
If I don't get disclaimers from the fre
The recent addition of recursive mutexes to Asterisk is causing a lot of problems
on FreeBSD servers. I need help from someone that knows mutexes on FreeBSD to
make it work, otherwise the FreeBSD port of 1.0 will be useless.
See bug report http://bugs.digium.com/bug_view_page.php?bug_id=0001411
for
Matt Riddell wrote:
I have the sounds for French, but can record more if necessary. They are
available at www.sineapps.com
Is a disclaimer required on these?
To have all bases covered, sign a disclaimer, fax it in and add the sounds
to the bug report.
Is all the digit-files for the french say.c pa
Brancaleoni Matteo wrote:
hi Olle.
I have a patch for italian.
Great.
should it be for plain say.c or for your modified say.c ?
If you have one that builds on my patch, that'll make life easier
for me. THank you!
Also I have some the .it audio files, I'll ask
if I can distribute them
(perhaps wi
http://bugs.digium.com/bug_view_page.php?bug_id=0001429
* Support for other language syntaxes in saynumber
Accidentally I opened this can of worms to see if we can add support
for other language syntaxes for saying numbers. Seems like Swedish,
english and norwegian follow the same syntax. I've in
Chris Orme wrote:
exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
Isn't the 'r' forcing a 'ringing' signal from start, regardless
of what the device you are calling are signalling. If you are calling
a SIP device, that device might return 'busy' and that's propably
why you first hear 'ringing' and
[EMAIL PROTECTED] wrote:
I get this warning from Asterisk and I want to assess whether it is important, and if so, if I should complain to the telephone manifacturer or start up my programmer's editor:
chan_sip.c:5152 handle_response: Host '172.31.1.7' does not implement 'NOTIFY'
What does this m
http://bugs.digium.com/bug_view_page.php?bug_id=0001396
Indications for australia. Please confirm if this works for you so we know if this
is something to include in CVS or not.
Thanks, mate :-)
/O
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We need to architect a general structure for saying numbers in voices.
Say.c is broken as it is now and it needs to be changed.
If we work quickly, this can be sorted out and fixed to 1.1, if not
before that.
There's a number of separate patc
Matteo Brancaleoni wrote:
eh, very good idea...
but how about for alaw people?
And E1's and EuroISDN :-)
Any plans to make another conference in EU world?
We'll start with one conference for everyone. As I'm also
based in Europe, having a European followup is an idea
that is within our plans. (Al
mation.
Looking forward to meeting you all in Atlanta!
Steven SokolOlle E. Johansson
[EMAIL PROTECTED] [EMAIL PROTECTED]
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T
Eric Wieling wrote:
Olle E. Johansson wrote:
Eric Wieling wrote:
Erick Weber V. wrote:
I'll Like to now how to insert a pause on a SIP string. I have a ATA
186 and
a FXS => FXO converter so I will like to program a extension that
can be
dialed and it will dial the ATA extention,
Eric Wieling wrote:
Erick Weber V. wrote:
I'll Like to now how to insert a pause on a SIP string. I have a ATA
186 and
a FXS => FXO converter so I will like to program a extension that can be
dialed and it will dial the ATA extention, wait for dial tone and then
dial
the phone number.
You ca
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RE: Asterisk-Use
The latest recursive mutex additions in Asterisk will not compile on my
FreeBSD 4.9 systems. Anyone out there with FreeBSD or OpenBSD that got it working?
I guess I need an update to Gnu Pthreads, but can't find anything by looking
in ports. There's something called linuxthreads, but it's over my h
If you're using MGCP, we need your help. There's a patch in bugs.digium.com that
needs testing by the community. Please spend some time testing and adding your
comments to the bug tracker.
The author writes:
--
I'm trying to make work Asterisk against a Cisco IAD2431 with chan_mgcp
We're getting closer and closer to a 1.0 release of Asterisk. In order to get there,
the development is now 110% focused on solving major, critical and crash bugs.
(And yes, if you follow the CVS updates, you'll see the impossible extra 10% :-)
* YOU'RE NOT FORGOTTEN, BUT ON HOLD (WITHOUT ON-HOLD-M
Steven Kokinos wrote:
Hello-
I have several Sipura SPA-2000's at different locations (all behind Linksys WRT54G boxes). When setting:
nat=yes
qualify=yes
Things work properly about 90% of the time, however, if a remote end loses the connection briefly, then asterisk can't see the adapt
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