Progressinband=yes under [general] in sip.conf
NEXT!!!
- Joshua Colp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave DeChellis
Sent: Tuesday, December 28, 2004 8:57 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk / 183
A register line simply tells the provider where to send your calls. It is
still up to you to setup a user entry in your iax.conf that they will use to
send the call. This is simply a case of you not properly configuring iax.
NEXT!!!
- Joshua Colp.
-Original Message-
From: [EMAIL
Hello Everyone,
My name is Joshua Colp. I'm an employee of VoiceConduits, I'm responsible
for the software that runs the backend routing and billing systems. I have
not yet discussed with my boss what I'm about to write here but I would like
to end this statement now. VoiceConduits is not a scam
This is person normally and it is NOT AN ERROR. It just states that it's
ignoring the port. Simple as that? Okay? Okay? Everyone repeat after me:
WARNINGS ARE NOT ERRORS. Thank you.
- Joshua Colp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
/? Probably
not, but all of this was outlined there so please before asking
questions check the sites.
In regards to you using Broadvoice for
service with customers I wouldnt expect to be in business for a
long time. Their terms of service specifically prevents stuff like this. Have
fun.
- Joshua
Who has an
answer for this desperate problem? file has an answer for this desperate
problem! Who me? Yes you! So true!You wouldn't have happened to have
downgraded from CVS head to CVS stable by any chance? Stable has no idea
what realtime is... so if the old realtime modules are present, they
Asterisk stable still has the old capability of 'sipfriends' and
'iaxfriends' for putting data into MySQL for peers. This is what the
changelog note is referring to. If you need more information on either of
the above, feel free to browse the voip-info.org website! Have a great day.
- Joshua Colp
however. Has anybody experienced this with the IPKall service? are they not
passing the DTMF tones or am I doing something wrong? I've tried switching
dtmfmode to all the options, but still nothing. Thanks for your
help!
- Joshua Colp.
Greetings,
It appears you are correct, as a test I just set it
up so when an incoming call came in it dialed Tellme and their system didn't
pick up on the DTMF tones either. I guess I will have to wait for my other phone
number to be setup.
- Joshua Colp.
Joshua,I've been looking
around your IDE hardrive (for the license), thus using a SCSI hardrive
will not work. Digium is working with the company who made the asterisk codec
however so that SCSI will work in the future.
- Joshua Colp.
- Original Message -
From:
Unavailable ID
To: [EMAIL PROTECTED
I just faxed my disclaimer into Digium through my Sipura an hour ago, it
worked fine.
- Joshua Colp.
- Original Message -
From: Nicolas Bougues [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, March 06, 2004 5:03 PM
Subject: Re: [Asterisk-Users] Sipura SPA 200 Fax
On Fri, Mar
for example).
Asterisk also provides other features such as voicemail, hold on music, call
display, etc.
- Joshua Colp.
- Original Message -
From: Reza Kordi [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 14, 2004 10:55 AM
Subject: [Asterisk-Users] ASTERISK V. SER
Hi Guys,
Can
Well since a person would normally go for usability and asterisk is
originally created for Linux, I said it was a Linux PBX Solution. I have
nothing against BSD myself, I have a FreeBSD sitting a few feet away from
me.
- Joshua Colp.
- Original Message -
From: Aaron J. Angel [EMAIL
Hello Hank,
Would using parking not work for this? You can just have the timeout number
set extremely high, or set so it never times out and when you want to return
to the call - just dial the number that asterisk read to you. Read up on
parking, it may be what you want.
- Joshua Colp
of info. Oh, be on
the watch... I may end up selling the phone when my Ciscos come.
- Joshua Colp.
/proc cat version
Linux version 2.4.10-uc2 ([EMAIL PROTECTED]) (gcc version 2.95.3 20010315
(release)) #1 Fri Mar 21 12:39:17 PST 2003
/proc cat cpuinfo
Processor : STMicro STLC1502 rev 0
I'm in a good mood so I'll respond...
It wasn't able to link against the ssl library because it was not found. You
may need to find the path to the library and add it to the ld.so.conf -
Remember, the filename should be libssl.so
- Joshua Colp.
-Original Message-
From: [EMAIL PROTECTED
*cough* not a thing.
- Joshua Colp.
- Original Message -
From: James Moran [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 29, 2004 9:46 PM
Subject: RE: [Asterisk-Users] FreeBSD
Does any of the hardware work with FreeBSD??
On Mon, 2004-03-29 at 20:25, Steven M. Sokol
.
- Joshua Colp.
- Original Message -
From: Muiz Motani [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, April 02, 2004 7:29 PM
Subject: Re: [Asterisk-Users] Seattle IAX Termination
Nufone has an 800-number service? How did you find out about that.
I have looked at NuFone's website
Hrm, Jeremy told me it was.. but oh well - just send NuFone an e-mail or
Paypal them an amount with a user/pass and they'll set up your account ASAP.
- Joshua Colp.
- Original Message -
From: Hermann Wecke [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, April 02, 2004 9:39 PM
Hello,
The Sipura is used as an FXS adapter, in that it
allows you to plug a phone into either of it's 2 lines and have a connection to
asterisk. I'm sorry to tell you that it can't be used as an FXO
adapter.
- Joshua Colp.
- Original Message -
From:
San Singhania
The cygwin has me Brian... please - I must follow the white rabbit instead!
AHHH!
- Joshua Colp.
- Original Message -
From: brian [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, May 21, 2004 6:28 PM
Subject: RE: [Asterisk-Users] Asterisk Cygwin Port.
I know who you
Nothing, it's normal to get those errors - I get them all the times I
compile asterisk on Linux, FreeBSD, and Windows. Your failure to compile is
being caused elsewhere.
- Joshua Colp.
- Original Message -
From: Julian Pawlowski [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May
.
Joshua Colp
Senior Software Developer
VoiceConduits, LLC.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nicolas
Sent: Sunday, October 17, 2004 1:59 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: can not compile chan_capi 0.3.5
Patrick wrote:
On Sun
it,
taking away time that I could have spent on other issues.
I'm going to end this email with a question myself... how many people
have Asterisk on a development/staging server before deployment, test,
and isolate the issues they may have in their specific scenario?
--
Joshua Colp
Software
James FitzGibbon wrote:
On 8/24/07, *Joshua Colp* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
I'm going to end this email with a question myself... how many people
have Asterisk on a development/staging server before deployment, test,
and isolate the issues they may have
an OPTIONS packet won't tell you that. You send a call,
they reject (and sometimes they even use a response code that doesn't indicate
it's DND). Same goes for call forwarding. You send a call, they reject saying
go here instead.
Joshua Colp
Software Developer
Digium, Inc
Hi Bruce,
It was not deleted, it was closed automatically when the commit to 1.4 to fix
it happened and then an additional note was added for the commit to trunk. If
you didn't get an email detailing this as you should have I will test and pass
it off to get fixed.
Joshua Colp
Software
that is sent out.
Joshua Colp
Software Developer
Digium, Inc.
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this
apart a bit you could perhaps directly trigger a reinvite. The better
question is why are you asking this?
Joshua Colp
Software Developer
Digium, Inc.
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thing?
Bart
Slight correction: It is NULL, not 0. Something can't be broken that was
never expected to work or coded to work... ANSWEREDTIME only gets set by
app_dial when you dial something else and it is answered or not answered.
Joshua Colp
Software Developer
Digium, Inc
SIP though you can just dial it, make sure canreinvite is set to
yes, and audio should go direct.
Joshua Colp
Software Developer
Digium, Inc.
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is [EMAIL PROTECTED].
Taking your example I would get
From: Asterisk PBX [EMAIL PROTECTED]
Envelope: [EMAIL PROTECTED]
so I guess there's something wrong here...
The voicemail email gets handed off to sendmail for actual sending. It's
adding on the envelope above.
Joshua Colp
Software Developer
and the individual who looks after that stuff will look
at it.
Thanks!
Joshua Colp
Software Developer
Digium, Inc.
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function to get that specific header and then CUT to
get the specific part you need.
Joshua Colp
Software Developer
Digium, Inc.
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.
Here is a basic entry for a user:
[myserver]
type=user
secret=password
disallow=all
allow=ulaw
context=servers
Here is the respective dial line:
IAX2/trunk-out/${EXTEN}
Joshua Colp
Software Developer
Digium, Inc.
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chan_sip versions... but for the most part the other side usually just
wants you to respond with something/anything. Is the other side unhappy
with the 404 Not Found?
Joshua Colp
Software Developer
Digium, Inc.
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talking to Sangoma support. They are extremely helpful
and should be able to answer your questions in no time, give them a ring.
Joshua Colp
Software Developer
Digium, Inc.
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?
Joshua Colp
Software Developer
Digium, Inc.
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mixed
together, 'nor should it care. The sources could have been Zaptel
channels for example in which case they couldn't be added to the list.
Joshua Colp
Software Developer
Digium, Inc.
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asterisk
T = Trunking. If it's present then trunking is enabled.
Ronaldo wrote:
Hi Anthony,
It doesn't make sense. This peer is an IAX peer. It was supposed to use
UDP.
Does Asterisk also use TCP for IAX?
Thanks
Ronaldo.
--
Joshua Colp
Software Developer
Digium, Inc
entry with
the dtmf=rfc2833 is being used.
I'll chime in since nobody has yet corrected this... it's
dtmfmode=rfc2833 not dtmf=rfc2833
--
Joshua Colp
Software Developer
Digium, Inc. - The Genuine Asterisk Experience (TM)
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chan_features.so :)
--
Joshua Colp
Software Developer
Digium, Inc. - The Genuine Asterisk Experience (TM)
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: In function `pri_dchannel':
chan_zap.c:9292: structure has no member named `call'
make[1]: *** [chan_zap.o] Error 1
make: *** [channels] Error 2
You need to download and install the latest libpri first.
--
Joshua Colp
Software Developer
Digium, Inc. - The Genuine Asterisk Experience (TM
?
In a perfect world maybe that would happen but this is a simple PBX
running on Linux.
Joshua Colp
Software Developer
Digium, Inc.
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would be 2. Is this what you are looking for?
Joshua Colp
Software Developer
Digium, Inc.
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functionality.
Hopefully this enlightens you a bit.
Joshua Colp
Software Developer
Digium, Inc.
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.
Joshua Colp
Software Developer
Digium, Inc.
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a method of
better determining things please feel free to share it.
Cheers,
Arik
Joshua Colp
Software Developer
Digium, Inc.
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the mappings of SIP response code - Q.931 are hard coded in chan_sip though
so that is where you can find what maps to what.
--
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
you are using 1.6.0 I will make some time to create a branch with the
changes in it based off of 1.6.0 so
I can get further testing.
--
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
Extension: 2
priority:1
And upon further examination... don't put T38CALL in as a variable. It will
cause the initial INVITE to only
have T38. Leave it out and things should hopefully reinvite.
--
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806
- Benny Amorsen benny+use...@amorsen.dk wrote:
Joshua Colp jc...@digium.com writes:
This was filed as an issue and is being tracked at
http://bugs.digium.com/view.php?id=12437. Thus far
I have created a branch for Asterisk 1.4 that changes the behavior
to accept the incoming INVITE
it a test it is already available in 1.6.2 and the
documentation for it available by
typing core show application ConfBridge.
--
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
or linksys. The
control address is only needed for the linksys type.
Any feedback is welcome as a note on
https://issues.asterisk.org/view.php?id=11797 and will help to getting this
into the tree.
Thanks!
--
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
The one you tell it to. It's configured in whatever technology your call
comes in on (SIP/IAX2/Zap).
--
Joshua Colp
Software Developer
Digium, Inc.
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Rushowr wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Actually, isn't there SLA work being done in the trunk right now?
It doesn't work how you think it does, you can still only have 1 SIP
device registered to a SIP peer at a time.
--
Joshua Colp
Software Developer
Digium, Inc
?
Thank you in advance,
Juanjo
This is a debug message and should only appear if the debug level is 3
or above I believe.
--
Joshua Colp
Software Developer
Digium, Inc.
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.
ie:
Dial(SIP/145_1SIP/145_2SIP/145_3)
3 phones would each be registered on the machine as 145_1, 145_2, and 145_3.
The first one to pick up would get the call and all the rest would stop
ringing.
--
Joshua Colp
Software Developer
Digium, Inc
Peder @ NetworkOblivion wrote:
How does it work?
Joshua Colp wrote:
Rushowr wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Actually, isn't there SLA work being done in the trunk right now?
It doesn't work how you think it does, you can still only have 1 SIP
device registered
]: *** [chan_zap.o] Error 1
make[1]: Leaving directory `/home/voipuser/asterisk-1.2.11/channels'
make: *** [subdirs] Error 1
How can I fix it?
gc
--
Joshua Colp
Software Developer
Digium, Inc.
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Do you have a backtrace so we can see where it crashed and have you reported a bug with the backtrace?JoshuaColpSoftwareDeveloperDigium,Inc.- Original Message -From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Monday, September 18, 2006 11:45:14 AM GMT-0800Subject:
the 1.4 branch
instead and try it if you are interested. (Any of you).
--
Joshua Colp
Software Developer
Digium, Inc.
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, or on the Dial line) then the remote
Asterisk box will guess who you want to authenticate as which may be
incorrect. This will cause an authentication failure.
--
Joshua Colp
Software Developer
Digium, Inc.
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.
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happening as well!
Did you file a bug report about this or try the 1.2 branch? I know I may
have fixed this issue, and I take spy related bugs quite seriously these
days and try to get them fixed asap.
Joshua Colp
Software Developer
Digium, Inc
guess that isn't true any more. I'm taking a
couple of mac's
with me.
Okay folks, give the latest 1.4 branch a try. I spent some time this
morning isolating the issue and think I have it.
--
Joshua Colp
Software Developer
Digium, Inc
individual saying it just hung for
him but it didn't for me... quite odd.
--
Joshua Colp
Software Developer
Digium, Inc.
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) to 256
Translated Error:
I cant convert from ULAW to G729, fool!
Joshua Colp.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kurt turner
Sent: Wednesday, August 17, 2005
5:46 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Comfort
Noise incomplete
Hello,
They are being rejected because the extensions (your DIDs) do not exist in
the context from-pstn. How did I know? I read the error ;)
- Joshua Colp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick M.
Gray, Jr.
Sent: Saturday, April 30
Someone stop me I'm replying to posts again...
Anyway, your preferences are setup to prefer g729 over ulaw, and the other
end offered g729... so it was used first. Thus, change your order in
iax.conf so ulaw is first and it will magically start working magically!
- Joshua Colp.
(file
the time, there's
no real way to do this... if you want to REALLY make it happen modify the
source code...
- Joshua Colp.
(file in #asterisk on Freenode)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Tuesday, June 07, 2005 7:49 PM
How are you using DISA? What protocol? If it's SIP, do you have the dtmfmode
option properly set?
- Joshua Colp.
(file in #asterisk on Freenode)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Madeira
Sent: Tuesday, June 07, 2005 6:25 PM
and agreements,
you could get the original number that was dialed sent as another SIP header
along with other information... But that's likely not going to happen.
- Joshua Colp.
(file in #asterisk on Freenode)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mirko
for it.
Note that if the Nortel is incapable of handling a challenge for
credentials, you'll have to use a peer entry with insecure=very to match
based on it's host/IP address.
- Joshua Colp.
(file in #asterisk on Freenode)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
I'd like to thank you for making this, knowing that someone else out there
will probably use it and not give you the recognition that you deserve. Keep
up the great work and thanks for contributing to Asterisk!
- Joshua Colp.
(file in #asterisk on Freenode)
-Original Message-
From
A network booboo occurred and and just like it warns (note the word
WARNING), it received a mini frame before the first full voice frame...
Nothing too serious, audio might sound odd for less then a second but it
should recover.
- Joshua Colp.
(file in #asterisk on Freenode)
-Original
/bleh)
... SO ON...
Use your head to figure out some of the stuff for what you should put in.
- Joshua Colp.
(file in #asterisk on Freenode)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Denis Galvão -
iSolve
Sent: Tuesday, June 07, 2005 11:48 PM
Can you paste a sip debug by chance, some CLI output? I'd love to see what's
actually happening.
- Joshua Colp.
(file in #asterisk on Freenode)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Manousos
Sent: Tuesday, June 07, 2005 8:08 AM
to compensate with adding 30 seconds to the answered time. Instantly
changing CDR records isn't exactly what Asterisk was made to do easily.
- Joshua Colp.
(file in #asterisk on Freenode)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent
username and password authentication - to strictly match based on the IP
address of the originating SIP packet and send it to the specified context.
- Joshua Colp.
(file in #asterisk on Freenode)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Denis Galvão
One side is using G729, the other is using ULAW. Asterisk is having to
convert between the two and can not, probably because you do not have the
G729 codec with the proper license ($10/channel from Digium).
- Joshua Colp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Codecs are negotiated between asterisk and the device, not device to
device... So since you specify G729, one side negotiated to G729 first...
Then when you dialed the other device, that one negotiated at ULAW... And
then when they attempted to be bridged together - voila, failure.
- Joshua Colp
That made no sense to me. Please try again. If you mean why did it not go to
the next line when it tried to bridge it's because you can't switch codecs
in the middle of a call.
- Joshua Colp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
I don't quite know what you mean but usually the conferencing portion of the
call is actually done by the phone. If you're using a phone that is
incapable of this and want asterisk to take over, yes meetme is the only way
to do it... There's no other way to do the audio mixing easily.
- Joshua
Make sure you're not using asterisk or you will have no T.38 support, not
even passthrough.
- Joshua Colp
On 6/20/05 6:46 AM, Adam Megacz [EMAIL PROTECTED] wrote:
So, I've been able to receive faxes quite reliably through teliax with
g711 so far; I think I can live with it.
For outbound
I was just avoiding a potential nightmare when you tried to use a T.38
capable ATA to your provider through asterisk, and wondered why it didn't
work.
- Joshua Colp.
On 6/21/05 3:25 AM, Adam Megacz [EMAIL PROTECTED] wrote:
Joshua Colp [EMAIL PROTECTED] writes:
Make sure you're not using
Rich is indeed correct, Asterisk does not yet support multiple registrations
for a single peer entry. Thus when you register the previous registration is
discarded and the new one is used. Thus like he said, the last one that
registered gets the call.
- Joshua Colp.
On 6/21/05 9:39 AM, Rich
failed to
patch.
- Joshua Colp.
On 6/21/05 10:00 AM, harry gaillac [EMAIL PROTECTED] wrote:
hello,
I need help with ast_data
I downloaded asterisk from cvs
cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co
-r HEAD asterisk
and the latest ast_data.
When i run ./INSTALL.txt i get
your hints needs to
be accessible to the SIP phone.
- Joshua Colp.
On 6/21/05 9:58 AM, Bjørn Ove Kristiansen [EMAIL PROTECTED] wrote:
Hello all!
First of all, thank you for all suggestions. As suggested, FOP does show
who's online, but it's not really what I'm looking for. As said before
Multiple entries in sip.conf, with a macro specifying multiple places to
call for extensions... That's what I do.
- Joshua Colp.
On 6/21/05 10:29 AM, Anton Krall [EMAIL PROTECTED] wrote:
In environments where users have their hard and soft phones... How do you
glue everything together
My client (Entourage) did a word wrap... Couldn't fit it all on one line.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20extension
Try that ^^^
- Joshua Colp.
On 6/21/05 11:04 AM, Anton Krall [EMAIL PROTECTED] wrote:
Page cannot be found
|-Original Message
...
It's just that in the Asterisk world everything is designed with a one
device per peer concept.
- Joshua Colp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Tuesday, June 21, 2005 2:18 PM
To: 'Asterisk Users Mailing List - Non
It depends really. There's about 3 different ways to send DTMF with SIP. One
is inband, as audio. Another is rfc2833, which is not as audio - but still
goes via the RTP stream as separate packets. The last one is info, which
sends it over the control stream as SIP packets.
- Joshua Colp.
On 6
You can reload most anything individually, despite not having a CLI command.
You just need to execute reload filename. Example: reload chan_iax2.so
That would reload IAX2... Yay!
- Joshua Colp.
On 6/25/05 1:23 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Jun 23, 2005 at 02:57:36PM +0200
In CVS you can reload chan_zap, but not totally... I believe you can't
change the signalling type without restarting asterisk.
- Joshua Colp.
On 6/25/05 3:59 PM, Rene Ott [EMAIL PROTECTED] wrote:
I tried to reload chan_zap.so but it didn't work. Do you know a way how to
reload it?
René
Matt - catch me on IRC (it's file).
- Joshua Colp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Sunday, June 26, 2005 6:30 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk
Title: Re: [Asterisk-Users] [Fwd: Asterisk Balancing solution]
Maybe... Wait for it... Realtime? Keep the information in a database that is shared or replicated between all servers.
- Joshua Colp.
On 6/30/05 10:54 AM, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote:
Dear All,
I am using
You do realize you're not sending along a username so it's using another method
to try to discover the username you're trying to authenticate as on the server
side? Apparently not.
IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]
Joshua Colp
- Original Message -
From: Douglas Garstang
[mailto
It still needs to know the username so it knows what entry in iax.conf to use
for that information, such as the key to use.
Joshua Colp
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL
to reload from the astdb?
Joshua Colp
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