- Original Message -
From: Ex Vito
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, November 12, 2009 3:59 PM
Subject: Re: [asterisk-users] BLF with SPA941?
Although I've never tested such feature on those devices, I know
that it was only
hi all,
i had installed and configured asterisk on centos 5.3, i had
made a minimum dial plan in which i had made two extentions. i am easily
able to make call from one extention to other extention. i know its just a
basic thing which i had done n i had done from this place only. now i want
to
I think just renaming the [default] to [public] or [unautorized], and a comment
saying
Don't put outgoing calls in this context, as unauthorized users, even from
outside, are routed here by default.
would be enough.
I'm not sure if local phones should automatically be routed to a [local]
- Original Message -
From: aster...@opensourcesolution.in
To: asterisk-users@lists.digium.com
Sent: Friday, November 13, 2009 9:47 AM
Subject: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE
hi all,
i had installed and configured asterisk on centos 5.3, i had
Is there any documentation on the CallWaitingRing?
Thanks
Dan
-Original Message-
From: Danny Nicholas da...@debsinc.com
Sent: 12 November 2009 14:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Incoming Call
On 13 Nov 2009, at 08:47, aster...@opensourcesolution.in
aster...@opensourcesolution.in
wrote:
I had installed and configured asterisk on centos 5.3, i had made a
minimum dial plan in which i had made two extentions. i am easily
able to make call from one extention to other extention.
Hi Everybody,
Sorry for my bad english I'm french!
I need your help!
I installed asterisk1.6.1 with update 1.6.9, Dahdi for musiconhold and chatroom.
I created sip users, and a IVR.
I can use my webradio to put in musiconold.
That's good but the music is cutted.
I would like to know how can I
Martin,
This Grandstream HT503 makes it possible for me to send incoming
PSTN-calls to an Asterisk-server on the local network (for IVR) ???
How about voice quality ? (the negative point of Linksys SPA)
Best regards,
Jonas.
On Thu, 2009-11-12 at 18:59 -0500, Martin wrote:
Grandstream
I have just established a call between 2 sip phones and I have noticed
that all RTP traffic goes through Asterisk Server.
I was expecting RTP traffic went to one phone to another phone directly.
I set canreinvite=yes in sip.conf in both sip peers.
I also tested it with 2 mgcp phones and same
On Fri, Nov 13, 2009 at 11:31 AM, Manu et...@manu-dpk.net wrote:
Can you help me please?
Thank you very much.
Voici un meilleur site pour poser des questions de tout genre en français :
http://asterisk-france.net/
/r
___
-- Bandwidth and Colocation
Tilghman Lesher tles...@digium.com writes:
Many consumer-grade switches effectively turn into hubs when more than 1023
MAC addresses are seen on a network. This may be done intentionally by
somebody attempting to eavesdrop on all network connections sent through
the switch. A reboot of the
Cary Fitch ca...@usawide.net writes:
Is there a plain 64K codec that would simply pass through the SIP server and
be handed off to a PRI or phone co. trunk on a T1 on the other side of the
SIP server? Digital 64K telco sounds very good as a phone conversation.
You can't get a guaranteed
Hi,
I have two Asterisk systems that hang once every 4-6 days (more or less). One
has * 1.2.31.1 and the other 1.4.26.2. The last system collapsed today and I
saw several messages looping endlessly on screen:
hfcmulti_rx no memory for rx_skb
alloc_stack_skb(303,110): no skb size
During this
I basically agree, but I couldn't resist:
On Fri, Nov 13, 2009 at 09:51:59AM +0100, Leif Neland wrote:
Why should my call (and my money) go from my desk via my ip-pabc to
my voisp possibly through pstn (through echelon) to your voisp to
your ip-pabc to your desk, when it could go from my
Does the phone have some sort of NAT Keepalive setting? Often, the only
way to keep that port open on the user's NAT gateway is to have the
NATted client send the occasional data out through the port.
N.
Ron wrote:
i have also tried setting qualify='yes' but cpu usage spiked to 100%.
Ron
hi sir,
yes i am using Linksys SPA's i set NAT Mapping enable and NAT Keep-live
to Yes. still sometimes the phone cannot be reach even though it is
registered.
regards
ron
SIP wrote:
Does the phone have some sort of NAT Keepalive setting? Often, the only
way to keep that port open on the
On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote:
I have just established a call between 2 sip phones and I have noticed
that all RTP traffic goes through Asterisk Server.
I was expecting RTP traffic went to one phone to another phone directly.
I set canreinvite=yes in sip.conf in both sip
Please stop emailing me personally.
If no one replies to a post, it means that everyone is busy or they think you
should read through the documentation before posting.
If you can't figure out simple things like Music on hold from the
documentation, then i dont think VOIP is for you.
On Fri, 13 Nov 2009, aster...@opensourcesolution.in wrote:
i had installed and configured asterisk on centos 5.3, i had made a
minimum dial plan in which i had made two extentions. i am easily able
to make call from one extention to other extention. i know its just a
basic thing which i
This can also be caused by IRQ conflicts. You could try a different slot
to see if it clears it up
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir
Cohen
Sent: Thursday, November 12, 2009 1:44 PM
To:
Its not just you mate. He's doing it to everyone, and sadly the list
server is too clever to accept forged unsubscribes..
Steve
On 13 Nov 2009, at 15:22, Dan Journo wrote:
Please stop emailing me personally.
If no one replies to a post, it means that everyone is busy or they
think
Sorry, I can't resist.
How do I join the Mail List Nazi Corp? Do I have to be invited, or can I
just self appoint myself? Asking neophyte questions are objected to by
some, top posting by those who blast others, etc.
How about leaving member chastisement to the sponsor of the list?
Some
This is happening here also :(
On Fri, Nov 13, 2009 at 9:02 PM, Cary Fitch ca...@usawide.net wrote:
Sorry, I can't resist.
How do I join the Mail List Nazi Corp? Do I have to be invited, or can I
just self appoint myself? Asking neophyte questions are objected to by
some, top posting by
On 13 Nov 2009, at 16:02, Cary Fitch wrote:
Sorry, I can't resist.
Evidently
How do I join the Mail List Nazi Corp? Do I have to be invited, or
can I
just self appoint myself? Asking neophyte questions are objected to
by
some, top posting by those who blast others, etc.
You just
If you missed @voicegal last time or didn't go to Astricon, join us
today on the Voip Users Conference to meet Allison Smith, the voice of
Asterisk.
Or go listen to the FBI talk about security...
http://VoipUsersConference.org for details.
/r
___
--
What does the zombie call look like in core show channels?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aggio Alberto
Sent: Friday, November 13, 2009 10:33 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Hi all,
Some time ago I posted an issue regarding the hangup of active calls from the
CLI and someone told me that soft hangup should work. Well, in fact it does
work, but only if the channel is known, i.e. it doesn't work for zombie
channels. For example, I have this scenario (CLI output of
Sorry, I can't resist.
How do I join the Mail List Nazi Corp? Do I have to be invited, or can I
just self appoint myself? Asking neophyte questions are objected to by
some, top posting by those who blast others, etc.
How about leaving member chastisement to the sponsor of the list?
As is occasionally pointed out by discerning people, questions are not
dumb because they are posed by newbies, or because they reflect a
lack of familiarity or knowledge of the product.
Questions are dumb when they are formulated in a manner consistent
with extreme intellectual laziness,
On Thu, 2009-11-12 at 20:18 -0700, Joseph wrote:
Digium has discontinued their ATA iaxy adapter; don't blame them, too
expensive so they can not compete.
Compete, With which iax-ata ???
___
-- Bandwidth and Colocation Provided by
How do you know a good soccer player when you see one? If you are a good
scout, just by his body language. Just by seeing him how he walks and
positions himself on a field. By the time he touches the ball, he is either
eliminated from my list of prospects or he is marked as good to be
On Thu, 2009-11-12 at 18:59 -0500, Martin wrote:
Grandstream HT503. For me works just fine. 1xFXO 1xFXS port. Each port
has its own sip account.
Martin
- Original Message -
From: jonas kellens
To: Asterisk Users Mailing List - Non-Commercial Discussion
Just picked up Asterisk the Future of Telephony, every other listed
program is there (Book does not tell you about the changeover to
dahdi_toolname). But there is no dahdi_zttools. I have dahdi-tools
installed, tried to install via yum and says it is already installed.
Thanks
Just picked up Asterisk the Future of Telephony, every other listed
program is there (Book does not tell you about the changeover to
dahdi_toolname). But there is no dahdi_zttools. I have dahdi-tools
installed, tried to install via yum and says it is already installed.
Thanks
I received this with a Sangoma card and CentOS 5.4. Downgrading to 5.2
resolved the issue.
--
James Texter III
Sr. Software Engineer
NOBLE SYSTEMS
4151 Ashford Dunwoody Road, Suite 600 | Atlanta, GA 30319-1452
(o) 404.851.1331 ext. 357
(f) 404.851.1421
(e) jtext...@noblesys.com
(w)
Quoting Jaap Winius jwin...@umrk.to:
The question remains: how can a remote Asterisk server be receiving
SIP packets that still contain the private net IP address of a client?
Okay, I fixed it: by installing siproxd on the firewall system of the
local network. With the Debian systems I'm
Humanx2000 schrieb:
Just picked up Asterisk the Future of Telephony, every other listed
program is there (Book does not tell you about the changeover to
dahdi_toolname). But there is no dahdi_zttools. I have dahdi-tools
installed, tried to install via yum and says it is already installed.
Joe Greco wrote:
Sorry, I can't resist.
How do I join the Mail List Nazi Corp? Do I have to be invited, or can I
just self appoint myself? Asking neophyte questions are objected to by
some, top posting by those who blast others, etc.
How about leaving member chastisement to the
OK, I know it's only just out today but this is what I get when
compiling dahdi-linux.
make -C drivers/dahdi/firmware firmware-loaders
make[1]: Entering directory
`/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware'
make[1]: Leaving directory
`/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware'
At 03:53 PM 11/12/2009, you wrote:
I Have a home line connected to a tdm400p with 3 extensions and a
siemens sip-dect , it seems to work fine but during a call there is
always a digital squeal every so often does anyone know what this could be?
If it sounds like the tones you get when you press
On 11/11/2009 02:08 PM, Ex Vito wrote:
We've been experiencing some tough time regarding a new Asterisk
installation
connected to the PSTN via an ISDN PRI with a Digium TE121 with the optional
VPMADT032 echo cancellation module.
It appears there may be a regression in dahdi-linux 2.2.0
Slightly paraphrasing a very old and wise saying:
Give a man a fish,
he eats for a day.
Teach him how to fish,
he eats for a lifetime.
--
JohnM
I see no teaching, just no help.
He doesn't eat today or tomorrow either.
Cary Fitch
___
--
On 11/13/2009 01:11 PM, Dave Cotton wrote:
OK, I know it's only just out today but this is what I get when
compiling dahdi-linux.
make -C drivers/dahdi/firmware firmware-loaders
make[1]: Entering directory
`/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware'
make[1]: Leaving directory
On Fri, Nov 13, 2009 at 1:06 PM, John Millican
jmilli...@sentinelcommunications.com wrote:
Slightly paraphrasing a very old and wise saying:
Give a man a fish,
he eats for a day.
Teach him how to fish,
he eats for a lifetime.
I've always liked...
Build a man a fire
stays warm for a day
What say you to the proposal that some approaches to seeking help are
so ridiculous they should not be tolerated?
Community standards neither conceive nor enforce themselves.
--
Sent from mobile device
On Nov 13, 2009, at 2:32 PM, Cary Fitch ca...@usawide.net wrote:
Slightly paraphrasing a
Hello all,
How can I ask Asterisk to ignore a sip hang-up request for XX seconds from
the beginning of the session?
Thank you
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To
On Fri, 13 Nov 2009, Jon Moore wrote:
I've always liked...
Build a man a fire
stays warm for a day
Catch a man on fire
stays warm the rest of his life.
Funniest thing I've read all day :)
--
Thanks in advance,
-
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi List,
What I hope is a simple question...
As the subject states, I would like to know if anyone has setup a
Multi Tenant Asterisk Server ?
If so, what would I need to do to get to a Multi Tenant setup
(preferably an Open Source solution) ?
Any
I added some examples a while back to the extensions.conf.sample and the
voicemail.conf.sample code to show how to support distinct domains for voice
mail contexts... which was a big obstacle to multi-tenancy... otherwise, you
couldn't have individual greetings, etc.
For places (like
On 13/11/09 20:42, Shaun Ruffell wrote:
The easiest thing to do is comment out the following line in
drivers/dahdi/Kbuild.
obj-$(DAHDI_BUILD_ALL)$(CONFIG_DAHDI_WCTC4XXP) += wctc4xxp/
Or you can grab the head of the 2.2 branch or trunk which has all the
build issues for recent
snip
What say you to the proposal that some approaches to seeking help are
so ridiculous they should not be tolerated?
/snip
I say give me a break.
Pre-judging people doesn't work on mailing lists given the inherent language
barriers, etc.
___
--
David Gibbons wrote:
snip
What say you to the proposal that some approaches to seeking help are
so ridiculous they should not be tolerated?
/snip
I say give me a break.
Pre-judging people doesn't work on mailing lists given the
inherent language barriers, etc.
It can. There are
What say you to the proposal that some approaches to seeking help are
so ridiculous they should not be tolerated?
Community standards neither conceive nor enforce themselves.
This community standard is entirely self-enforcing.
If everybody thinks the request for help is unwarranted and
On Fri, 13 Nov 2009, Cary Fitch wrote:
Sorry, I can't resist.
Next time please try harder.
How do I join the Mail List Nazi Corp?
And of course, lacking any sense of history, let's blame it on the
Nazis.
Asking neophyte questions are objected to by some, top posting by those
who blast
On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi List,
What I hope is a simple question...
As the subject states, I would like to know if anyone has setup a
Multi Tenant Asterisk Server ?
If so, what would I need to do to get
I strongly concur with this realistic and very well-researched thesis.
--
Sent from mobile device
On Nov 13, 2009, at 5:59 PM, Steve Edwards asterisk@sedwards.com
wrote:
On Fri, 13 Nov 2009, Cary Fitch wrote:
Sorry, I can't resist.
Next time please try harder.
How do I join the
My point was the two previous posters could have ignored the request and
made no post at all. That they were violating a rule by top posting to
tell a person not to bug them.
And, someone criticized me for an off topic post and of course there have
been 15-20 more. And some have top posted and
On 17:54, Fri 13 Nov 09, John A. Sullivan III wrote:
On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi List,
What I hope is a simple question...
As the subject states, I would like to know if anyone has setup a
Multi
On Nov 13, 2009, at 6:16 PM, Cary Fitch wrote:
My point was the two previous posters could have ignored the request and
made no post at all. That they were violating a rule by top posting to
tell a person not to bug them.
And, someone criticized me for an off topic post and of course
On Sat, 2009-11-14 at 00:30 +0100, Michiel van Baak wrote:
On 17:54, Fri 13 Nov 09, John A. Sullivan III wrote:
On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi List,
What I hope is a simple question...
As the
Hi,
I have upgraded an Asterisk installation with a Xorcom BRI Astribank
that was working under Ubuntu 8.04 to Ubuntu 9.10 and the device is no
longer initialized.
When I reload the udev rules, I see that the rules seems to be correctly
loaded:
udevd[452]: reading
On 18:55, Fri 13 Nov 09, John A. Sullivan III wrote:
On Sat, 2009-11-14 at 00:30 +0100, Michiel van Baak wrote:
On 17:54, Fri 13 Nov 09, John A. Sullivan III wrote:
On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
Dear All
Can you please do me favor and let me know how can I stop my Asterisk ? Can
you please confirm if the following procedure is correct to stop it ?
#/etc/init.d/asterisk stop
#cd /etc/init.d
#chmod asterisk
Let me thank you in advance
___
--
cli stop now
or
cli stop gracefully
:)
otherwise
pkill -9 asterisk
On Sat, Nov 14, 2009 at 7:39 AM, hadi motamedi motamed...@gmail.com wrote:
Dear All
Can you please do me favor and let me know how can I stop my Asterisk ? Can
you please confirm if the following procedure is correct to stop
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