4 jan 2010 kl. 09.34 skrev Remco Barendse:
Is there any fix or workaround for the DNS problem (old standing bug that
when the box starts and domain names do not resolve quickly enough from
DNS then asterisk stops using the outgoing trunks.
I read on the list before that it is considered
4 jan 2010 kl. 14.46 skrev Kevin P. Fleming:
hadi motamedi wrote:
Sorry . I didn't get the point clearly . In the SIP Invite message , it
says my audio endpoint is IP x.x.x.x port x, and I can use codecs
A,B,C. The remote endpoint responds with a 200 OK, saying my audio
stream is at IP
On Tue, Jan 5, 2010 at 8:59 AM, Olle E. Johansson o...@edvina.net wrote:
4 jan 2010 kl. 14.46 skrev Kevin P. Fleming:
hadi motamedi wrote:
Sorry . I didn't get the point clearly . In the SIP Invite message , it
says my audio endpoint is IP x.x.x.x port x, and I can use codecs
A,B,C.
It seems dahdi is needed for meetme, but not available under FreeBSD.
So what do I do then?
Asterisk has only SIP-channels.
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asterisk-users mailing list
To UNSUBSCRIBE or update
Hi,
I have a difficulty on my Asterisk's database.How can I get the info
about list of ringing agents on my queue
In console :
-- Started music on hold, class 'default', on DAHDI/77-1
*-- SIP/6002-00cc0f90 is ringing
-- SIP/6004-00c23270 is ringing
-- SIP/6005-00be6220 is ringing*
Hi,
That model HP or Dell server that I recommend for a TE412P card for about 200
users?
Thank you very much.
_
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On Monday 04 January 2010 07:16:49 Joseph L. Casale wrote:
Looking at the source in the rpms from the asterisk package site
appears that oslec is not built and enabled for the kmod rpms.
Anyone know an existing repo or have direction on how to enable
this to built for those rpms?
I build
Another day, another error..
Am now getting:
Plugin zaptel.so loaded.
Zaptel Plugin Initialized
Using zaptel device 'stdin'
Zaptel device is 'stdin'
Unable to put device 'stdin' into HDLC mode
Should ZapRAS see the channel as stdin and not /dev/zap/x?
Will
On 4 Jan 2010, at 16:46, Will
I've tried several different qualify settings (including 10), but it didn't
change the situation much :(.
Regards, Alex
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, January
I build them for the kernels I use (Fedora 12 x86_64 and i686.PAE) but
you can check http://messinet.com/trac/rpms/ and checkout the svn-build-rpm
tool to build from an svn checkout if you already have a build setup
configured.
Anthony,
I appreciate the pointer, and I do have a build environment
Hi all,
I've updated Asterisk trunk LDAP schema [0] [1] to include queues and other
attributes needed for a working LDAP backend (I'll open a bug to include these
changes on svn).
SIP users and dialplan are perfectly working, but when I call a queue the
whole Asterisk (1.6.2.0) crashes:
on
Olle E. Johansson wrote:
But it's fairly common to have asymmetric media in the call. If the caller
offers A, B and C and the callee responds with B, the caller sends B but the
callee might send A.
Only for non-Asterisk endpoints, since Asterisk will never do this.
Is this really that
Hi,
I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing Unknown when there is an incoming call. I think the
same problem listed here: https://issues.asterisk.org/view.php?id=6683
Hello,
can the Asterisk API be used to automate a MITEL 5330 telephone?
If not, are there any other API which can used to do that?
Many thanks.
phiroc
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asterisk-users mailing
When it is running, nerdvittles.com is an excellent resource for this kind
of question. Voip-info.org is almost always up and has more technically
oriented answers to this type of query.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi,
I tried again getting DTMF detection on my ISDN devices with dahdi going
again. I used the channel debug to see whether asterisk sees the frames
and detects them as DTMF.
Interestingly here's what works:
1. GSM phone - chan_dahdi g1 - asterisk - can_sip - SIP phone
Both the GSM phone and
On Mon, Jan 4, 2010 at 6:44 PM, Karl Fife karlf...@gmail.com wrote:
Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x
instance AND do it reliably? If so, I can think of a number of locations
with copper loops that could be scrapped. I'm actually quite surprised at
what
On Tuesday 05 January 2010 04:54:52 Asterisk wrote:
I've tried several different qualify settings (including 10), but it didn't
change the situation much :(.
Realize that qualify=10 is 10ms, not 10 seconds. You probably want
something on the order of qualify=3000.
--
Tilghman Lesher
Digium,
Hope I'm not the only one who doesn't know this; is the time value MS
across the board?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Tuesday, January 05, 2010 9:37 AM
To: Asterisk
On Tuesday 05 January 2010 09:50:42 Danny Nicholas wrote:
Tilghman Lesher wrote:
On Tuesday 05 January 2010 04:54:52 Asterisk wrote:
I've tried several different qualify settings (including 10), but it
didn't change the situation much :(.
Realize that qualify=10 is 10ms, not 10
5 jan 2010 kl. 10.08 skrev hadi motamedi:
On Tue, Jan 5, 2010 at 8:59 AM, Olle E. Johansson o...@edvina.net wrote:
4 jan 2010 kl. 14.46 skrev Kevin P. Fleming:
hadi motamedi wrote:
Sorry . I didn't get the point clearly . In the SIP Invite message , it
says my audio endpoint is
I build them for the kernels I use (Fedora 12 x86_64 and i686.PAE) but
you can check http://messinet.com/trac/rpms/ and checkout the svn-build-rpm
tool to build from an svn checkout if you already have a build setup
configured.
Anthony,
So this script builds them with the dahdi-tools-libs
hello,
we have been using a couple of US based
VoIP providers for outbound calls completed
within the US, without any issues.
We recently started making calls to Canada
and have received a few complaints about
the call quality.
Questions :
- Could this be because of the number of
Sadly I suspect you're right. I suspect the other business problem would be
abuse. Anyone in that business would doubtless get their hands dirty trying
to combat T.38 subscribers whose intention is to send Junk Faxes.
Flat-roof repair! Employee vacation discounts! Health insurance for small
Hello All -
I've been poking around the past few weeks, trying to familiarize
myself with all of this. I am new to Linux, VoIP and Asterisk -- to
be complete. This is my first exposure to all of these technologies.
I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge
2400 and
On Mon, Jan 04, 2010 at 01:16:49PM +, Joseph L. Casale wrote:
Looking at the source in the rpms from the asterisk package site
appears that oslec is not built and enabled for the kmod rpms.
Anyone know an existing repo or have direction on how to enable
this to built for those rpms?
On Tue, Jan 5, 2010 at 8:53 PM, UIT DEVELOPMENT uit...@gmail.com wrote:
I've been poking around the past few weeks, trying to familiarize
myself with all of this. I am new to Linux, VoIP and Asterisk -- to
be complete. This is my first exposure to all of these technologies.
I think one of
UIT DEVELOPMENT wrote:
Sorry for what might seem as really silly questions, but I am not sure
how to proceed.
Thanks in advance for any insight that you folks can provide!
Hello Mike.
Welcome to the wonderful world of Asterisk. Before you sludge through a
GUI and all the attendant bad
I can't help you two much with configuration of linux, but as to the call
question. You will need some route for the server to be capable of
sending/receiving calls. There is a couple of ways to do this cheaply.
Buy a standard telephone modem (usb, pci, or serial). And plug into wall a
jack.
On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:
CAN I make call from that box to my cell phone using a soft-phone? If
so, how can I do that?
You need to get an account with a VOIP provider -- someone to accept your
call via the Internet and place a call on the PSTN to call your cell
number --
Steve Edwards wrote:
On Sun, 3 Jan 2010, Steve Edwards wrote:
You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple
addresses and ports and forward to Asterisk on the same or
different boxes.
On Mon, 4 Jan 2010, Vikram Ragukumar wrote:
Would it be more efficient to use
Will do Barry. Thanks for the links! Downloading now.. Mike
On Tue, Jan 5, 2010 at 3:25 PM, Barry L. Kline blkl...@attglobal.net wrote:
UIT DEVELOPMENT wrote:
Sorry for what might seem as really silly questions, but I am not sure
how to proceed.
Thanks in advance for any
Thanks Randy!
On Tue, Jan 5, 2010 at 3:25 PM, Randy R randulo2...@gmail.com wrote:
On Tue, Jan 5, 2010 at 8:53 PM, UIT DEVELOPMENT uit...@gmail.com wrote:
I've been poking around the past few weeks, trying to familiarize
myself with all of this. I am new to Linux, VoIP and Asterisk -- to
be
James,
Thank you for the reply. I do not have phone service in my home.
I've been 100% cell since 2003. I do have an old analog phone - big
heavy thing... If I connect it to the wall outlet there is nothing.
I've tried every outlet in the house. I didnt expect to find a tone
as we've never
On Tue, Jan 5, 2010 at 12:20 AM, Trevor Benson tben...@a-1networks.com wrote:
Its called speechbackground. from asterisk console type 'core show
applications speech' (and hit the tab key) these are the speech applications
used. Speechbackground being similar to background.
Thanks, Trevor,
There are some free-trial and low-cost services out there. Gizmo comes to
mind but buyer beware; look through this site for recommendations and
warnings.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT
Steve-
Got an iPhone but no SIP client that I am aware of. I just make
regular calls to other others/receive calls as usual. Nothing fancy.
I was hoping to create the fancy stuff in my home here.
As I got to reading I began to see things like provider, as you've
said here, and unfortunately
On Tue, 5 Jan 2010, Vikram Ragukumar wrote:
If Kamailio is setup to listen on ports 5060 and 9090, port 9090 carries
unknown SIP signaling information. Is it possible for Kamailio to dump
these unrecognized signaling packets to a user space application which
would process and return
Thank you Danny. I shall investigate that.
On Tue, Jan 5, 2010 at 3:57 PM, Danny Nicholas da...@debsinc.com wrote:
There are some free-trial and low-cost services out there. Gizmo comes to
mind but buyer beware; look through this site for recommendations and
warnings.
-Original
On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:
I've got a lot of old hardware laying around and I do have MODEMs -
internal and external 56k types.
None of your externals will be of any use and I suspect you will spend
more time than it is worth trying to get any of your internals working. A
On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:
As I got to reading I began to see things like provider, as you've
said here, and unfortunately if that is the only way then I will have
to stop here as I do not have funds to further this little experiment.
Really? A buck-fifty a month is going to
Going along the internet between us and canada doesn't add much
distance, but bouncing back and forth between east and west coast
does.
On Tue, Jan 5, 2010 at 11:25 AM, Max McGraw max.mcg...@gmail.com wrote:
hello,
we have been using a couple of US based
VoIP providers for outbound calls
Gotcha on the MODEMs.. thanks.
On Tue, Jan 5, 2010 at 4:12 PM, Steve Edwards asterisk@sedwards.com wrote:
On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:
I've got a lot of old hardware laying around and I do have MODEMs -
internal and external 56k types.
None of your externals will be
Kyle Kienapfel wrote:
Going along the internet between us and canada doesn't add much
distance, but bouncing back and forth between east and west coast
does.
I'm not so sure that is the case, what I do know is both Rogers and Shaw
can never seem to fix complaint issues with voip unless
Hi,
I have installed Asterisk with iaxmodem to send faxes with Hylafax.
But I have problems to send some faxes because the receiver does not accept
speech. I must send the faxes as 3,1 kHz Audio
But I do not find a possibility to do this. I need urgent help!
Chris
Could use the free http://www.sipgate.com/one for some testing (assuming that
Asterisk is connected to the Internet)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT DEVELOPMENT
Sent: Tuesday, January 05,
git clone http://git.tzafrir.org.il/git/dahdi-extra.git
cd dahdi-extra
make gen-patch
And use the generated dahdi_linux_extra.diff . It includes OSLEC and
some other things. See the Makefile there for more information. The
patch should be applied with -p1 .
This repository includes the
Really? A buck-fifty a month is going to kill the project? I live in San
Diego, California where SDGE screws us for thirty cents a kilowatt-hour.
Yea. Sort of. I am recently unemployed. Got plenty of time on my
hands now and I am trying to not incur any more costs than I need.
How much
On Tue, Jan 5, 2010, UIT DEV wrote:
Steve-
Got an iPhone [...]
As I got to reading I began to see things like provider, as you've
said here, and unfortunately if that is the only way then I will have
to stop here as I do not have funds to further this little experiment.
I guess I
Jamie - I will check that out! Thanks! It is just for testing and
yes, the Asterisk box is connected to the Internet. Cool.
-M
On Tue, Jan 5, 2010 at 4:39 PM, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
Could use the free http://www.sipgate.com/one for some testing (assuming
Jon/Kyle,
thank you for the feedback.
I checked with someone who manages a much
higher volume of calls to Canada and he said
there are some pockets some providers that
report issues with call quality. Overall the calls
sound the same as they do in the US.
--
On Tue, Jan 5, 2010, jon
Yep. Its called unemployment. Got the iPhone a little less than a
year ago. Someone in India got my job in mid-November. I got stuck
holding the 2-year contract.Oh well. Such is life.
Look - I am going to retire from this thread. Everyone's been a
great help and I know you and
my apologies, I do understand.
sorry.
--
On Tue, Jan 5, 2010, UIT DEV wrote:
Yep. Its called unemployment. Got the iPhone a little less than a
year ago. Someone in India got my job in mid-November. I got stuck
holding the 2-year contract. Oh well. Such is life.
Look - I
Hi,
Having problems with getting either RxFax or FaxReceive
to compile. Running Asterisk 1.4 on CentOS 5.
Does anyone have the free/open source executables
that you could send me?
Thanks for your help!
P. S.: TxFax and FaxSend would also be appreciated.
Quinn Weaver wrote:
On Tue, Jan 5, 2010 at 12:20 AM, Trevor Benson tben...@a-1networks.com
wrote:
Its called speechbackground. from asterisk console type 'core show
applications speech' (and hit the tab key) these are the speech applications
used. Speechbackground being similar to
On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote:
Hi,
Having problems with getting either RxFax or FaxReceive
to compile. Running Asterisk 1.4 on CentOS 5.
What version of SpanDSP do you use?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
On Tue, Jan 05, 2010 at 09:42:48PM +, Joseph L. Casale wrote:
git clone http://git.tzafrir.org.il/git/dahdi-extra.git
cd dahdi-extra
make gen-patch
And use the generated dahdi_linux_extra.diff . It includes OSLEC and
some other things. See the Makefile there for more information.
On Tuesday 05 January 2010 12:21:15 Joseph L. Casale wrote:
So this script builds them with the dahdi-tools-libs package requirement, I
thought the fedora spec built all of these? Any idea?
Fedora packages the dahdi-tools* suff, but can't include the kernel modules.
I did not realize you
Basically - yes. It's an extra patch to add to your source RPM. Are you
familiar with modifying them?
Tzafrir,
Vaguely, I would very graciously take any suggestions you could provide:)
The whole dahdi package routine has change since the last time I used it,
was shortly Jason Parker started
On Tuesday 05 January 2010 17:09:32 Joseph L. Casale wrote:
From what I can tell so far, I can continue to use his user tools unchanged
but I need to apply this patch to the tar file in the
dahdi-linux-2.2.0.2-1_centos5.src.rpm and rebuild it, but that ,
`dahdi-linux` pulls in
atrpms.net
atrpms.net also provides packages for RHEL5, if those would work.
http://atrpms.net/dist/el5/
Just on my way to work on this server now, this would be great! That
way I don't have to work all night:) Does the atrpms ones finally do oslec?
Thanks!
jlc
Siax is a pretty good working sip and iax2 softphone for the iPhone.
Easy to connect to your own Asterisk box
If you have an Android phone (I have HTC Hero with Android 1.5) ASip
is a good choice. It is working and and calls using umts are working
surprisingly well.
Erik
They are both
At 12:48 PM 1/5/2010, you wrote:
So that was the plan but first I needed to be able to get this thing
set up. I THINK you're saying I need to purchase another service to
get myself to make calls. I dont know anyone with a SIP server..
There are services that will give you free incoming minutes
You can practice Asterisk using free SIP phones. This way you can call from
extension to extension.
SJ Phone
http://www.sjlabs.com/sjp.html
X Lite
http://www.counterpath.com/x-lite.html
From: UIT DEVELOPMENT uit...@gmail.com
To: Asterisk Users Mailing List
Thanks and no problem. There was no way you would have known. Thank
you for the info - it really is helpful and I have learned a LOT in
this thread. This is a great list with a lot of helpful folks on it!
Mike
On Tue, Jan 5, 2010 at 5:16 PM, Max McGraw max.mcg...@gmail.com wrote:
my
No Android phone. But I will read up on this anyhow. The softphone is
probably all that I need then, and of course a functioning Asterisk
setup.
On Tue, Jan 5, 2010 at 7:29 PM, meetmecall i...@meetmecall.nl wrote:
Siax is a pretty good working sip and iax2 softphone for the iPhone.
Easy to
Ah, good idea. :-) Are you saying that if I got a number that was
in my parents area code then they could be making a local call to my
Asterisk, which is physically a 1000+ miles from them? Now that is
cool.
On Tue, Jan 5, 2010 at 7:51 PM, Ira i...@extrasensory.com wrote:
At 12:48 PM
Thank you for these. I will be reading up on these sites shortly.
On Tue, Jan 5, 2010 at 7:59 PM, hin lee hi...@yahoo.com wrote:
You can practice Asterisk using free SIP phones. This way you can call
from extension to extension.
SJ Phone
http://www.sjlabs.com/sjp.html
X Lite
On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:
Are you saying that if I got a number that was in my parents area code
then they could be making a local call to my Asterisk, which is
physically a 1000+ miles from them? Now that is cool.
See
On Tuesday 05 January 2010 17:30:31 Joseph L. Casale wrote:
Just on my way to work on this server now, this would be great! That
way I don't have to work all night:) Does the atrpms ones finally do oslec?
I don't use them myself, but I was thinking that the RHEL5 spec files might be
another
Folks,
I have a Merlin Legend R7 V10.0 with a 2 100D cards.
I have 1 card in slot 4 going to CenturyTel, and the card in slot 10 going
to a flip cable to a TE110P card in a Asterisk 1.6.x box.
I have routing setup on the Merlin to send a block of numbers to the
Asterisk.
Currently the PSTN can
Thank you Steve. It is clear that I've only hit the tip of a massive
iceberg with this stuff. Its all very cool, I've got the time so I
might as well make good use of it when I am not out on interviews and
such. It is all such an interesting topic.
On Tue, Jan 5, 2010 at 9:12 PM, Steve
I don't use them myself, but I was thinking that the RHEL5 spec files might be
another place to look for what you need to build with OSLEC included, more
specifically for CentOS. I just tried taking a look at ATrpms, but the site
is having some connection issues at the moment.
How about this
Hi all,
I an using the Originate() dialplan command but I cant get it to save cdr's.
Here is the line I am using:
exten =
_61X,53,Originate(SIP/${TRUNK}/${PREFIX}${PHONE},exten,${DESTCONTEXT},${PHONE},1);
The call goes out fine, but CDR's get inserted into the DB.
Any ideas on why
On Tue, Jan 5, 2010 at 9:40 PM, Shane Brath sh...@brath.net wrote:
Folks,
I have a Merlin Legend R7 V10.0 with a 2 100D cards.
Sorry, I feel your pain.
I have 1 card in slot 4 going to CenturyTel, and the card in slot 10 going
to a flip cable to a TE110P card in a Asterisk 1.6.x box.
I
Steve Edwards wrote:
On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:
I've got a lot of old hardware laying around and I do have MODEMs -internal
and external 56k types.
None of your externals will be of any use and I suspect you will spend more
time than it is worth trying to get
Asterisk 1.4.29 or so.
access-list _dmz_acl extended permit udp 10.129.42.0 255.255.255.0 any range
1 2
access-list _dmz_acl extended permit udp 10.129.42.0 255.255.255.0 any eq
5060
But yes, all your feedback worked. I didn't need to port-forward any
incoming ports, only
Please respond.
Hi,
I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing Unknown when there is an incoming call. I think the
same problem listed here:
On Tue, Jan 5, 2010 at 5:24 AM, Arun Sasidhar
arun.sasid...@cabotsolutions.com wrote:
Hi,
I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing Unknown when there is an incoming
Jailbreak your iPhone and install Cydia to have a Unix like open
source environment (based on Debian), then install Siphon SIP client,
and have fun!
Regards.
Em 05/01/2010, às 18:04, UIT DEVELOPMENT uit...@gmail.com escreveu:
Yep. Its called unemployment. Got the iPhone a little less
On Wed, Jan 6, 2010 at 6:48 AM, Allann Jones allan...@gmail.com wrote:
Jailbreak your iPhone and install Cydia to have a Unix like open
source environment (based on Debian), then install Siphon SIP client,
and have fun!
There are at least 4 iPhone SIP clients available for $3-10 that work
well
Yes, I know - thanks. Currently I have set it to 1 (10 seconds) at which
point the problem is not that evident, but it still ocurs on a daily basis. So
I should probably look into the network, right?
Regards, Alex
-Original Message-
From: asterisk-users-boun...@lists.digium.com
2010/1/6 Asterisk aster...@abraxas.si
Yes, I know - thanks. Currently I have set it to 1 (10 seconds) at
which point the problem is not that evident, but it still ocurs on a daily
basis. So I should probably look into the network, right?
When it occurs, does it always come from the same
But jailbreaking increases the freedom to develop a application and
put on the iPhone only creating a repository for it or using a
existing repository, without the Apple Store burocracy and $$$. But
you can be right if the purpose is only to install applications that
are available on Apple
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