Regarding the DB I can't help you here, maybe someone else can.
Well. If somebody can add something on this subject, will be welcome.
Thanks for your reply.
Regards,
Daniel
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (GNU/Linux)
iEYEARECAAYFA
s if the update of Asterisk
or DAHDI is independent or the update of a component requires to also
update the other.
Thanks in advance for your reply.
Regards,
Daniel
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (GNU/Linux)
iEYEARECAAYFAku1IFAACgkQZpa/GxTmHTdZaACdGP8CAFLaGP2ek4pvdC2eHLOF
3n
roceed, does some one have any idea? Thanks
--
Daniel Abreu
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www
[400]
username=400
type=friend
secret=passwd
qualify=yes
callerid="Daniel" <400>
host=dynamic
nat=no
context=from-internal
mailbox=...@voicemail
canreinvite=no
- ---
I tried with both "nat=yes" ---as it
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi, Alyed.
On Sun, 11 Apr 2010, Alyed wrote:
>> Daniel, you are having a problem often seen in pre 1.4.14 versions.
>>
>> Before this release srvlookup=no was the default for sip.conf and
>> guess the same for iax.conf . So i
I am doing it to the capture with:
# tcpdump -i eth0 -n host 10.1.0.65 -w dump
where 10.1.0.65 is the PC with softphone.
Thanks in advance for your reply.
Regards,
Daniel
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iEYEARECAAYFAkvPpYAACgkQZpa/GxTmHTenpwCfcL3gBTTf0jRiEpv0k+j
65), the Asterisk server and a VMHost that has the
virtual machine where I use ettercap and tcpdump.
Thanks for your reply.
Regards,
Daniel
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (GNU/Linux)
iEYEAREC
a practical
demonstration of the countermeasures. Although a direct form to avoid
this is using VLANs, it seems that the idea is to demonstrate the
countermeasures with some software. Then I was thinking about trying,
for example, SRTP or SIP over TCP/TLS. Do you have implemented it
body help me with this issue.
Regards.
Daniel.
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Contact Digium For this issue
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohammed
Firdosh Nasim
Sent: mercredi 13 avril 2005 16:12
To: asterisk-users@lists.digium.com
Cc: Mohammed Firdosh Nasim
Subject: [Asterisk-Users]Unable to register license for
Hello,
Is there any way to turn Loop Detection off or tune the params a bit?
I am having an issue with Call Forwarding on my SIP Proxy Server which
is causing me great pains.
Here is the issue:
1) I have a SIP UA which registers with a SER proxy server.
2) I have an Asterisk TDM gateway in my n
; however
after enabling this and placing a few test calls I seem to get
extremely long delays in establishing the forwarded leg of my calls
and the RTP stream is not being relayed correctly.
I'm currently at a loss. I am seriously considering replacing this
Asterisk TDM gateway with a Cisco 535
There seem to be a little hack for "AMP" to assist with the conf files for
asterisks relating to polycom phones, has anyone come accross this little
fix?
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Is there a "Plugin" for AMP to ease Polycom 500's Configurations?
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ely must have
> encountered (overcome?) this problem since it is so fundamental. Perhaps a
> bug should be raised?
>
> Regards
>
> Cameron
> - Original Message -
> From: "Daniel Corbe" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List -
rsal.
Is there other standards how to indicate to the caller that the callee has
answered the call? How does it work in other countries?
Thanks!
--
Daniel Nyström
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ing to test how successfully this type of number can be used
with an Asterisk server half way around the world in the UK, so would
prefer to be using a service that offers fairly constant performance.
Regards,
Daniel
smime.p7s
Description: S/MIME Cryptogr
lly a requirement.
Any suggestions?
Thanks,
Daniel
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to, somehow, indicate on
the FXS-line that the other user has answered (lifted his/her handset).
By changing battery polarity or maybe an signal?
Automatic equipment does use almost every FXS-line of ours, and they need to
know when the call is answered.
Any ideas please?
--
BR
Daniel
everything worked perfectly. Does anyone have any idea what
settings I need to use on the UIP200 or why would the media be
"blocked"?
Thanks,
Daniel
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The SPA-841 doesn't seem to have conference call feature. This is
extremely important.
- Daniel
On Apr 20, 2005, at 11:12 AM, Kerry Garrison wrote:
I currently use an SPA-841 on my desk and don't have any problems with
it
http://www.geekgazette.com/index.php?option=com_content&t
Does anyone have any experience with this phone? I'm considering
purchasing it but wish to hear if anyone has any experience with it.
Thanks,
Daniel
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additional software like Hylafax?
Thanks,
Daniel
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t: Wednesday, April 20, 2005 2:37 PM
Subject: Re: [Asterisk-Users] SIP Phone Compatability
On Wed, 20 Apr 2005, Daniel Salama wrote:
Every once in a while I read messages about people having problems with
certain models of SIP phones, some of them being well known models.
I'm interested in pu
that even is Asterisk is "monitoring" a call on
another Asterisk box, then the box with the call being monitored will
not suffer any load overhead for Monitoring? I guess what I mean is
that, will that offload the Asterisk box handling the actual call?
Thanks,
Daniel
On Apr 20, 2
olding, be transfered to another agent group, or leave a message. Is
this possible? Does anyone have any sample of how to do this?
Thanks,
Daniel
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So the options are?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins
Sent: Thursday, April 21, 2005 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Trevor Harrison
Subject: RE: [Asterisk-Users] BYOD provider other than broad
Would you happen to have some sample config on how to do this? Is this
done in queues.conf or in the dial plan? Even pseudocode will help.
Thanks,
Daniel
On Apr 21, 2005, at 2:55 PM, Henry Devito wrote:
3) When callers call into the * box and the agents are busy, they
will be put on the queue
t
server which will also execute lame after soxmix.
Comments?
- Daniel
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.
What would be your approach? Would you still use SER for anything?
Is this the right list to post this question?
Thanks,
Daniel
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is reporting. i
tryed to keep everything as simple as possible.
thank you for your responses
daniel
[SIP]
exten=> 1000,1,Dial(SIP/cipi,20)
exten=> 1000,2,Answer
exten=> 1000,3,Hangup
exten=> 1001,1,Dial(SIP/nicu,20)
exten=> 1001,2,Answer
exten=> 1001,3,Hangup
exten=> 1002,1,Di
Hi try adding the following line to zapata.conf at the bottom:-
#include zapata-channels.conf
http://sourceforge.net/forum/forum.php?thread_id=1266472&forum_id=420324
asterisk-users@lists.digium.com
asterisk-users@lists.digium.com
When ever I try to dial a pstn number, I get this message.
(through an E1 and EuroISDN).
Any advice will be appriciated!
Thanks!
--
Daniel
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audio streams congest the WAN during busy periods.
- Daniel
On Apr 22, 2005, at 9:10 PM, Brian Roy wrote:
On 4/21/05, Matt Roth <[EMAIL PROTECTED]> wrote:
Daniel,
I would be interested to hear if anyone knows of a method to
completely
offload the Monitor command from the master server. It is
I'd like to create a dial rule that when someone tries to dial a
particular number, the same number is dialed, except that prefixed with
some additional digit(s). How can this be specified on extensions.conf?
Thanks,
Daniel
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8 blade to make
calls that aren't on the server? We would go through the CCM, but we're
trying to get rid of the cisco server altogether, and we don't really
want to bother with T1 cards (at least, not at this time).
Aaron Daniel
Senior Voice Analys
Franco/Tony/Josiah,
Thanks for the feedback. That did it.
- Daniel
On Apr 25, 2005, at 3:53 PM, Franco Bellagamba wrote:
Daniel, try
_X.,1,Dial(123${EXTEN})
That will prefix "123" to the dialed number.
Franco
- Original Message -----
From: "Daniel Salama" <[EMAIL PR
b_2000)
Note, this is DEFINITELY not tested and is only a suggestion.
- Daniel
On Apr 26, 2005, at 3:01 PM, Wiley Siler wrote:
The short answer is No.
The method you describe is intrinsicly illogical.
Assuming there is an IVR, how will I know which extension 2000 I am
calling if that were possible?
I mig
Is there any documentation for all the variables available in
extensions.conf? Every day I read this list, I read of at least a new
variable name that I wasn't aware of so I go out and read bits and
pieces about it.
Thanks,
Daniel
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ny clues on how to do this, if at all possible?
Thanks,
Daniel
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ow do you specify how you want Monitor to save the audio. Sorry for my
ignorance.
Thanks,
Daniel
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ut if you didn't take
a look.
I'd still like to hear feedback from others in the list regarding my
original post or the accuracy of this file.
Thanks,
Daniel
On Apr 26, 2005, at 10:37 PM, Jason Walker wrote:
I second this. Thanks.
-Original Message-
From: [EMAIL PROTECTED]
Yes, I read them. But, then my question is: how can I make the file
name include the agent that "will" get the call once it's distributed?
Thanks,
Daniel
On Apr 27, 2005, at 3:40 AM, Anton Krall wrote:
You can record queue conversations, check out the configs in queue.conf
|-O
I'm attempting the following set up.
During Hours - Receptionist Takes the call (no problem works great)
After hours I would like to add a item to the receptionist to transfer the
call to my cell, any direction would be a great help.
I have 4 PSTN incoming lines as backup and Voicepulse.
I apologize if the subject of this message is not the correct one.
My question is: does anyone have any statistics as to how an asterisk
box will behave "transcoding" SIP calls to IAX2 calls? How many
simultaneous calls can it handle?
Than
Does anyone know what this mean?
Apr 28 11:53:51 WARNING[907]: chan_iax2.c:6039 socket_read: Received
mini frame before first full voice frame
Thanks,
Daniel
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Thanks,
Daniel
On Apr 28, 2005, at 5:31 PM, Andy Hamilton wrote:
http://lists.digium.com/pipermail/asterisk-users/2005-April/103136.html
On 4/28/05, Daniel Salama <[EMAIL PROTECTED]> wrote:
Does anyone know what this mean?
Apr 28 11:53:51 WARNING[907]: chan_iax2.c:6039 socket_read: Receive
ands), that are ready for this?
Thanks,
Daniel
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Could you point me in the direction where you read that? Maybe there is
more there to read.
Thanks,
Daniel
On Apr 28, 2005, at 6:31 PM, Michael D Schelin wrote:
I just read a great paper that said turn off anything that won't be
used. Serial, USB , Printer ports, ETC. No Xwindows!
D
eryone. Just do what Matt
says: copy the -in and -out to archive server separately several times
a day :) - don't record to NFS mounted drive.
Thanks,
Daniel
On Apr 28, 2005, at 6:42 PM, mattf wrote:
I have never been able to do more than 50 concurrent recordings with
Zap ->
SIP p
ing or
IAX-SIP transcoding?
2) Will a single CPU machine handle the 4 T1s to do TDM-IAX
transcoding, if another machine is doing the actual recording (IAX-SIP
transconding) (Scenarios 2,3,4,5). Basically, just setup "cheap"
Asterisk boxes to act as VoIP gateways and the distribute the &q
quot; the channels on my
T1s to these clients. I just want to have a pool of channels from all
my T1s available to all my clients and restrict channel usage in some
other way.
Did I make any sense? Hope you understood what I'm asking for.
Thanks,
Daniel
__
.
I don't know if these tests are conclusive yet. However, from the
results so far, I would recommend staying away from recording to NFS
mounted point. I will continue running simulations to see if anything
else can be identified.
Thanks,
Daniel
On Apr 28, 2005, at 7:26 PM, Matt Roth
call (PSTN channel) with internal IVR script.
I like Scenario 6. Will look into that further. However, if the above
information gives you more grounds to make additional comments, please
do so :)
Thanks,
Daniel
On Apr 29, 2005, at 10:21 AM, mattf wrote:
If price would truly not an option just get
from the other.
Comments?
- Daniel
On Apr 29, 2005, at 2:42 PM, Matt Roth wrote:
List members,
Does anyone have an interest in forming a hardware architecture group?
It seems that Asterisk is so tightly linked to specialized hardware
and its corresponding architecture that developing the software al
This is an interesting question. I haven't tested it but would love to
know if it works or not. Anyone?
- Daniel
On Apr 29, 2005, at 3:38 AM, Michael Welter wrote:
I haven't seen this before--can an agent log into a queue on a remote
(i.e. over IAX) Aster
Does anyone have any experience with servers from siliconmechanics.com?
Are they reliable? How does * run on them?
Thanks
- Daniel
On Apr 29, 2005, at 4:22 PM, snacktime wrote:
Personally I would buy an * box from someone like asaservers.com. At
least companies like that really know their
to have to manually edit all contexts. Is there a
way to do something global to create something like a black list of
caller IDs to block?
Thanks,
Daniel
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What's what I'm trying to avoid. To answer your question: I have TE4XXP
with T1s (not PRIs). What I want to do is block it based on the
caller-id and not the DID Number. That way, I don't have to write 100+
lines.
Thanks,
Daniel
On Apr 29, 2005, at 6:23 PM, Stefan Gofferje wrote
sterisk_1
should be automatically routed to Asterisk_2 preserving all call
features, such as DID, CallerID, etc.
Any ideas?
Thanks,
Daniel
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Tim,
This certainly looks interesting. I just have a question about the
recipe: it makes reference to some AGI perl scripts. Is the source
available? Or may be it's irrelevant.
Thanks,
Daniel
On Apr 29, 2005, at 9:10 PM, Tim Litwiller wrote:
Daniel Salama wrote:
Question: how can I
x27;d like for the connection to be
done using SIP instead of IAX. Can anyone help me, if at all possible,
write this configuration?
Thanks,
Daniel
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I was reading on the wiki about the supported kernels and I __THINK__
the main issues with the kernel versions have more to do with Zaptel
driver and not necessarily Asterisk itself. Is this correct?
Thanks,
Daniel
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??)
exten => 1234,2,Hangup
Question is, when the agents dial 1234, how do I tell the application
to connect to the agent with context test-ivr of Asterisk_1?
Thanks,
Daniel
On Apr 30, 2005, at 7:12 PM, Tim Connolly wrote:
Maybe I'm missing something, but as long as you have the entension
def
Thanks. That's what I needed.
- Daniel
On Apr 30, 2005, at 8:00 PM, Tim Connolly wrote:
Asterisk Box 2 (agents register)
extensions.conf
[agents-context]
exten => 1234,1,Dial(SIP/[EMAIL PROTECTED])
exten => 1234,2,Hangup
Asterisk Box 1
Sip.conf
[ab1]
type=friend
host=
context=incoming
Along the same lines, is there some sort of capacity chart that maps
capacity based on translations/transcoding?
- Daniel
On May 1, 2005, at 2:45 PM, Greg Boehnlein wrote:
On Sun, 1 May 2005, David John Walsh wrote:
what sort of level of PC is required for 300 concurrent calls?
Doing what? Ulaw
gt; 200 result=0
Any clues?
Thanks,
Daniel
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manually execute a command that will give me the extension it was
transferred to?
Thanks,
Daniel
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Anyone know if Digium cards, especifically TE410P, are compatible with
BSD (FreeBSD or NetBSD)? How does * run on BSD?
Thanks,
Daniel
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nnect conventional telephone equipment, then you have more systems to choose from, like FreeBSD, Mac OS X and Solaris
It sounds as if BSD-like OS are good to run asterisk without the digium boards.
Thanks,
- Daniel
On May 2, 2005, at 12:06 AM, skamp wrote:
asterisk runs great on BSD if you follo
os, so the
problem seems to be within Asterisk.
Any tips is appriciated!
--
Daniel
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enant
basis, so that I could define something like operator => SIP/123 for
tenant A and operator => SIP/456 for tenant B?
I read about SetGlobalVar, what I think that would make the variable
available to all contexts (in my case tenants).
Tha
.
Any suggestions would be greatly appreciated.
Thanks,
Daniel
On Apr 30, 2005, at 8:00 PM, Tim Connolly wrote:
Asterisk Box 2 (agents register)
extensions.conf
[agents-context]
exten => 1234,1,Dial(SIP/[EMAIL PROTECTED])
exten => 1234,2,Hangup
Asterisk Box 1
Sip.conf
[ab1]
type=friend
host
Sorry if this is posted twice. I sent this about an hour ago and haven't seen it in the list yet.
Thanks,
Daniel
Begin forwarded message:
From: Daniel Salama <[EMAIL PROTECTED]>
Date: May 3, 2005 1:12:51 PM EDT
To: "Tim Connolly" <[EMAIL PROTECTED]>
Cc: "&#
multiple T1s from the
same provider? If they should share their timing setting, what would be
the right syntax to specify it?
Thanks,
Daniel
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municating using IAX.
Comments?
Thanks,
Daniel
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n, some
of our T1s tend to flap and I'd like to be able to monitor that. Is
there something that can do this?
Thanks,
Daniel
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To U
What exactly should I need to change in indications.conf??
Thanks.
Daniel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of wells
zheng
Sent: Wednesday, May 04, 2005 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
ing as basic as this?
Thanks,
Daniel
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
administrator tootai wrote:
| Daniel Nyström a écrit :
|
|> My server is located in Sweden. And as many European countries,
|> we use a 0 to indicate area codes, and 00 to indicate
|> international calls. And, when not having any leading 0,
Thank you all for pointing this out to me. I wasn't aware of the
physical difference between the 3.3V and the 5V PCI slots.
I guess it's time to change the motherboard.
Thanks,
Daniel
On May 4, 2005, at 9:32 PM, Rob Thomas wrote:
Well, no, it looks like the 8S661FX (which is actua
people "spam" calling and hiding their caller
id, specially with the DND list.
- Daniel
On May 5, 2005, at 4:10 AM, Robert Rozman wrote:
- Original Message - From: "Peer Oliver Schmidt"
<[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercia
uses
RealTime and one that uses Asterisk Database, and all I wanted is to
manipulate the dial plan, which would be more efficient/recommended to
use - RealTime or Asterisk Database?
Thanks,
Daniel
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be tied to our workstations, so being able to take calls
from any given phone is an important consideration. In the same vein,
knowing the status of other staff (i.e. if they are on a call or idle)
would be very useful, and is something we are used to with our current
setup.
Thanks,
Daniel Bingham
[
support 6 lines under SIP, or is the line functionality of the
IP-500 and IP-600 identical?
Thanks,
Daniel Bingham
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Bingham
Sent: Thursday, May 05, 2005 10:36 AM
To: aste
I have most agents using Alaw and/or ulaw and a handful of agents using
gsm.
Thanks,
- Daniel
On May 5, 2005, at 1:12 PM, Mehdi Chouikh wrote:
If use Alaw or ulaw as codec, i think it's
enough.
But if you need to make transcoding to a hard codec like g729, g723,
you have to look othe
ki says the IP-500 requires an additional chip to support power
over ethernet. Is this true of the IP-600 as well?
If anyone can answer any of these questions, I would really appreciate
it.
Thanks,
Daniel Bingham
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTE
e phone doesn't support redirects, it gets a little complex, in that the
script will need to open the file from the filesystem and return it directly.
If I misunderstood or I didn't make sense, I'll be happy to try again.
Thanks,
Daniel Bingham
[EMAIL PROTECTED]
-Orig
ake a better look in the source when back at work on monday,
but some tips and facts will help for sure.
- --
Daniel
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.5 (MingW32)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFCfIIJ/4dZjWjLCy0RAgHpAJ9Vs6qWuzioO
it or other channel banks?
I'm also unable to use V.90 modem through my setup (Adit600 via MGCP
- -> Asterisk -> E1).
Fax worked once though.. Does the Echo Cancelling make the problems
with V.90?
- --
Daniel <http://www.faqs.org/rfcs/rfc2833.html>
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Version:
Not entirely sure, but, I wonder what would happen if you define RING
in the [globals] section first, and the use SetVar or SetGlobalVar in
the other contexts to override its value.
- Daniel
On May 8, 2005, at 2:48 AM, Mark Wormgoor wrote:
Hi,
Is it possible to set a variable for a context
ants to make ANY
outbound call, it will simply send it to the [outbound] context of
the gateway server for it to place the actual call(s).
Does this make sense to you guys? Am I missing anything? Is there
anything I should be concerned with or that I should watch out for?
Thanks,
Daniel
___
And how/where can you download the latest firmware onto /var/lib/
asterisk/firmware/iax?
Thanks,
daniel
On May 7, 2005, at 9:13 AM, Time Bandit wrote:
I'd like to known what I have to do to upgrade
the firmware into a IAXy device.
It does it automagically when it connect to Asterisk if a
moved from another machine running REL3
and all 4 T1s were working there. I don't know why only two T1s are
being "recognized" on the Debian system.
Secondly, I no longer have zttool, which I used to have on REL3. How
can I query from Zaptel the status of my T1s?
Is it possible to set a variable for an IAX device in iax.conf that
can be read from the dial plan (extensions.conf)? If so, can you
explain?
Thanks,
Daniel
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ccess them as ${LOCAL},
but LOCAL is never defined in extensions.conf. Instead it is defined
in iax.conf.
I haven't seen any facilities to define variables in iax.conf, so
that's why I asked.
Thanks,
Daniel
On May 10, 2005, at 7:14 PM, Alfredo Manrique wrote:
In extensions.conf in the
the context used in extensions.conf? By the
time the call is going to be directed to the agent, the call is
already in the system and has gone through several contexts and
queued by AppQueue.
Thanks,
Daniel
On May 10, 2005, at 9:11 PM, Moises Silva wrote:
at the moment i think is not
Sorry for the confusion.
On May 10, 2005, at 10:45 PM, Tzafrir Cohen wrote:
On Tue, May 10, 2005 at 02:52:39PM -0400, Daniel Salama wrote:
I just installed a TE410P on a Debian Sarge system running kernel
2.6.11-1-686-smp.
Which is a kernel from sid, actually. What version is it exactly?
dpkg -l
>call for you.
>
>
>
Broadvoice is having issues. For me inbound calling is down (this is
day two).
Their mantra is : "We are currently experiencing in-bound call issues
with a carrier partner in some areas. We are aware of the issue and our
engineers are working to have it resolved as soon
I'm noticing by watching the CLI that my inbound calls coming via T1s
on a TE410P are using GSM codec. Why wouldn't it use ULAW as default?
How can I make it use ULAW as default?
Thanks,
Daniel
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