Hi,
I'm using asterisk 1.14.19
I'm making a video call between two SIP end-points, using h263p and iLBC. I
notice the video is jumpy and I believe the cause is due to RTP timestamps.
The sending device is working at 8fps and correctly increases the timestamp
by 11250 every frame. It appears
Venefax wrote:
I need to dial a DTMF string with the Dial function using the D(“DTMF”)
function. What is the character for a delay? I mean, normally in other
technologies we use the comma to mean “wait 200 ms “. Is there an
equivalent in Asterisk? If it is the comma indeed, how many ms
Alex Balashov wrote:
Venefax wrote:
I need to dial a DTMF string with the Dial function using the D(“DTMF”)
function. What is the character for a delay? I mean, normally in other
technologies we use the comma to mean “wait 200 ms “. Is there an
equivalent in Asterisk? If it is the comma
Alex Balashov [EMAIL PROTECTED] writes:
Yes, indeed. Encapsulation protocols such as IPSec/GRE won't work at
all over high RTT latency (= 400 ms).
Sure they will. They just won't benefit from TCP acceleration
performed by the satellite company. Sadly TCP itself is very slow with
high RTT, so
Probably I did not read well the information
I am concerning, if I am going to use ARA for the SIP
and I have
register = user:secret:[EMAIL PROTECTED]:port/extension
how I should input that line?
If I am going to delete it from the DB I am forced to reload everything or
there is a way to tell
I'd like to do the following can someone guide me on how to accomplish this?
Call comes in via PRI and tries to go out via SIP if for some reason the ISP
is down and the call can not go out i want it to fail over and send the same
call through a different PRI.
I was thinking something like
On Sun, Aug 24, 2008 at 8:11 AM, Jon Weisman [EMAIL PROTECTED] wrote:
I'd like to do the following can someone guide me on how to accomplish this?
Call comes in via PRI and tries to go out via SIP if for some reason the ISP
is down and the call can not go out i want it to fail over and send
John,
This is the default behaviour anyway. If Dial() is successful,
execution of subsequent priorities in the dial plan for that extension
is not resumed. It'll only fall through to the other priorities if
Dial() fails.
I do, however, suggest supplying a timeout argument to your Dial()s.
Hi all,
i;m obviously a newbie, its been 2 days that im trying to figure out a way to
deny a specific extension (300) from calling another specific extensions (03)
except if the caller punch a specified password.. sorry if im not explaining
myself well.. heres an example:
i called my pstn
You want to use Authenticate() between answer and dial.
http://www.google.com/search?q=asterisk+authenticateie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a
Thanks,
Steve Totaro
On Sun, Aug 24, 2008 at 11:26 AM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
Hi all,
i;m obviously
Is there a way to force PSTN line on ATA (Linksys / Sipura) to go ON hood
via dial plan?
I have two Linksys units one connected to old POTS line and one connected via
Shaw Digital Phone.
The one connected to Shaw Digital Phone the PSTN Line fails to go ON Hook at
the end of the conversation.
Hello Roland,
You can use the cmd Read for this.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
Pretty straight forward. Whenever you need to accept DTMF input from the user
collect the required digits using Read; check the collected digits; if yes jump
to required extension; else
I have a Sipura 962 endpoint on Asterisk 1.4 where the MWI works
perfectly, however my theory is that it should be broken.
Obviously I'm wrong but Sip show subscriptions does not show the
endpoint subscribing to the MWI status on Asterisk, even though all of
the other endpoints on the system DO
Just tried upgrading to 3.3.0 by using the 3.2 spec and removing the patches
etc, but after the system booted and it tried to start the images I got the
message :-
Error: Boot loader didn't return any data
Has anybody else upgraded yet ?
Regards,
--
--[ UxBoD ]--
// PGP Key: curl -s
Gordon,
I have decided to ditch using Xen for Asterisk and build a separate server. It
will be used in the home office and this rack case looks great as I am going to
get a small comms cabinet. We will be using a TDM400P to bring PSTN to the
server, plus we have a few SNOM M3 phones around
On Aug 24, 2008, at 11:29 AM, Karl Fife wrote:
FYI, it's not an issue of the subscription not YET subscribing etc.
If
I were to restart the system and endpoints, all the subs slowly show
up
one by one, but the 962 never does -- even after days, weeks, and
months. Yet the MWI always
Better solution for Comms Cabinet.
http://www.hoffmanonline.com/product_catalog/product_detail.aspx?cat_1=34cat_2=2410cat_3=42359catID=42359itemID=3599
On Sun, Aug 24, 2008 at 12:38 PM, --[ UxBoD ]-- [EMAIL PROTECTED] wrote:
Gordon,
I have decided to ditch using Xen for Asterisk and
Just about anything bootable will serve you well enough. My home office
is presently running of an HP T5700 thin client (1 GHz transmeta CPU,
256 MB RAM, 256 MB Flash DOM) running Astlinux. Cheap, diskless,
fanless, silent, cool, reliable. Almost perfect IMHO.
It's connected to a snom m3 system,
In the UK these look very nice :) going to get a dark grey/black mini one
http://www.orionuk.biz/default.asp?productDetails=3
Regards,
--
--[ UxBoD ]--
// PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import
// Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84
//
SECURITY QUESTION SANITY CHECK:
If only my SIP ports and a small range of RTP ports are facing the
public internet, what is the method by which an evildoer would be able
to do fraudulent long distance on my nickel?
Would it REALLY be as simple as guessing the credentials for ANY of my
local
From what people have said Asterisk does not require a huge amount of memory
or CPU then ? I only have a couple of extensions. Running the G729 codec and
will look at the Octava software for the PSTN to reduce echo.
Regards,
--
--[ UxBoD ]--
// PGP Key: curl -s
For a couple of years I used a Soekris Net4801 as an Astlinux host. It
features a 266 MHz Geode CPU. That system was able to sucessfully
transcode only two G.729 encoded calls at a time. Since moving to the 1
GHz host I've not bothered to try and overtax the transcode capability
as it does as much
Reason for the G729 is that I have a couple of numbers which come in directly
over IAX. Switched office numbers away from BT onto VoIP to reduce cost :)
Regards,
--
--[ UxBoD ]--
// PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import
// Fingerprint: F57A 0CBD DD19 79E9 1FCC A612
On Sunday 24 August 2008 14:17:47 Karl Fife wrote:
For crude IPS/IDS is there an Asterisk method to blacklist registrations
from a specific IP address after a certain number of failed registration
attempts, or would I need an SBC or IDS/IPS for that?
There is no solution in Asterisk currently,
Hello Steve,
thanks for the advice :)
though one prob! if i add the authenticate line itll require all callers to
enter 1234 to access *ANY* sip account..
even though this would come in handy at some point but at the moment i just
want to deny the extension 300 from being able to call 01
Thanks all for your suggestions.
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On Sun, 24 Aug 2008, --[ UxBoD ]-- wrote:
Reason for the G729 is that I have a couple of numbers which come in
directly over IAX. Switched office numbers away from BT onto VoIP to
reduce cost :)
Don't use g729 unless you really need it. Even on an old ADSL-1 connection
in the UK (256Kbps
It was driver error caused disk failure. It is fixed now but we still have
problems with D channel up/down messages. Can we fix it reconfiguring
software or changing something?
- Original Message -
From: Rob Hillis [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
I have one solution in mind, maybe it is an overkill but:
You can create a db entry for each sip account, DB(family/key) lets name
family=destination sip number and key=${Callerid(num)} and assing a value 0
or 1, so string will be like this DB(301/300)=1 fot that 300 sip account,
and for all
I have one solution in mind, maybe it is an overkill but:
You can create a db entry for each sip account, DB(family/key) lets name
family=destination sip number and key=${Callerid(num)} and assing a value 0
or 1, so string will be like this DB(301/300)=1 fot that 300 sip account,
and for all
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all,
I've got a problem with RemoveQueueMember. If an agent receives a call
at the same time that they're being removed from the queue, they receive
that call and remain in to the queue (and will receive calls until
they're able to be removed).
Roland,
The simple solution is to utilize the power of contexts (put exten 300
in a different context in sip.conf or db) and includes to separate yet
include 300 (so 300 can be called and call other internal extensions).
Add authenticate before the dial statement.
The easiest way to do it, is
Anyone has tried wct4xxp drivers' alarmdebounce parameter ?
I search the internet no one seems to have used it,
is this how the parameter can be specified, eg :-
# modprobe wct4xxp alarmdebounce=200
Does it have the effect of making asterisk PRI more
tolerant with poorer quality lines ?
I see. Thanks Russel.
And I now notice that if I explicitly tell my 962 the IP address of the
mail server, it will also subscribe.
Do I understand correctly that we are not talking about redundant MWI
status traffic here, we're ONLY talking about the notion that asterisk
ignores MWI
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