This feature would be really great but i dont think asterisk supports it .
It either shows dialplan execution of all extensions when verbosity is
increased or of none when set to 0 . You can set verbose 0 and sip debug a
single peer but you cant enable dialplan execution viewing for single
Hello guys,
i have some problems with chanisavail and misdn.
Used the following syntax
Chanisavail(misdn/g:TEPortsIAX2/trunktosecondserver)
Checked with
Chanisavail(misdn/1IAX2/trunktosecondserver)
Chanisavail(misdn/1/${EXTEN}IAX2/trunktosecondserver)
too.
I always get the reply mISDN/0-u11
Hi Kevin,
Hi list,
you are right, acting now is not needed, when callbacklogin will be removed
anywhere in future...
But thinking how to realice alternatives can't be so wrong.
Callbacklogin gives a very simple way to use more queues for one agent,
which only has to logon to only one system.
No
sorry, was only for users list...
Hi Kevin,
Hi list,
you are right, acting now is not needed, when callbacklogin will be removed
anywhere in future...
But thinking how to realice alternatives can't be so wrong.
Callbacklogin gives a very simple way to use more queues for one agent,
which only
Yes just download new version of asterisk,zaptel,libpri . make install
for all 3 ( first libpri , then zaptel, then asterisk ) . It is recommended
to stop asterisk b4r doing make install of new version . Do not do make
samples or it will overwrite you config's . After installing newer zaptel
do
Dear all
I have asterisk 1.2 and now i want to install E1 card with
support Q.SIG singaling so which E1 card is best for my setup i need single
port E1/PRI card which support Q.SIG
Regards
Satish patel
-
Take the
On Wed, Jul 04, 2007 at 12:58:47AM -0400, Wai Wu wrote:
I have a X100P and a TE110P in my Asterisk box. I can either get the
X100P or the TE110P to work, but never both. Here's my zaptel.conf
span=1,0,0,d4,ami
em=1-24
fxsls=25
When I load wcte11xp and wcfxo, I will get this error.
On Thu, Jul 05, 2007 at 08:09:32AM +0400, Eugene Prokopiev wrote:
Hi,
Is it possible to filter messages on asterisk console, which was started
with -, to see messages only for one extensions? By default there
are all messages for any extensions displayed so dialplan debuging is
very
You are right but my concerns is the ITSP's may stop allowing it because they
don't want to get in to trouble. They may request a list of all the DID's that
I have and limit me setting my CID to the list that I gave them.
- Original Message -
From: Andrew Joakimsen
To: Asterisk
Hi,
I have an issue related to SIP channels not able to drain the udp queue-
I have only 150 uas registered to * ,v. 1.2.8
When Problem appears all uas loose their reachability. After a while the
service
Becomes available again-. UDP messages still comes to my ethernet board but
The process is
. . .
We let you win, you were terrorists and England's never been good at
fighting terrorists. Now you're having the same problem !!!
One is stuck by the semi-irony. Those who do not learn from History are doomed
to repeat it. However, the current unpleasantness has dis-similar roots.
Hi,
I'm testing attended transfer with 3 SIP phones. I noticed about 10% of
my transfers make the call drop and I get this on my log:
Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback:
Failed to write frame
-- Playing 'beep' (language 'it')
Jul 5 10:42:32 WARNING[23960]:
Hi All,
Does anyone know what the current status is of the G729 codec on
Solaris? According to the following link:
http://www.asteriskvoipnews.com/asterisk_releases/_digium_g729_codec_now_available_forsolarissparc.html
there is a version available for SPARC processor's. However, I have just
had
Am Mittwoch, den 04.07.2007, 11:00 -0400 schrieb Noah Miller:
Is it just me? After the mail list server upgrade, the average delivery
time for messages to the users list is between 4 and 5 days. The Dev
list seems fine!
I'm getting new messages within a matter of minutes. I dunno.
As
Hi Everyone.
I'm in a quandry don't know which way to go. - Obviously I'm an Asterisk
newbie although I've been watching this list for over 2 years now.
I've got an Asterisk box (actually, it's an AsteriskNOW box) up and running
here at home. - It's on my home LAN - NAT'ed behind my LinkSys
It is recommended
to stop asterisk b4r doing make install of new version .
It should work to do a make install of a new version while the
previous version is running. At least, I've never had any issues
doing it.
- Noah
___
--Bandwidth and
List,
just my two cents here on BRI cards for Asterisk - sorry if the
following info was posted before/is redundant.
You will find the following BRI cards for Asterisk. Depending on the
chipset/Asterisk module they will/will not support NT mode, have
different numbers of ports, scale well when
Hi Giorgio -
I'm testing attended transfer with 3 SIP phones. I noticed about 10% of
my transfers make the call drop and I get this on my log:
Some questions:
1. What asterisk version are you using?
2. What are your SIP devices?
3. Who is your SIP provider? (Judging by your CLI output, I'm
Hi Gary -
What I want to do is take one of my SIP devices to my office (which is ALSO
behind another NAT) and try to connect with my home Asterisk box with it.
For port forwarding, my AsteriskNOW box has a static IP on the inside of my
NAT and I've configured the LinkSys router to
Hi Noah,
1 - my asterisk version is 1.2.18
2 - my SIP devices are SNOM phones
3 - no SIP provider is involved...they are connected to my
Asterisk...this is the strangest thing.
This happens sometimesI think it could be a network overload...can
it be?
TIA
Giorgio
Noah Miller wrote:
Hi
Quoting Frank Ochmann [EMAIL PROTECTED]:
List,
just my two cents here on BRI cards for Asterisk - sorry if the
following info was posted before/is redundant.
You will find the following BRI cards for Asterisk. Depending on the
chipset/Asterisk module they will/will not support NT mode,
Verizon wants to pretend the service doesn't exist
I don't know, they're advertising it:
http://www.verizonbusiness.com/us/voice/local/compare.xml#isdnbri
- Noah
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Is there something special in connecting a 1.2 and 1.4 systems???
I have connected a number of 1.2 systems using SIP between them and had
no issue.
In this case I am getting 407 Proxy Authentication Required
I did the same steps I did for 1.2
Basically I have entries in sip.conf on both
Since the list was switched over to API-Digital almost
every message I get is older than a week. Coincidence?
Is anyone else having trouble?
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Hooyoo kiddin? Exit 34, I-80.
And betta Inglish, myass...
Bill, Exit 8, NJTP
Date: Tue, 03 Jul 2007 18:13:47 -0400
From: Mark Phillips [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
Damn!!! Beat me to it ;-}
As an Englishman now living in New
Hola,
What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
NXXNXX?
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
tel:+1 219 836 8918 x325
fax:+1 219 836 1138
email:[EMAIL PROTECTED]
web:www.torrenga.com
Joe acquisto wrote:
. . .
QOS across the internet is pointless and further more doesnt really
exist, I would suggest setting qualify=200 in sip.conf so that asterisk
will not send a call to the remote end if they are more than 200
milliseconds away.
Away, in what sense? Are
Hi Giorgio -
1 - my asterisk version is 1.2.18
2 - my SIP devices are SNOM phones
3 - no SIP provider is involved...they are connected to my
Asterisk...this is the strangest thing.
This happens sometimesI think it could be a network overload...can
it be?
Well, that's possible, but
Jeff Davis wrote:
Stephen Bosch wrote:
Your rep at Sangoma? Or your reseller?
That wasn't very clear. Sorry. It was Sangoma.
(I would be more verbose, but I don't want to spam the list)
I just wanted to make sure it wasn't stale information.
This is a real chicken-and-egg problem. More
Noah Miller wrote:
Verizon wants to pretend the service doesn't exist
I don't know, they're advertising it:
http://www.verizonbusiness.com/us/voice/local/compare.xml#isdnbri
Sure, but when you call someone up to buy it it's a different story. Or
perhaps it was just Verizon's usual good
On 7/4/07, Jon Pounder [EMAIL PROTECTED] wrote:
Quoting Darren Wright [EMAIL PROTECTED]:
I wonder if this is issue is largely limited to to Canada. (thus
limiting the market) In the states I think you can get PRI for around
$250. Am I right? In Canada, you have to have about 9 or 10
This would let you include/ignore a leading 1
1{0,1}[2-9]{2}[0-9]{8}
Brent Torrenga wrote:
Hola,
What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
NXXNXX?
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana
Quoting Stephen Bosch [EMAIL PROTECTED]:
Jeff Davis wrote:
Stephen Bosch wrote:
Your rep at Sangoma? Or your reseller?
That wasn't very clear. Sorry. It was Sangoma.
(I would be more verbose, but I don't want to spam the list)
I just wanted to make sure it wasn't stale information.
This
Maybe I am looking at things too naively, but if all thats different
is the signalling standard is this really a monumental effort to make
a driver work with more than one standard ?
I assume you would have various line encodings to deal with just like
t1, so you have a driver layer that
I have a customer asking about the type of equipment there is for
visually impaired employees working in a call center for inbound sales.
--
Best regards,
Al Bochter
http://www.BochterServices.com
---
See what we are selling at
Hi Again Brent -
What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
NXXNXX?
I think you've got it right already. What do you need to do?
If you wanted to get more specific and identify ONLY NANP, you may
have to break it out into more than just one rule:
Hi,
Just a quick question. Is there a way when making an IAX call to
transmit some additional call-data, perhaps in a variable? I could
overload callerid-name, but that is nasty and ugly :)
Thanks for any suggestions.
Regards,
Steve
___
--Bandwidth
Jon Pounder wrote:
most of the first level reps I have ever talked to in the last 10
years don't even know it exists, higher level people claim they don't
offer it, still higher level people know what you are talking about
when you say its tariffed and finally cave in to what you want.
Hi Brent -
What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
NXXNXX?
I think you've got it right already. What do you need to do?
- Noah
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
I have asterisk 1.2.18
Each my extention number has a mailbox; I have SIP extentions and IAX
extentions.
When I call an unregister extention number (SIP/IAX) from SIP
hard/soft-phone, asterisk actives the voicemail (I can put a message in the
mailbox).
Instead when I call an unregister
But those are not REGEX expressions, those are asterisk dialplan
pattern-matching expressions. great for the X in:
exten = _X.,1,blah
but not for use with REGEX() function.
I think it would be close to what Michael said, but like this:
1{0,1}[2-9][0-9]{2}[2-9][0-9]{6}
Michael, your
Asterisk is poor with codec negotiation . It does not check if it can avoid
transcoding by forcing codec available to both sides .. instead it will
read it's config file and will select first allowed codec that is also
available on other device on each leg of call and happily transcode between
On 7/5/07, Stephen Bosch [EMAIL PROTECTED] wrote:
I would be willing to help out with a driver, but without a line and
card I am not sure how productive that would be.
As I've already said, I can get one, and it's not a big deal, so I'll be
the test case.
-Stephen-
Ok, how about this for
I am using the Find-me/Follow-me example below with screening:
http://www.voip-info.org/wiki/view/Asterisk+tips+findme
Here is my actual config:
[macro-screen]
exten = s,1,Wait(1)
exten = s,n,Background(press-1-to-be-connected-to-the-caller)
exten = s,n,Set(TIMEOUT(response=5))
exten =
Quoting Dave Donovan [EMAIL PROTECTED]:
On 7/5/07, Stephen Bosch [EMAIL PROTECTED] wrote:
I would be willing to help out with a driver, but without a line and
card I am not sure how productive that would be.
As I've already said, I can get one, and it's not a big deal, so I'll be
the test
On 7/5/07, Dave Donovan [EMAIL PROTECTED] wrote:
I wonder if we need piece of reference hardware. Imagine this scenario,
we order the circuit and Bell wants to turn it up. What do we turn up
with? It would be great if we had something that we could plug into it and
get that working just to
Before poking Digium too much, I would look at exactly what YOUR mail
servers are doing that may potentially be the real cause of the delays.
Already did that. I use ASSP for filtering. Digium and associated
mailing lists are white listed. There was only 1 attempt for deliver
and there
On Thu, Jul 05, 2007 at 01:40:50PM -0400, Doug Lytle said:
Well, this is now the third active thread on this subject, but I guess
you won't see this message for a while. Has anyone dissected the
headers of a delayed message yet? We should be able to tell for sure
where the holdup is. All
My mistake...you're correct...should have tested it.
Mojo with Horan Company, LLC wrote:
But those are not REGEX expressions, those are asterisk dialplan
pattern-matching expressions. great for the X in:
exten = _X.,1,blah
but not for use with REGEX() function.
I think it would be
Jon Pounder wrote:
I have a bunch of old cisco stuff with BRI ports on it but it was
never meant for voice, just purely data, so I don't think its very
useful for this purpose, but some of the basic signalling could
probably be tested with it.
is exploring some sort of back to back
On Thursday 05 July 2007 2:38 pm, Doug Lytle wrote:
Already did that. I use ASSP for filtering. Digium and associated
mailing lists are white listed. There was only 1 attempt for deliver
and there were no delays. I subscribe to 10 mailing lists (Including
the dev list) and they are not
Dovid B wrote:
You are right but my concerns is the ITSP's may stop allowing it
because they don't want to get in to trouble. They may request a list
of all the DID's that I have and limit me setting my CID to the list
that I gave them.
I doubt this will ever be an issue. The telco companies
Quoting Jeff Davis [EMAIL PROTECTED]:
Jon Pounder wrote:
I have a bunch of old cisco stuff with BRI ports on it but it was
never meant for voice, just purely data, so I don't think its very
useful for this purpose, but some of the basic signalling could
probably be tested with it.
is
Hi,
I am getting half channel audio on cisco 7960?
Any idea why? Details below.
Jerry
This same phone has been working for MONTHS using a TDM2400p
with no issues.
Today we got a T1 installed coming into Box A with all incoming calls
going to Box B (the TDM2400P box).
The TDM2400 is no longer
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion
Sent: Thursday, July 05, 2007 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call Screening Not Working
I am
Hi everyone:
I've searching for a while and haven't found what i
need.
The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to be
able to call other numbers(any number) in the pstn
through my zap fxo channels. I have a big number of
sip users so as
I know there is FMFM in 1.4, but I want to know why the macro isn't working. I
added a NoOp and Wait to the macro as lines 4 and 5 and neither gets executed.
As soon as I hit any number such as 2, 3 or 4, I immediately get bridged to the
call. I may be wrong, but I'm pretty sure that
On Thu, 2007-07-05 at 15:08 -0400, Jon Pounder wrote:
Quoting Jeff Davis [EMAIL PROTECTED]:
Jon Pounder wrote:
I have a bunch of old cisco stuff with BRI ports on it but it was
never meant for voice, just purely data, so I don't think its very
useful for this purpose, but some of the
satish patel wrote:
Dear all
I have asterisk 1.2 and now i want to install E1 card
with support Q.SIG singaling so which E1 card is best for my setup i
need single port E1/PRI card which support Q.SIG
All T1/E1 cards using libpri have basic Q.SIG support. I recommend the
Hi Eve -
The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to be
able to call other numbers(any number) in the pstn
through my zap fxo channels. I have a big number of
sip users so as you can imagine there will be
congestion when some of
Noah Miller wrote:
Hi Eve -
The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to be
able to call other numbers(any number) in the pstn
through my zap fxo channels. I have a big number of
sip users so as you can imagine there will be
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Floyd wrote:
Hi everyone:
I've searching for a while and haven't found what i
need.
The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to be
able to call other numbers(any number) in the pstn
Dear Alex;
I am asking about:
What is the configuration that I can do it to let the
traffic between the two Asterisk PBX and another IP BX
to be g729 or G711 or g723?
In other words, how can I let the ued codec for the IP
Trunk between my Asterisk and the other IP PBX to be
g729 and not g711?
Bilal,
There are allow= options you can use on the peers in sip.conf to define
what codec capability Asterisk advertises toward them, and therefore, what
the negotiation on both call legs will ultimately settle upon. When those
legs are bridged -- as Asterisk does unto them -- they will be
Just curious,
I noticed that with SetLanguage() you can change it into a lot of other
languages. Yes one can record them easily enough with record, but
don't like to re-invent the wheel..
Browsed through a lot of google-pages but failed to find any other
languages (except for FR and ES on the
Has anyone observed a problem where using Local channels with AddQueueMember
results in missing TRANSFER events?
Right now I'm using straight SIP channels when I call AddQueueMember(). I'm
contemplating moving to Local channels because the non-state-based
wrapuptime blows when you have a
Dave Donovan wrote:
On 7/5/07, *Stephen Bosch* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I would be willing to help out with a driver, but without a line and
card I am not sure how productive that would be.
As I've already said, I can get one, and it's not a big
No idea if this is where your problems are coming from, but change:
exten = s,n,Set(TIMEOUT(response=5))
to
exten = s,n,Set(TIMEOUT(response)=5)
(the parenthesis moved a bit)
Peder @ NetworkOblivion wrote:
I am using the Find-me/Follow-me example below with screening:
Doug Lytle wrote:
Before poking Digium too much, I would look at exactly what YOUR mail
servers are doing that may potentially be the real cause of the delays.
Already did that. I use ASSP for filtering. Digium and associated
mailing lists are white listed. There was only 1
Jon Pounder wrote:
I have a bunch of old cisco stuff with BRI ports on it but it was
never meant for voice, just purely data, so I don't think its very
useful for this purpose, but some of the basic signalling could
probably be tested with it.
is exploring some sort of back to back
James FitzGibbon wrote:
Has anyone observed a problem where using Local channels with
AddQueueMember results in missing TRANSFER events?
Right now I'm using straight SIP channels when I call
AddQueueMember(). I'm contemplating moving to Local channels because
the non-state-based
Jeff Davis wrote:
I'm seeing ISDN phones on ebay for US $15-$40. Does anyone know if the
line simulator and a phone would work. Then get a BRI line when there's
a driver that looks like it works.
You'd think it would -- otherwise the line simulator is somewhat
pointless, isn't it?
I saw
David Boyd wrote:
I seem to remember that the wan Pipeline units supported BRI, and also
provided a couple of analog phone jacks. I will dig around in the
basement and try to find the one that I had, if I find it, who wants it
for play?
Well, whoever ends up with the simulator should get
Quoting Stephen Bosch [EMAIL PROTECTED]:
Jon Pounder wrote:
I have a bunch of old cisco stuff with BRI ports on it but it was
never meant for voice, just purely data, so I don't think its very
useful for this purpose, but some of the basic signalling could
probably be tested with it.
is
Hi Bilal -
In other words, how can I let the ued codec for the IP
Trunk between my Asterisk and the other IP PBX to be
g729 and not g711? Ofcourse, I am assuming that the
other side also supporting g729.
You can have multiple allow lines, i.e.
allow=g729
allow=ulaw
Be
stand s a large probability that the list server is trying that address
first.
We'll test your theory, I don't get anything but spam on my second
server, so I've had it shut down for a good portion of this year. I
just went and started it back up. If this was the case though, I'd
expect the
I highly recommend the Sangoma cards. They have good support for Asterisk
also for other systems as well :) Asterisk does support Q.SIG that is not an
issue.
On 7/5/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
I have asterisk 1.2 and now i want to install E1 card with
On Thu, 2007-07-05 at 16:42 -0600, Stephen Bosch wrote:
David Boyd wrote:
I seem to remember that the wan Pipeline units supported BRI, and also
provided a couple of analog phone jacks. I will dig around in the
basement and try to find the one that I had, if I find it, who wants it
Is the 'buddy' soft button on the bottom of the screen?
PaulH
On Tue, 2007-07-03 at 21:43 +1000, Farooq Ahmed wrote:
Hi all,
I am working on
asterisk 1.2.18
zaptel 1.2.17
Polycom 650
polycom 430
SIP version 2.0.3.0131 for IP 650
SIP version for IP430 2.0.3.0127
freepbx 2.2.1
I
David Wrote:
On Thu, 2007-07-05 at 16:42 -0600, Stephen Bosch wrote:
David Boyd wrote:
I seem to remember that the wan Pipeline units supported BRI, and
also
provided a couple of analog phone jacks. I will dig around in the
basement and try to find the one that I had, if I find it, who
Hi,
You can see some sample configurations from the link below
http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+recommendations
It really depends on the hardware, the codecs, what you are going to use
it for, etc,etc.
I, for example, have a Pentium IV 2.4Ghz, 2Gb RAM, using G729
Quoting Dan Austin [EMAIL PROTECTED]:
anyone interested take a read of this listing
http://cgi.ebay.com/4x-BRI-ISDN-2B-D-Asterisk-like-Digium-B410P-VOIP_W0QQitemZ180135963254QQihZ008QQcategoryZ61841QQrdZ1QQcmdZViewItem
I already did ask for details and mentioned that this thread was underway.
http://www.crtc.gc.ca/archive/ENG/Orders/2007/o2007-56.pdf
some discouraging directions being taken by the idiots at the crtc.
Essentially laying the groundwork to phase out bri completely in
Canada, probably fcc has similar idiots making similar decisions as we
discuss this.
Read the
Hi Farooq,
I've done just that for one of our customers. All I did was
add an exten such as *56 that set a custom database value to
nightmode=true. Then as calls come in I just check the database value to
see if it is set to true or not. Note I have asterisk patched with
Bristuff so
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