Hi Alberto,
I think that meetme and/or park should work.
We done something like this suing queue as well.
Could you please let us know why task #6 does not work in your case.
Map
On Nov 23, 2007 8:38 AM, Alberto Pastore [EMAIL PROTECTED] wrote:
Hi everybody.
I am in the following
On Thu, 22 Nov 2007, Mike wrote:
Hi,
I have a wholesale provider that allows me to put any caller id I want when
dialing out. In some cases, I`d like the outgoing callerid to be hidden.
How do I do this?
I`ve set callerid name to unknown, that works well, but when I put an
empty number
On 11/23/07, Jakub Syrek [EMAIL PROTECTED] wrote:
Well it's not working as it should. Every call go to Dail(SIP/sip1) and if
no one respond then to the next one :(
You should post a log then. Also CLI command group show channels
could be usable.
And please don't top-post.
Regards,
Atis
Hello list,
I'm observing what I believe to be inconsistent behaviour
regarding Newstate AMI events for the Ringing state.
As such I come to you asking for experience or advice: am
I wrong or should I file a bug ?
I present you a short introduction which I feel is relevant;
Hi,
Has onyone heard of successful deployment of 3Com ToIP over IBM iSeries
system (formely AS/400) ?
A prospective customer seems to looking for this but, in my whole life, I've
never of a such setup.
Does it work ?
regards
___
--Bandwidth and
On Wed, 21 Nov 2007 15:45:35 -0500, Baji Panchumarti
[EMAIL PROTECTED] wrote:
STAT() and record() are doing exactly what they are
supposed to. Use the s flag to fetch the file size. You
have to try a few hangups and figure out a minimum
file size that qualifies as a recording in your setup.
Just search in google IBM 3com and you will find what is this about.
I got a notification about one year ago about this solution. It is
basically 3com software PBX running on the IBM System i.
3COM VCX product.
On Nov 23, 2007 6:58 PM, Moises Silva [EMAIL PROTECTED] wrote:
I added the senddialevent, but not the condition you see below. That
one was added by someone else. It seems that determine wheter or not
the current extension will be set for outgoing calls. Setting
OPT_ORIGINAL_CLID may fix your
My guess is that the B channels are in fact bouncing in and out of service
and the message is a reflection of it.
On Fri, 23 Nov 2007, Michael J. Liberatore wrote:
Hi all, i have recently setup a p2p t1 using sangoma t1 cards and
asterisk 1.4. Its working great but i am getting an annoying
Hi all, i have recently setup a p2p t1 using sangoma t1 cards and
asterisk 1.4. Its working great but i am getting an annoying message
every little while in asterisk:
[Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/16
restarted on span 2
[Nov 23 19:17:57] VERBOSE[6487] logger.c:
Sounds remarkably like a deployment that I just did. Almost like you
reverse engineered it and wrote a FAQ. Pure genius!
Thanks,
Steve Totaro
888.777.1888
Alex Balashov wrote:
I made a little write-up that attempts to synthesise a lot of the
information out there about how to get HylaFAX
Alex, I thought asterisk 1.4 supports faxing internally now without the
need for extra software? Is your solution a different one? I have no
experience with faxing yet but plan to soon, that's why I ask and will
read your blog entry.
Thanks
Mike
-Original Message-
From: [EMAIL
Hello Alex,
On Nov 24, 2007 8:06 AM, Alex Balashov [EMAIL PROTECTED] wrote:
If you get a chance and take a look, I would appreciate it.
First of all, thank you for sharing your experiences on how you setup
Fax to E-Mail.
I am currently trying to figure out on how to setup an inter-Asterisk
Great thanks steve and bj. As long as its normal I guess I can deal
with leaving it at the default. I was just concerned it could be an
error with the line, when I first hooked up the t1 I noticed the line
going up/down/up/down for 4 or 5 cycles before finally working.
Is there a reason it
Asterisk 1.4 does have this ability natively. However, it is somewhat
limited in its flexibility / in terms of what I can do with it, and
I have gotten reports that HylaFAX works better. I haven't actually
done a comparison between the two.
Being someone who hates 1.2, I was strongly tempted
Not entirely. :-) Your deployment involved a lot more FreePBX than I
would have liked, not to mention some unsolicited WorldComputingGrid
clients.
I mostly just tried to take all the information already out there on the
voip-wiki, etc., follow it, and then sum it up more coherently.
On Fri,
Problem solved. Disabled usb serial ports and other unnecessary
hardwares built-in.
On Nov 23, 2007 2:38 PM, Mark Quitoriano [EMAIL PROTECTED] wrote:
On Nov 19, 2007 2:31 PM, Mark Quitoriano [EMAIL PROTECTED] wrote:
On Nov 19, 2007 12:10 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
I think it is just there kind of as a handshake or ping if you will.
Just saying, hey I am still here, are you?
Thanks,
Steve
Michael J. Liberatore wrote:
Great thanks steve and bj. As long as its normal I guess I can deal
with leaving it at the default. I was just concerned it could be
Michael J. Liberatore wrote:
Would this be normal? Could this be a problem with the line?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Friday, November 23, 2007 8:20 PM
To: Asterisk Users Mailing List - Non-Commercial
I made a little write-up that attempts to synthesise a lot of the
information out there about how to get HylaFAX working with Asterisk
by way of IAXmodem for inbound faxing:
http://blog.evaristesys.com/?p=24
Of course, there are bound to be some things I've left out or are grossly
in need
M == Mike [EMAIL PROTECTED] writes:
M Thanks Benny, that's great info that I didn't think of. So we
M want to send the number out, but notify our provider that it
M shoud be hidden unless it's a call to 911 or something.
M What we have is a SIP connection, not a PRI, is there anyway to
M do
You can try MOR FREE - it has nice gui and is very fast.
LiveCD is available: http://www.kolmisoft.com/mor/content/view/83/95/
It is covered in extensive manual:
http://www.kolmisoft.com/mor/component/option,com_remository/Itemid,40/func,
fileinfo/id,25/
And yes - it's FREE as name suggests.
Wow, what a long post :)
I must confess I did not read slowly, but, I was very interested
because I was the one adding the Dial even feature to Asterisk, and I
remembered it was a mess. From reading Asterisk code one can learn
some stuff. I found this piece of code:
Digium Cards have been just great on my experience and their support has
been simply the best one, via IAX (free Call) Remote Acess and hardware
config review and troubleshooting.
Many Thanks to Digium and their official reseller for Portugal and Spain
Avanzada7 great work!
Best regards,
Marco
Good evening,
Have you got any idea which prepaid application will be the best to do
simple prepaid calls with a MySQL storage...?
PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch
Thanks
___
--Bandwidth and Colocation Provided by
hi,
I use asterisk as a gateway which forwards external calls from pstn to
my internal sip network.
all sip signaling is passed to sip proxy.
I also use asterisk as a voicemail server.
everything works well when calls are passed to asterisk from local network.
but when calls are forwarded from
Fernando Berretta wrote:
Hi,
I'm trying to install g729 codec in an Athlon 64 x2 Dual core processor
4000+ but.. all packages I've download haven't worked. Could someone
please let me know what package should I download ?
Did you register it? It is not free to use.
Ugo
Jakub Syrek wrote:
I thing there was an error in last version of my macro, correct one (i
hope):
Just test it :)
[macro-call]
;sip1 - firs channel from sip outgoing cals operator
;sip2 - second channel from sip outgoing cals operator
;sipn - N channel from sip outgoing cals operator
See the doc/queues-with-callback-members.txt - it has good samples of
GROUP_COUNT and OUTBOUND_GROUP commands.
Regards,
Atis
According to this i wrote macro like this below, is it correct?
[macro-call]
;sip1 - firs channel from sip outgoing cals operator
;sip2 - second channel from sip
Hi,
I'm trying to install g729 codec in an Athlon 64 x2 Dual core processor
4000+ but.. all packages I've download haven't worked. Could someone
please let me know what package should I download ?
Best Regards,
Fernando
[EMAIL PROTECTED] modules]# cat /proc/cpuinfo
processor : 0
Hello.
Is it possible to check if SIP chanell is busy in asterisk?
I have N accounts from my provider and i can dial only one call per account.
I wanto my asterisk to check if first acount is busy, if yes try second and
so on..
I was wondering if ChanIsAvail will suites my needs but i have read
Ask for your telco to enable polarity reversal for these lines. Then enable
hanguponpolarityswitch in your zapata.conf.
About crosstalk I don't have any idea. Maybe a telco or cabling problem...
Sds,
Gustavo
Date: Fri, 23 Nov 2007 02:33:34 -0800From: [EMAIL PROTECTED]: [EMAIL
Robert Lister wrote:
Hello,
I think I have encountered an odd bug in Siemens C460 IP/dect handsets,
which is a bit annoying, and I'm not (yet) sure how to get round it without
lots of hacks.
Basically, on all external incoming calls, we set:
exten = s,n,SIPAddHeader(Alert-Info:
Dear all
I have TDM808B 8 port FXO it is configure perfectly but i got some
problem of incomming phone Hangup and callerid display problem
i am going to explain you the issue i have install asterisk 1.4 and
i have 100 of SIP phone now everything is fine but problem is
PH == Paul Hales [EMAIL PROTECTED] writes:
PH The dialplan command 'setcallerpres' is also good.
Please don't toppost, it makes quoting hard.
Anyway, setcallerpres has the disadvantage for SIP that the number is
not sent at all. This means that the upstream provider cannot give the
number to
I spent several months trying to figure out something similar to this
myself a while back. The solution I came up with finally really works
great, and I think it should work for you too.
Once the incoming caller is in the dialplan, issue a Dial() command using
both the m option and the M()
No functional FreePBX, I just used the ISO for a quick linux install and
World Community Grid is a better benchmark than bogomips.
Neither of which have any bearing on how I setup Hylafax and Asterisk,
otherwise, great job of reverse engineering what I did and documenting
it as your own
On Nov 23, 2007 11:10 AM, Vincent wrote:
On Wed, 21 Nov 2007 15:45:35 -0500, Baji Panchumarti
STAT() and record() are doing exactly what they are
supposed to. Use the s flag to fetch the file size. You
have to try a few hangups and figure out a minimum
file size that qualifies as a
[EMAIL PROTECTED] ~]# strings /usr/lib/asterisk/modules/chan_zap.so | grep
polarity
...
hanguponpolarityswitch
this is the output of my chan_zap i have put
handuponpolarityswitch=yes in zapata.conf but still it is not working
anyway i will talk 2 telco and figure out what is the problem
Well it's not working as it should. Every call go to Dail(SIP/sip1) and if
no one respond then to the next one :(
Arkon
- Original Message -
From: Atis Lezdins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday,
Anybody here have experience they could share on this switch?
Thanks,
Jon
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Fri, Nov 23, 2007 at 02:33:34AM -0800, satish patel wrote:
Dear all
I have TDM808B 8 port FXO it is configure perfectly
but i got some problem of incomming phone Hangup and callerid display
problem
i am going to explain you the issue i have install asterisk
Jakub Syrek wrote:
Hello.
Is it possible to check if SIP chanell is busy in asterisk?
I have N accounts from my provider and i can dial only one call per account.
I wanto my asterisk to check if first acount is busy, if yes try second and
so on..
I was wondering if ChanIsAvail will suites
Ugo Bellavance wrote:
Ugo Bellavance wrote:
Hi,
On my linksys/sipura phones/ATA, there is a setting called NAT
Mapping Enable and another called NAT Keep Alive Enable
These settings must be on in my setup so that my phones/ATA remain
connected to my * server. My setup is:
Home
Anyway, setcallerpres has the disadvantage for SIP that the
number is not sent at all. This means that the upstream
provider cannot give the number to emergency services.
In the regular phone network, CallerID is always sent, but
showing it is suppressed just before it reaches the
Thank for co-operate
[EMAIL PROTECTED] ~]# strings /usr/lib/asterisk/modules/chan_zap.so | grep
polarity
polarity
polarityonanswerdelay
answeronpolarityswitch
hanguponpolarityswitch
Setting IDLE polarity due to ring. Old polarity was %d
Answering on polarity switch!
== Starting post polarity
For testing purposes you can try one of these:
http://kvin.lv/pub/Linux/Asterisk/
Mindaugas Kezys
http://www.kolmisoft.com
Advance Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fernando
Berretta
Sent: Friday, November 23,
I thing there was an error in last version of my macro, correct one (i
hope):
[macro-call]
;sip1 - firs channel from sip outgoing cals operator
;sip2 - second channel from sip outgoing cals operator
;sipn - N channel from sip outgoing cals operator
;ARG1 - outgoing telephone number
exten =
On Wednesday 21 November 2007 12:13:41 Matt wrote:
On Nov 21, 2007 11:45 AM, Tilghman Lesher
[EMAIL PROTECTED]
wrote:
On Wednesday 21 November 2007 09:09:13 Matt wrote:
I have installed Asterisk with FreeTDS many times before (this same
Asterisk and same TDS version)... but today when
On Thursday 22 November 2007 07:03:17 bilal ghayyad wrote:
Is Digium the best telephony cards to be used with
Asterisk? The prices are some how high, any
suggestion?
Actually, compared to other telephony cards, Digium's cards are among the
least expensive on the market.
--
Tilghman
Why would anyone hate the most stable version of Asterisk?
What is ABE using these days? If it is not 1.4, I wonder why? Maybe so
all the free developers and eager and silly early adopters can iron out
the bugs, submit patches and sign away their rights. I am sure if they
are not using 1.4
On Nov 20, 2007 6:13 PM, Philip Prindeville
[EMAIL PROTECTED] wrote:
Yeah, I looked at LinksysSPATFTPProv.pdf... It doesn't say, however,
how to get the phone's configuration out as a flat XML file.
Only how to push the file back into the phone.
wget
There are many reasons to buy digium cards, mainly digiums owner creating
asterisk and all. so when i asked myself your question when starting with * i
bought them. well, i myself have had bad luck with their products,2 failed out
of warranty, and the others have bad echo and random weird
Would this be normal? Could this be a problem with the line?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Friday, November 23, 2007 8:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Wrong. Set resetinteral if it is too annoying but it is normal behavior
although I remember it causing issues with some people in Italy if
memory serves me correctly.
From the wiki
*resetinterval*: sets the time in seconds between restart of unused
channels, defaults to
3600 minimum 60
FYI, World Community Grid is great for burn in testing on boxen. Just
run it for a couple days and see if you can fry the CPU(s) or cores (not
to mention that it is a good cause)
Anyone wishing to join the Asterisk team can do so here http://
Actually if you rule out all the clone tormenta cards (nothing wrong..
but very dated design... I wouldnt buy one today) the Digium cards
aren't too expensive. Those tormenta cards are the ones you see for
$300-400 typically.
Some people like Digium others Sangoma. Personally I'm a Sangoma man.
Is there a reason it resets? Aka does it serve any kind of purpose?
Just curious: what protocol variant (i.e. 4/5ESS, DMS, NI2, etc.) are
you using? Also, which carrier? Finally, have you turned on PRI
debugging to see if it is the telco that is requesting the restart? In
some cases the telco
Michael J. Liberatore wrote:
Alex, I thought asterisk 1.4 supports faxing internally now without the
need for extra software? Is your solution a different one? I have no
experience with faxing yet but plan to soon, that's why I ask and will
read your blog entry.
You need extra software,
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