PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, December 20, 2005 12:01 PM
Subject: RE: [Asterisk-Users] Asterisk Broadvoice help??
Thanks Steven
Works great.
They should put a little more detail in the setup page
What is the proper way to email to multiple email addresses.
I have been intending to also email my cell phone when there is a message, but
have yet to try different options like comma,
semicolon, etc.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope
Having the strangest time getting the uip200 to work with Asterisk.
We can send outgoing calls, however we can not receive phone calls.
I have tried listening to all of the recommendations in this list such as
setting the nat=never in the sip.conf and that didn't work at all (phone
stopped
, Steven Job wrote:
Having the strangest time getting the uip200 to work with Asterisk.
We can send outgoing calls, however we can not receive phone calls.
I have tried listening to all of the recommendations in this list such
as
setting the nat=never in the sip.conf and that didn't work at all
I have:
1) nat=route
2) dtmfmode=inband
Tried that and no luck. :-(
Yes, I have local and remote (behind NAT) UIP200.
You also need to make sure to specify in the [general] section:
externip
localnet
That really wouldn't matter for me since my Asterisk box is not behind NAT.
Only the
would only be a solution if it handles the faxes like
ZAP and doesn't pocketsize them.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of having
a better past
I agree.
I am sure it is a programming issue with DTMF on Stations vs. Trunks.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of having
a better past
I requested Polycom firmware from VoipSupply and they send it to me within 4
hours.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of having
a better past
was for)
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of having
a better past.
---- --- - - - -- - - -- - - - --- - -- - -
--- - - -- - -- -- - --
Steven [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED
I was wrong.
This patch is for channels/chan_zap.c
I have been hesitant to go to 1.2.1 without config testing.
Should I have any negative issues going from 1.0.9 to 1.0.10? ( I have to see
if the changes are in the 1.0.10 version of
channels/chan_zap.c)
--
--
Steven
It looks like http
at asterisk 1.2.1.
If I backup my configs and upgrade to libpri 1.2.1 and asterisk 1.2.1, is it
just as simple as reinstalling the older versions if it
isn't working correctly?
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of having
a better past
I do not think that speex is installed by default.
run show translations in asterisk and see what you get.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
I assume that it would have to use a key sequence (Ctrl+Shift+A, etc.) that
does a copy of whatever is highlighted (in any app that supports text copy)
and pastes it into the dialing app. (either Tapi or softphone)
--
--
Steven
May you have the peace and freedom that come from abandoning
/var/log/asterisk/full text file may give you a more specific error.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
are upgrading your production systems.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
Does anyone know if there are plans by Digium to have a PCI-X T1 card?
If so, any timing information?
Although throughput is higher on PCI-X, is interrupt processing any
better/worse than standard PCI?
--
--
Steven
May you have the peace and freedom that come from abandoning all hope
Yes, I meant Express.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - -- - -- -- - --
Kevin P. Fleming [EMAIL PROTECTED] wrote
What is the best way to get the busypattern tone ms on/ms off from my
Panasonic DBS?
I am getting false hangups.
I just added busycount=8, but I figure I should add the busypattern= as
well.
Any advise on the best toll to record the tone into to get the ms on/off
readings?
--
--
Steven
May
into my Nagios
server, but I am only doing notifications via email and modem/TAP.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
Hi
In my
extensions.conf shown below when the external
number 123 is dialed it goes to phone ext1. I can forward to another phone
using exampleline below but I would only like to forward after 5pm and
before 9am. How can this be done?
Thanks for your help.
exten =
Note:
I upgraded Zaptel to the 1.2 stable and changed digits.h line to #define
DEFAULT_DTMF_LENGTH 250 * 8.
I was told that there is still a problem.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
=no
accountcode=I
musiconhold=default
channel = 1-23
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
seem the best if I
could get it to work.
Has anyone here dealt with DISA on a Panasonic DBS?
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
I am using http://www.gmane.com/ with my newsreader.
You still have to be a list member to post.
You can then turn on the vacation option in the list manager to stop
receiving emails.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
I read that it was supposed to integrate with X10 modules and the that the
@home was reference to home automation.
That being said, I have never seen any X10 specific functionality.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
You may have already done this, but my first approach would be to look hard
at the Vocal Data switch and see if you can disable G723 support on the
switch.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
So does this problem only surface with delete=yes?
I am using 1.0.9 and do not have the second comma.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
I do not know if asterisk uses standard regexp, but in regexp you would use:
[(201)(202)(203)(205)(206)]
This would match any of the groups () of numbers.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
Could you just use a different start number?
9 to dial out. 8 to dial out with blocked callerID.
Then just preface the callerID block code for the Telco.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
jitterbuffer=no and when jitterbuffer=yes. I have run zttest and am getting pretty much 100% accuracy from the card.
Does anyone have any ideas what the problem could be?
Many thanks
Steven
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Note: http://www.citel.com/products/handset_gateways/ sells a SIP handset
gateway that will let you still use your Digital phones.
We used it for our old NEC phones.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
If, on the other side, asterisk continue accepting incoming call, how can I
be sure that I wll reach a convenient moment ?
If you are not sure if it will even reach a convenient moment, you will also
not get a chance to run stop now.
--
--
Steven
May you have the peace and freedom that come
successfully installed the same version of Zaptel. Any ideas what the problem could be?
Thanks
Steven
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in, where in a group of 5
trunks, asterisk will use the next unused trunk.
But SIP and IAX do not seem to get tagged as in use as far as I can see.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
I used to use ConText, but now I prefer Notepad++.
Both are free and for Windows.
They both let you easily edit Unix formatted text files.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
Title: Re: IAX2 calls being droppped
Hi
Thanks for the reply. The host can still be pinged. In fact, it is usually only 1 user which is dropped from the session while other users are in the session. I dont think it is a problem with routing to the host.
Steven
Message: 13
Date: Sat, 12
PROTECTED]:4569/3 (type = 2, subclass = 1024, ts=655380,
seqno=177)
This error is pretty erratic. It mostly happens the first time you try to
dial, but also seems to sometimes be happening in the middle of a
conversation. Any ideas what the problem could be?
Many thanks
Steven Langley
attachment
I apologize if this question has been asked before. Did something change
with the behaviour of the 'sip show inuse' command between 1.0.9 and
1.2-rc1? I used to be able to see a list of extensions and the number of
in/out calls. Now it just reports:
asterisk*CLI sip show inuse
* User name
I use a newsreader pointed at gmane.org.
It is agregated and only uses my internet connection when I tell it to.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
SO is he definitively saying that the asterisk software is not involved
here? (listening or regenerating tones)
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
Hi
That seems to work fine now using Z for the last line. Thanks very much for
your help and explanations.
Incidentally, we are using VoipTalk but are looking to trial another
provider as have been experiencing the occasional call cut out and quality
issues. Out of interest, do you have
and errors. (etc.)
If I could grep for a phone number in the log, get an ID tied to that thread
(???) , then grep for that ID, I could see only what I want to see much
faster.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
.
Please advise if I can get echo can off with Zap to Zap bridge, but have
echo can on with Zap to Sip bridge.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
Thanks for the info.
I had tried that one before, but was missing that I had to completely
restart asterisk when changing those settings.
It is working as expected now.
Thanks.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
I am also curious if anyone is using the larger Dell servers.
I currently have a 1U 1750 (2 PCI) with two TE110P cards and now have the
need for some FXO/FXS ports.
I am a bit hesitant to just buy a larger box and then find out about BIOS or
Interrupt issues after the fact.
--
--
Steven
May
Thanks, thought that should work but had a type error which have now
corrected. One further question, how can I set up a line so that if 440 is
dialled before a number the 0 is taken out so only 44 is actually used?
Thanks again.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
O'reilly had a book out before the docs team wrote theirs.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - -- - -- -- - --
Leif
Hi
I have inserted the lines you suggested but Asterisk keeps the 0 when
dialling with all alternatives. Also, I am unsure what the phrase
${EXTEN::2}${EXTEN:3} does? Could you explain this abit?
My extensions.conf is:
[default]
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
Hi
Currently, in
extensions.conf I have:
exten =
_0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})exten
= _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
This enables numbers
to dialledstarting with 0 and 00 and changes them to start with 44.
How can I configure my extensions.conf to
that the delay might be
a client issue instead of a server issue. What is the best tool to use to run
tests on my server and clients to narrow down the source of the delay?
Many thanks
Steven
Date: Wed, 12 Oct 2005
10:41:33 + (UTC)
From:
[EMAIL PROTECTED] (Tony Mountifield)
Subject
Hi
Thanks for the reply
I do actually use the |q
option to disable the enter/exit sounds.
Steven
Message: 15
Date: Thu, 27 Oct 2005
10:25:32 -0500
From: Eric
\ManxPower\ Wieling [EMAIL PROTECTED]
Subject: Re:
[Asterisk-Users] Re: delays with IAX2 and Meetme
I'll give it a shot.
Do you compile it with Zaptel running or diasable it and reboot first?
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
Receiving faxes do not generate a fax tone.
They will generate a modem tone when answered if that is usable/detectable.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
It worked.
Thanks for the 1.2 info.
Hopefully it hasn't created any unforeseen issues.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
Will this work if I am using text file configs?
I started with AMP, but didn't like the limitations.
I disabled the DB config parts, but still use the other features of AMP.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
if anyone has had this issue and figured it out.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
in the Panasonic, so I do not have CID and can
not make outbound rules from that info.
A replacement PRI card is $1900 and because Asterisk is to replace the
Panasonic, I do not want to invest in it.
Please advise.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope
Great.
All of the references I read mentioned the card specifically, not zaptel or
asterisk.
Thanks for the info.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
Hi!
I am trying to configure a series of Grandstream GXP2000 phones. I have
downloaded the Grandstream Configuration Generator v1.3 to generate the
cfgmacaddr files that the phone expects to see on the tftp server. I
watch my tftp server's diagnostic output, and verify that it is
downloading
Hi
I am using Asterisk
TAPI driver with Outlook and have many contacts with numbers listed as +44 1XXX
XX which is international dialling for UK. My Asterisk context is as
follows:
[outlook]
exten =
_0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten =
_00.,1,Dial(IAX2/[EMAIL
the DigitTimeout and try again)
exten = s-gathermoredigits,2,DigitTimeout,5; Increase the 'finished
dialing' timeout to 5 seconds
exten = s-gathermoredigits,3,WaitExten(4) ; and give the caller 8
seconds overall to do their thing
--
--
Steven
May you have
Hi
Is it possible with
Asterisk to tellthe called party which number was dialled by the
caller? Or in place of the number dialled have a description such as
'Sales' or 'Accounts'? Ideally, I would like to show a description
corresponding to the number dialled followed by CIDName.
How might
})
- Original Message -
From:
James Steven
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 19, 2005 9:33
AM
Subject: [Asterisk-Users] DNIS/DNID
Hi
Is it possible
with Asterisk to tellthe called party which number was dialled by the
caller? Or in place
Hi
Is it possible with
Asterisk to tellthe called party which number was dialled by the
caller? Or in place of the number dialled have a description such as
'Sales' or 'Accounts'? Ideally, I would like to show a description
corresponding to the number dialled followed by CIDName.
How might
in again. When dialing in
again the delay seems to go.
It seems to me as though as soon as the server registers a
delay from a participant, then it causes delay on all further packets from that
participant.
Does anyone have any ideas what the problem could be?
Many thanks
Steven
I am using http://www.nathanpralle.com/software/hoodahek.html for this.
Just this first part, not the notification.
It is working great for looking up my manual CID entries and will also add
new numbers to the list so I can easily modify them as well.
--
--
Steven
May you have the peace
I bought some USB soundcard/handsets from them with
no issues.
I did not deal with them on any PBX or config
issues though.
-- -- Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
I just copied the *98 extension to the extension of one of our DID numbers.
So now if I dial 5686 from inside or 1-xxx-xxx-5686 from outside, I get the
same prompts as dialing *98.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
AstriCon Update: Only Two Weeks To Go!
October 12 - 14, 2005
Anaheim, CA
AstriCon 2005 starts two weeks from today. We now have a complete
roster of speakers covering Asterisk from soho to carrier. We've
added the Code Zone, a working lab with a full compliment of VoIP and
TDM equipment. We
I am also looking for this functionality for an emergency support number.
Except, I want to notify a different person after 7 minutes, then a third,
then back to the first, etc. until the message has been listened to.
--
--
Steven
May you have the peace and freedom that come from abandoning
Can I ask how you are providing calls to us domestic numbers for free?
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
/Teliax1${EXTEN}) ; Will it skip this if it is in use or down?
exten = _9.,3,Macro(dialout-trunk,1,${EXTEN:1},) ;or could be
Dial(Zap/g1/${EXTEN})
exten = _9.,4,Macro(outisbusy) ; No available circuits
?
--
--
Steven
May you have the peace and freedom that come from abandoning all hope
, but my concern is that if
there is too much traffic, I will miss the alarm, or if there is too little
traffic, I will keep getting notified even if it is fixed.
Are there any triggers in Asterisk that can run a script if this error
occurs?
What are others doing for this?
--
--
Steven
May you
OK Great, I'll give it a shot.
I did find this other option
http://archives.free.net.ph/message/20050309.013714.2d1bf446.en.html , but I
do not really want to imbed this info in the asterisk database if I can have
it external. (note: this other option did work when tested)
--
--
Steven
May
prefix
exten = s,4,SetCIDName(${CALLERIDNAME:${LEN(${RGPREFIX})}}) ; strip off
prefix
exten = s,5,AGI,dialparties.agi
exten = s,6,NoOp(Returned from dialparties with no extensions to call)
exten = s,7,SetVar(DIALSTATUS=BUSY)
exten = s,10,Dial(${ds})
--
--
Steven
May you have the peace
I found configuration via MySQL too limiting.
I went back to text files.
I do not know if it was realtime or not, it was the sql in [EMAIL PROTECTED]
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
[2223]: disabled echo cancellation on channel 47
Sep 18 11:45:35 VERBOSE[2223]: -- Hungup 'Zap/47-1'
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
On 9/16/05, Steven Sokol [EMAIL PROTECTED] wrote:
Hi,
I'm taking a straw-poll to see who out there is planning on going to
AstriCon. I would like to hear from both new members of the community
and gurus. What kinds of things would you like to see at an Asterisk
Conference? What topics
.
T.
FYI - AstriCon is October 12 - 14 in Anaheim. For more information on
what we currently have planned, see the web site (listed below).
Thanks,
Steve
--
Steven Sokol
Sokol Associates/AstriCon
Ask Me About AstriCon 2005!
http://www.astricon.net
On 9/17/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Sat, 17 Sep 2005, Steven Sokol wrote:
(Anybody out there want to volunteer to bring in the hardware?)
I'll bring some Digium hardware, Sirrix boards. Assuming there'll be some
security against them disappearing...
Cool
something like this.
Olle and I have talked with the Digium guys about doing road shows
for Asterisk. Not so much of a Conference, but a regional or even
local event that's a combination of our one-day introductory class
(yes, commercial -- bad Steven, bad!), a users meeting (code session,
gab session
On 9/17/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Steven Sokol wrote:
That is an awesome suggestion! We'll do it! We have a room we've
labeled the Email Garden. We'll rename it the Code Domain or
something and try to get at least one guru to man the desk in there,
dispensing
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me know if
either of those suggestions would be a big positive or a big negative.
(Remember that we would be on a coast either way.)
Thanks,
-S
--
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Ask Me About AstriCon 2005!
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,
Steven
/edg
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!).
-S
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like to set it up to
IGNORE the distinctive ring pattern that I have for a fax machine.
Many thanks
Brad
--
Steven Premeau [EMAIL PROTECTED]
-
Gebt mir endlich einen
group) fodder? What parts of Asterisk require the most
attention?
FYI - AstriCon is October 12 - 14 in Anaheim. For more information on
what we currently have planned, see the web site (listed below).
Thanks,
Steve
--
Steven Sokol
Sokol Associates/AstriCon
Ask Me About AstriCon 2005!
http
On 9/16/05, Brian Roy [EMAIL PROTECTED] wrote:
On 9/16/05, Steven Sokol [EMAIL PROTECTED] wrote:
Hi,
I'm taking a straw-poll to see who out there is planning on going to
AstriCon.
Enjoyed it last year, but putting it on the west coast seems to be pretty
restrictive. I
and the people with Asterisk
extensions know that they may not always have service, but when I make this
change, the Panasonic MUST still be fully functional.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past
Title: BRI debug, national ISDN speech call problem
hello,
I have a Junghanns QuadBRI card in my asterisk server. I'm able to dial connect to local numbers through the ZAP interfaces however when I try to dial national numbers with the according area code the connection fails, an intense
2005 site: Speak
At AstriCon). We will begin making final selections this week.
_Any Questions?_
If you have any questions about AstriCon 2005, please let us know.
Email [EMAIL PROTECTED] We want to make this the best AstriCon yet.
Thanks,
Steve
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Steven Sokol
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Sokol
, Aheeva,
Snom, Cylogistics, Sayson, Xorcom, and more. Contact us today for
information on exhibiting: [EMAIL PROTECTED]
-
Steven Sokol
CEO/Manager
Sokol Associates, LLC
Ask Me About AstriCon 2005!
http://www.astricon.net/
AstriCon is produced by Ipsando in partnership with Digium
the following link:
https://www.astricon.net/2005/hotel.shtml
(Click on the Special Rate link on the Hotel Travel page to
reserve your room.)
--
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http://www.astricon.net
extension Steven Hall
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extension
Steven Hall
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with my configuration
Steven
Hall .
[EMAIL PROTECTED]
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Even if you get it working the PSTN line is a gateway chances are its a
loop start and will have problem with disconnect supervision issues.
go to http://www.voip-info.org and search for disconnect supervision and
see.
Regards
steve kalcevich
Carlos Trallero wrote:
Hello,
I have
Hi there
I am in the process of setting up a production Asterisk
server, which will mainly be used for meetme conferencing. I am considering
running a firewall, but wondering whether this will slow Asterisk down if all
packets are being scanned. Any ideas?
Many thanks
Steven
to go through.
[ Download IAX Phone Pro ]
https://www.astricon.net/phone/ipbeta.php
Steven Sokol
CEO/Manager
Sokol Associates, LLC
Ask Me About AstriCon 2005!
http://www.astricon.net/
begin:vcard
fn:Steven Sokol
n:Sokol;Steven
email;internet:[EMAIL PROTECTED]
tel;work:816.822.1807
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