in at once (three people call the SIP
number).
Jason
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This sounds like a winner, are you using voicepulse?
Also, I downloaded the iso for AAH 1.5. Are there any noteworthy bugs
that are fixed? I don't really like to upgrade unless I need to.
Jason
Paul wrote:
connect.voicepulse.com allows up to 4 calls at a time coming into an
$11/month DID
Does anyone have any experience with Teliax for inbound IAX?
Jason
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Make sure you have turned off VAD as asterisk does not support Silence supperssion.
Jason
On 9/21/05, Juan Jose Comellas [EMAIL PROTECTED] wrote:
Have you tried upgrading the firmware? I had several problems with theoutbound volume of these phones until I upgraded them.
On Tuesday 20 September
On 9/19/05, Frank Tarczynski [EMAIL PROTECTED] wrote:
I'm new to asterisk and need some help with ideas to handle thisconfiguration question.I am trying to establish a termination point/DID number in another
country.I am currently running Asterisk CVS-HEAD.My foreign provideruses SIP and
This would be super-fantastical!!!
With all of the other conferences going on, I can only get away so much. I
love the idea of a webcast...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Saturday, September 17, 2005 8:37 PM
To:
That's what I have used...works until you change it. ;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rene Kluwen
Sent: Friday, September 16, 2005 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Hi,
I tried to install these cards using FC3 and FC4 on
various motherbords, but to fail.
I sent email to digium several times, but no response.
I think these cards are not for production use yet.
Regards,
Jason
__
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Tired of spam
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Thursday, September 15, 2005 7:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Is digium supporting new te405p and te406p
install?
Hi,
I tried to install these cards using FC3 and FC4
I tried both 1.0.9 and 1.2beta.
I couldn't see any interrupt from /proc/interrupt.
My email server has no spam filter.
--- Jason Walker [EMAIL PROTECTED] wrote:
I have not been able to get * 1.0.9 on a FC4 box...I
have an older IBM
server just waiting and try it every so often. When
I am
?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Thursday, September 15, 2005 8:54 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Is digium supporting new te405p and
te406pinstall?
I tried both 1.0.9 and 1.2beta.
I
I was happy with FC3, old te405p and * 1.0.7.
I've been thinking that kernel 2.6 is more stable and
secure.
--- Jason Walker [EMAIL PROTECTED] wrote:
I kept running into compile errors when dealing with
my Compaq (it is an
older quad 700 Xeon...not sure of the model number).
Once I dropped
I am curious...are you saying to use SIP locally and IAX from point to point
(over a WAN or VPN tunnel)? With that in mind, do you think that using a
lesser compressed codec over the IAX trunk would give an okay amount of
bandwidth savings?
Thanks.
-Original Message-
From: [EMAIL
5000-600?
Do you mean 5060? That is the port for 5060. 1-2 is
for RTP.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B.
Asterisk UsersSent: Sunday, September 11, 2005 12:46 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP
Connection
Hi,
I've installed an TE406p, asterisk1.2 on tyan opteron
board.
After installation there is no interrupts from TE406p.
Is this board stable?
Should i change * version to 1.0.9?
Regards,
Jason
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Thanks.
--- Alexander Lopez [EMAIL PROTECTED] wrote:
Did it take an interrupt??
Whats does /proc/interrupts say??
Did you check your span= settings in zaptel.conf??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Jason Kim
Sent: Sunday
]
[mailto:[EMAIL PROTECTED] On
Behalf Of Jason Kim
Sent: Sunday, September 11, 2005 7:48 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TE406p no interrupts
Hi,
I've installed an TE406p, asterisk1.2 on tyan
opteron
board.
After installation there is no interrupts from
reconfiguration (ztcfg) let me
know.
If you get stuck.. let me know.
Regards
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Jason Kim
Sent: Sunday, September 11, 2005 8:14 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] TE406p
: [1] SMP
astpbx kernel: CR2: a0362081
Regards,
Jason
--- Boris Bakchiev [EMAIL PROTECTED] wrote:
You should have just done this:
rmmod wct4xxp
rmmod zaptel
modprobe wct4xxp
It will do the same thing
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk
) with the pci_register_driver.
Regards
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Monday, 12 September 2005 11:28
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] TE406p no interrupts
I modified
channel = 17-31
channel = 32-46
channel = 48-62
--
Thanks, have a great holiday!
Regards,
Jason
__
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PRI channels will reset when not in use throughout the day. A reset on a
channel should not happen when that channel is in use. This happens all the
time on my PRI circuits (TE110P and TE410P). From what I gather, it's
somewhat like a handshake for the D chan between the cpe and net sides.
?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Saturday, September 10, 2005 5:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] TE110P reset
On Saturday 10 September 2005 19:40, Jason Walker wrote:
PRI channels
out your ENTIRE communications network?
-A.
Sage advice, but out of curiousity what happened to Digium's T3 card
(the DS3000P)?
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
Matthew Boehm wrote:
Jason Becker wrote:
Sage advice, but out of curiousity what happened to Digium's T3 card
(the DS3000P)?
IIRC, Digium's T3 card isn't expected to be channelized. Also, IIRC
it will have no on-board EC and no on-board encoding so I can't imagine
the machine you
Matthew Boehm wrote:
Jason Becker wrote:
Hmm, looks like someone in the know needs to update the wiki:
http://www.voip-info.org/tiki-index.php?page=Digium+DS3000P
Wow. Guess I'm not.
Matthew, I in no way meant to imply that you are not in the know. I
guess what I meant to say
txgain=-4.0
group=1
channel = 1-15
channel = 17-31
channel = 32-46
channel = 48-62
--
Anyone please help?
Regards,
Jason
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,
extensions_additional.conf, extensions_custom.conf or indeed oh323.conf.
- are relevant.
Please search the [EMAIL PROTECTED] forum and/or amportal list forum for
more info on these topics.
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
I would be willing to give time and be able to setup / manage some
devices. I was thinking that you could use standard POTS phones to a
adtran tsu600 asterisk t1 to fxo to pots. the wifi phones would be nice
but i think they would tend to walk off.
JasonOn 9/2/05, Damon Estep [EMAIL PROTECTED]
I installed/ran both MozPhone and DIAX but did not see in the debug any
information of the URL I sent. Perhaps the real question is: if
optionalurl is used, how is the url sent to the device(s)?
Has anyone applied this within a solution and is willing to share their
experience?
Thanks!
Jason
to ver
1.0.7.
For the client side, I am testing MozPhone and DIAX.
MozPhone ver 0.9.2-200507111326; IAXClient: CVS-2005/07/03; Jslib: 0.1.290
DIAX is version 0.9.15a; same IAXClient as MozPhone.
Am I dealing with a compatibility issue more so than anything else?
Thank you for your responses.
Jason
for us works
great and I would encourage anyone to use it.
As a side note, Michael is a great guy to work with and is extremely
reliable in supporting this software.
Thanks,
Waldo
On Aug 31, 2005, at 10:47 AM, Jason Walker wrote:
I installed/ran both MozPhone and DIAX but did not see
: Wednesday, August 31, 2005 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd
Jason Walker wrote:
I am testing this on * ver. 1.0.7 (I have another box with 1.0.9 and
another
one with CVS HEAD). Is 1.0.7 too
Jason Walker a écrit :
Now I don't feel so inadequate ;)
This is exactly what I am doing. Perhaps there is more to this particular
option.
Here is more information -
I am testing this on * ver. 1.0.7 (I have another box with 1.0.9 and
another
one with CVS HEAD). Is 1.0.7 too old
if there is a
softphone that supports this. The only thing that seems to happen is the
queue_log is updated with whatever is placed in the optionalurl
location of the Queue command.
Thank you in advance,
Jason
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--
Jason Becker
Director CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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Is there a way to get answer confirmation via IAX and not only via
ZAP? We get our outbound service via an IAX trunk to our provider so
we aren't in control of their ZAP configs, but ideally we'd like to
be able to achieve the answer confirmation functionality regardless,
especially in
for anyone wanting to see what ports the voice connection runs on:
Internet Protocol, Src: 66.162.X.X (66.162.X.X), Dst: 192.168.1.21
(192.168.1.21)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x80 (DSCP 0x20: Class Selector 4; ECN: 0x00)
Total Length: 116
that the
implementation is subject to change.
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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http
H323/[EMAIL PROTECTED]
OH323/[EMAIL PROTECTED]:
vpb/1-1/$OUTNUM$
--
Jason Becker
Director CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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Try setting your logger.conf to allow full output (uncomment the full
section) and see if there is something specific to the CLI crash.
Be careful though and do not let the logging get out of control, especially
on a big system. The file can get huge.
-Original Message-
From: [EMAIL
Do you have 5 or 6 scripts running against the interface for one instance of
an outside script? Or, do you have multiple connections (outside users)
attempting to run multiple instances of a script that are pulling 5-6 CLI
scripts?
This would exponentially increase the real number of scripts
it to work many months ago - even with help from the developer.
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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to the gateway
(in this case asterisk)
If you still have problems I may be able to dig up some configs for you??
Cheers
Jason
Jason Penton
PhD Candidate
Department of computer Science
Rhodes University
Tel: +27 46 603 8640
Mobile: +27 82 376 6811
VoIP: sip:[EMAIL PROTECTED]
Email: [EMAIL PROTECTED
Shot in the dark
Do you have to dial '9' on your outside line?
Perhaps if you changed your Dial command to this:
[outgoing]
exten = _9X.,1,NoOp(Call for ${EXTEN})
exten = _9X.,2,Dial(Zap/1/${EXTEN:1})
The :1 will drop the leading '9' when it hits the outside. If this is a
regular line,
phone configuration
Jason
Stefan Gofferje wrote:
Mark Johnson schrieb:
Jason wrote:
Hey all, I have set up my cisco 30vip using chan_skinny because
chan_sccp wont register. The problem I am having is, everytime a
call is sent to the phone Skinny/[EMAIL PROTECTED] it rings once
stack
-- Executing Dial(SIP/4437821638-7588, Skinny/[EMAIL PROTECTED]) in new
stack
Found device: jason
-- skinny_request([EMAIL PROTECTED])
-- Skinny cw: 0, dnd: 0, so: 0, sno: 0
chan_skinny: skinny_new: tmp-nativeformats=4 fmt=4
-- skinny_call(Skinny/[EMAIL PROTECTED])
voip*CLI
to it :). not sure exactly what that was about but it works now.
Now i just gotta put a second line in the configuration and try to make
it work
-Jason
Mark Johnson wrote:
Jason wrote:
Hey all, I have set up my cisco 30vip using chan_skinny because
chan_sccp wont register. The problem I
For ZAP cards, you can tell Asterisk to answer calls immediately across
trunks. Does CAPI have the same type of setting? I am not familiar with
Asterisk and CAPI so I am not sure of the options.
In Zapata.conf, setting immediate=yes will make the call drop into the 's'
extension of the
Sergio Chersovani wrote:
Jason ha scritto:
Could someone assist me in configuring this phone. It is saying in
the CLI that its registered and saying its capabilities are recieved
but i got no dialtone on the phone. Thanks
are you using chan_skinny or chan_sccp?
Sergio
Did you setup your T1s as trunk groups?
What channels are set up as d chans from the carrier?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan
Sent: Wednesday, August 10, 2005 4:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Sent: Wednesday, August 10, 2005 5:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco IP Phone 30 VIP
Sergio Chersovani wrote:
Jason ha scritto:
Could someone assist me
Where are the d chans in the trunk group? Which chan?
Here is the example from the zapata.conf.sample
;
; Trunk groups are used for NFAS or GR-303 connections.
;
; Group: Defines a trunk group.
;group = trunkgroup,dchannel[,backup1...]
;
;trunkgroup is the numerical trunk
VIP
those phones don't use .xml like the 7960s
http://voip-info.org/tiki-index.php?page=Configuring%20Cisco%2012SP%20phones
%20with%20Asterisk
On Wed, 2005-08-10 at 16:49, Jason Walker wrote:
The SEP file should be
SEPMACADDR.cnf.xml
You can also use XMLDefault.cnf.xml
These have
Derek,
This address will work fine for communication off list
Jason
Jason Walker wrote:
Misread the type of phone...sorry about that
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten
Sent: Wednesday, August 10, 2005 5:14 PM
To: Asterisk
N, Thats only if you want to use SIP which i dont, I just want to
use the standard sccp that comes with the phones that link to the call
managerand to be bluntly honest..the phone is EOL..cisco isnt gonna
support it anyway. It has been EOL since december of 2000
Jason
Tony Hoyle
Does anyone else have DTMF issues with SIPPhone? When calling into my
DID, and entering, say, 1002. Sometimes it will recognize it properly
(rarely), other times it will receive something different. Such as,
1102 or 1000, etc. Has anyone else been having these issues? I'm
only accepting ulaw
Louie,
On 8/8/05, Tarpo, Louie [EMAIL PROTECTED] wrote:
We encountered the same issue. change dtmfmode=rfc2833 to dtmfmode=inband
and make sure you're using a ulaw connection. If you use a lossy codec, it
will scramble the DTMF tones.
Are you using SIPPhone? When I use dtmfmode=inband,
So the way I understand this is with rfc2833, DTMF is sent out of
band. So does this mean that SIPPhone is interpreting the tones
incorrectly? Asterisk shouldn't be doing any actual tone detection
with this method, right?
On 8/8/05, Tarpo, Louie [EMAIL PROTECTED] wrote:
Yes we are. I just
On 8/8/05, Tarpo, Louie [EMAIL PROTECTED] wrote:
RFC2833 is sent out of band. What's the output on your asterisk console?
I don't see any output during this time on my asterisk console.
Unless there's additional logging I'd need to enable?
Thanks for the help!
-JD-
Was using this monitor line to get soxmix to mix test-in.wav and test-
out.wav into test.wav.
exten = 1200,1,Monitor(wav|/tmp/test|m)
When I start the conference, the * console shows this:
monitor executing ( nice -n 19 soxmix //tmp/test-in.wav //tmp/test-
out.wav //tmp/test.wav rm -f
I guess the problem is with SIPPhone then. I opened a ticket with
them. I'll post their response when I have one.
Thanks!
-JD-
On 8/8/05, Tarpo, Louie [EMAIL PROTECTED] wrote:
I find a verbosity of 10 (asterisk -rvv) gets me adequate logging for
my purposes. I've been really
I'll give it a shot.. Do you know if they have any plans to merge this in?
On 8/8/05, Gary Reuter [EMAIL PROTECTED] wrote:
On 8/8/05, Jason DiCioccio [EMAIL PROTECTED] wrote:
I guess the problem is with SIPPhone then. I opened a ticket with
them. I'll post their response when I have one
All of a sudden, my account doesn't appear to work, or even perhaps
exist with SIPPhone. Is anyone else having trouble?
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I'd like to provide the ability for a friend to conduct interviews
using an asterisk conference and then email them to him when done.
Kinda like a voicemail. There doesn't seem to be one single hook to
be able to do this so I'm wondering what other people have used to
jam this together
I have had to create two different types
of connections depending on what I connect any of the TE4XX cards and the TE1XX
card.
What are you connecting this to?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Dracevich
Sent: Tuesday, July 26, 2005 7:21
If all of your extensions are in the same schema (i.e. 7## or 7###) you
could do this:
Exten = _7XX,1,Dial(DEVICE/${EXTEN})
Exten = _7XX,2,Voicemail(u${EXTEN})
This would allow for any 7## number to call into the extension. ${EXTEN} is
the variable for the extension dialed. I am using DEVICE
Can you post your macro?
Thanks.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim King
Sent: Thursday, August 04, 2005
2:56 PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: [Asterisk-Users] Outbound
Extension problem
New
That would make all callers have to call 720 as there is not other extension
defined. As a result, all calls would go to 720. ${EXTEN} would always be
720.
I don't follow your logic.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday,
Could someone assist me in configuring this phone. It is saying in the
CLI that its registered and saying its capabilities are recieved but i
got no dialtone on the phone. Thanks
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(sipexten,${EXTEN})
exten = 721,1,macro(sipexten,${EXTEN})
exten = 722,1,macro(sipexten,${EXTEN})
and so forth.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 6:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Soft phones or hard phones?
There are many free VOIP soft phones out there.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, August 04, 2005 9:57 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ip phones
Up until today, I have had no issues with receiving faxes in *. One
change I made was that I now have the incoming DIDs "macro"'d since
they all start with 3 (3###).
>From /var/log/asterisk/messages
Aug 2 10:26:58 NOTICE[14938]: Unable to find
a path from unknown to unknown
Aug 2 10:26:58
=no on my DIDs is a better plan.
Perhaps there is a better way? Something I am missing?
Thank you in advance
Jason
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Joseph -
I would love to see something like this if you are willing to share.
Thanks.
Joseph wrote:
Hall, Eric M. wrote:
Looking for a good web app that will show agents that are login to
queue. I tried the operator panel but I'm unable to get the LED to
change color per the doco I
Are you calling an IP or an extension?
JASON WALKER
- Original Message -
From:
someshwarak
To: asterisk-users@lists.digium.com
Sent: Thursday, July 28, 2005 7:37
AM
Subject: [Asterisk-Users] help Windows
messenger configuaration
Hi,
I am trying
errors abound.
If you have any suggestions, I would appreciate any assistance.
Thank you,
Jason
-Original Message-
From: Jean-Denis Girard [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 26, 2005 9:31 PM
To: Jason Walker; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Has anyone had any luck with MozIAX (Mozphone) on FC4 with Firefox
1.6? jslib and moziax install through Firefox correctly - at least
that is the message I get.
I am able to log into the IAX Phone on Windows, however I get an error stating:
--
s helps me to keep track of inbound T1s and
outbound T1s.
Also, you have 2 (2) priorities listed in your
example. You can't really do this.
JASON WALKER
- Original Message -
From:
Angus
Comber
To: asterisk-users@lists.digium.com
Sent: Monday, July 25, 2005 8:11 AM
Any suggestions for IAX phones on Linux (without Wine preferred)?
Thanks,
JASON WALKER
- Original Message -
From: Joseph [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 25, 2005 11:05 AM
Subject: RE
Jason Walker wrote:
Any suggestions for IAX phones on Linux (without Wine preferred)?
Kiax.
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
___
Asterisk-Users
Round robin is designed to alternate between, in this case, the two agents.
At least that is how I understand the comment in the queues.conf file.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev
Sent: Thursday, July 21, 2005 11:18 PM
To:
I *believe* you can append '#' on the end of the dial string to tell
Nortel you are done dialing. I know it works on the Option 11.
Hope that helps!
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Thursday, July 21, 2005 10
On 22/07/05 02:49 +0900, Kuniyoshi Murata wrote:
Hi,
Now, I think I want to disable Asterisk's access to console audio device
based on the logic above. How can I do that?
Make sure the following is in your modules.conf file:
noload = chan_alsa.so
noload = chan_oss.so
of
hardware are you using for FXO?
Jason Stewart
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HTTP uses TCP. Too much overhead. Add SSL on to that and you have a
huge amount of overhead. The end result would be poor and choppy sound
quality.
Jason
On 21/07/05 21:58 +0200, Rob Scott wrote:
For work environments where you only get HTTP or HTTPS access, what is
the feasibility of doing
://www.oreilly.com/catalog/switchingvoip/
It makes heavy use of Asterisk for instructional purposes.
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
Enabling Open Source
that the 2102 is always
up-to-date.
--- end ---
You are supposed to use a web interface for initial set up.
Jason
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Has anyone had any luck in changing the voices for Festival and Asterisk?
I have Festival installed and working, but can not get the voice different
from the default.
Thanks,
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Archer
Sent
I may be a little late on this, but what permissions are on
/usr/local/sbin/mailfax?
I have a similar set up to execute a mysql query to grab
the email address based on DNIS (PRI T1 with multiple numbers on one circuit)
and then email the fax to the destination. I set the perm to 755 on the
I'm looking for a way to capture the Agent ID after login, to keep
track which agent
is associated in a certain call.
--check out updatecdr=yes in agents.conf
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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Are you getting any messages from the CLI on * pertaining
to a sip user not registering?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fabrizzio
ValenciaSent: Sunday, July 10, 2005 7:45 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
and three
Member three
records member three
I guess my
question is what happened to the 'r' recording option in
meetme?
Thanks,
Jason
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}
That should give a pause before sending your PBX the 9, then add the 9, then
send the NXX to the PSTN after your PBX has seized the line.
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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http
=45e12a1c,response=c862acc59c
3914311b52e1bad7f8f4a5,uri=sip:[EMAIL PROTECTED]
Does anybody know? :-(
I think I need a sip-proxy setting but I cannot find anything written on
how to do this.
Jason Frisch
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that
* used www-auth in this case. If I can tell * that
the host is actually a proxy it should use proxy-auth
instead.
Jason
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Hi again,
I don't know if I am asking the wrong questions or just nobody knows,
but I will try
again anyway because I am quickly running out of hair to pull out...
Is there any setting in asterisk that will force proxy-authentication on
every call?
Please help :-(
Jason Frisch
Come on now children. Is this not a place to share knowledge?
Jimmy Smith wrote:
coudnt agree more.. thats exact thing i was saying the other day..
please hold my di..k while i take a leak i don't want to wet my hands.
RTFM, google and test. || Pay
On 7/6/05, Brian West [EMAIL PROTECTED]
=as3bfef70d;lr=on for
address/port to send to
set_destination: set destination to 61.193.205.37, port 5060
Reliably Transmitting:
BYE sip:[EMAIL PROTECTED]:15060 SIP/2.0
Via: SIP/2.0/UDP 203.83.240.5:5060;branch=z9hG4bK30b77416
Route: sip:[EMAIL PROTECTED]:15060
From: Jason sip:[EMAIL PROTECTED];tag
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