I am trying to change a 1.6 realtime statement into a 1.2 realtime
statement and I know much has changed. I wish I could just upgrade, but
alas not right now.
exten =x,n,Set(NULL1=${REALTIME(schedules,id,${SCHEDULE})})
comes back with
pbx.c:1371 ast_func_read: Function REALTIME not
I know that this is a feature but I would like to have the hold music
recorded while a person is on hold. So I know the agent put them on
hold and not just muted.
I have
monitor-join=yes
monitor-format=wav
in my queues.conf
any ideas?
Per
I am getting a bunch of Primary D-Channel on span 1 up but there was not
a down message before that.
Is this normal?
Confidentiality Statement Notice: This email is covered by the
Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and
intended only for the use of the individual or
It seems that my realtime is not assigning channel variables correctly.
INFO
Asterisk 1.6.0.26
Exten.conf
exten = _X.,1,NoOp()
exten = _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)})
exten = _X.,3,Set(NULL=${REALTIME(agents,device,${DEVICE})})
exten = _X.,4,NoOp(DEVICE is ${DEVICE})
exten =
I am getting this error
[Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum
retries exceeded on transmission [EMAIL PROTECTED] for seqno
102 (Critical Response)
[Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging
up call [EMAIL PROTECTED] - no reply to our
I had to switch quickly to 1.4.9 from1.2.4 and now I can only get 2
digits into the dialplan.
error
-- Invalid extension '81' in context 'impact' on
SIP/207.174.111.34-b77167f8
I pressed 8107
and ideas
my dial plan is (part of it)
[impact]
exten=s,1,Answer()
) --
Are you sure you didn't have those extensions in another context that
you forgot to include?
According to the dialplan it is catching the invalid extension and
should be passing it to the i (invalid) handler to loop back into your
attendant.
On 8/1/07, *Jason Walker* [EMAIL
extension and
should be passing it to the i (invalid) handler to loop back into your
attendant.
On 8/1/07, *Jason Walker* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I had to switch quickly to 1.4.9 from1.2.4 and now I can only get 2
digits into the dialplan.
error
I am looking to allow some users to login to a website and change where
their ext is forwarded to. any ideas? It can be very simple or I can
install a full package and then allow certain users certain access.
Thanks in advance
Jason
___
I do not have any answer int he dialplan. what I mean is that when I
call any other SIP phone is does the answer in the CLI. Even if I put
and answer() in the dialplan still no ringing
Jason
Luki wrote:
shouldn't there be an answer in there somewhere?... like...
No... you can (and
I have 2 linksys SIP phones SPA-942
I have a dialplan of
exten = 144,1,Wait(1)
exten = 144,2,Dial(Sip/phil,20)
exten = 144,3,Voicemail([EMAIL PROTECTED],u)
The CLI looks like this when I dial 144
-- Executing Wait(IAX2/JASONSERVER-9, 1) in new stack
-- Executing Dial(IAX2/JASONSERVER-9,
Good Idea, but when the user has to do nothing is better for my users!
Thanks
JAson
Mojo with Horan Company, LLC wrote:
Another option is to have the user hit the forward button on their
phone and manually type in their cellphone number when they're going
to be out of the office.
Jason
exten = 111,1,Wait(1)
exten = 111,2,Playback(Randy)
exten = 111,3,Dial(Sip/Randy,20)
exten = 111,4,Goto(111-${DIALSTATUS},1)
exten = 111-BUSY,1,Voicemail([EMAIL PROTECTED],u)
exten = 111-NOANSWER,1,Dial(IAX2/${TELIAX_OUT}/212551212)
works GREAT
Thanks a lot
Jason
Doug Lytle wrote:
Mike
I have users in my dialplan that go from SIP to Cell
When they are at their desk and they hit reject call, it goes to the
next thing in the dialplan, thus transferring to their cell. Not what
they want. Is it possible to change the reject button to make it go to
voice mail or a new ext?
I have a SIP user and a remote IAX device
I want both to ring 3 times then if neiter pick up it to go to the next
thing in the dialplan. Can you do this from the dialplan or do I need
to set a hunt group up?
Thanks
Jason
___
--Bandwidth and
From what I know this log show everything working Perfect.
Then it goes to the Welcome screen then after a long time of processing,
it errors out with a 0x1 error
Any Ideas?
1005195711|so |4|00|-- Initial log entry --
1005195711|so |4|00|+++ Note that bootrom log
Fixed that issue but it does not change the error
0126204105|cfg |3|00|Image sip.ld has not changed
0126204105|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr
1 of 1)
0126204105|cfg |3|00|Downloaded application image is identical to
current version
0126204105|cfg |3|00|Phone
PROTECTED] On Behalf Of Jason
Walker
Sent: Friday, January 26, 2007 12:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom Provistioning Issue
From what I know this log show everything working Perfect.
Then it goes to the Welcome screen then after a long time
I have
1.2.12.1
Voicepulse using IAX
I get about 30-40% issues with not having the DTMF tones work.
I have 3 questions
#1. Voicepulse says they are sending them, Is there some setting I can
adjust to make sure my end is working?
#2. I have set the Dialplan to play a sound Operator then go to
I put my voicemail groups into different contexts so that I can use Dial
by name and escape.
I had set ext 500 as
exten = 500,1,VoiceMailMain(${CALLERID(number)[EMAIL PROTECTED]|s)
but now that the contexts are different. this does not work
#1 how do I have everyone use an ext to get the
Ok sorry for not being specific. I am having a problem when people
outside call in to my number which terminates at VoicePluse then The
send IAX to me and I do not get any tones. People press buttons but it
just goes to the next dialplan fall through. It happens 60-70% of the time.
Ken,
Also stay away from Swissvoice phones
I have found several ways to do the second thing.
http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers
It works great.
Jason
Tom Vile wrote:
I tend to stay away from the Grandstream phones for
business use because they simply break to
I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct
DTMF tones 25% of the time. I have to call several times to enter an
extension. I have a router and a packet shaper and some other stuff.
Anyone have any other ideas why this might happen. I do not have any
Zap channels
I used to have fonality and I could press * when I got to someones voice
mail to go back to the menu. I assume I add that to the dialplan but
how? Thanks
BTW I went back to 1.2.12 and transfer works and DTMF works and it seems
to be much better for now.
Thanks for you help
Jason
I am now running 1.4 beta3
I have an ongoing issue that it does not recognize my DTMF key press. I
will call and press as many numbers and the background message still plays.
I am also having an issue with transfers
NOTICE[30930]: chan_sip.c:13289 handle_request_invite: Unable to
create/find SIP
I am having a bunch of issues with 1.4 and want to go back to 1.2 any
ideas on the best way I saw someone say apt-get remove will this work
for asterisk or do I need to do it for each libpri, addons, zaptel and
asterisk?
Thanks
Jason
___
be great. I am a little new to asterisk and so if I
posted this incorrectly please let me know
Jason Walker
___
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To UNSUBSCRIBE or update options visit:
http
Some phones do not send DTMF automatically. What soft phone
are you using?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
LopezSent: Wednesday, January 25, 2006 9:23 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Dial
Julian -
What hardware are you using? Proc, RAM, SCSI or IDE, etc.
The reason I ask is that I have multiple hardware platforms, all on FC1 or
FC4, and none of them hit 100% for each IRQ. I am usually in the high 98%
with the occasional 100% on P3 servers (give or take 1 Gig RAM, 1 Gig CPU).
Two
The statement of zaptel being required is strange...I use IX trunking
exclusively for my servers. Two of them have no zaptel/Digium hardware and
the trunk calls are fine.
Based on your post, seems that you have an issue with codecs more than
creating an IAX trunk.
What version of Asterisk are
I have looked
through other postings to the user group for HDLC errors, went through what
worked for other people, and still can not seem to get past this
issue.
For 3 days, I have
been getting HDLC abort(6) errors in *. Prior to Tuesday, the circuits were
clean...I had maybe 10 HDLC
My 2 cents:
If you are running kudzu on RH or FC, new and remove hardware should be
detected...in most cases. I assume other distros have something similar...?
If 2 of 8 T1s are not coming up - sounds like you may have a wiring issue.
Can you swap cables from a bad circuit to a good circuit?
until I can
fix this.
Swapping card does not seem to follow issues.
Maybe I'll give support another :)
Bart
- Original Message -
From: Jason Walker [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, October 29
One thing to consider is if there were alarms on the T1 to SBC, they may
have something in place to take the circuit down. Even if you get your
configs right, the T1 just might not come up clean. MCI does this to us
sometimes.
Please post your /etc/zaptel.conf and your /etc/asterisk/zapata.conf
I have not read through the rest of your posts, but try some of the other
variations of switchtype:
; Switchtype: Only used for PRI.
;
; national: National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess: ATT 4ESS
; 5ess: Lucent 5ESS
; euroisdn: EuroISDN
;
What codec are you using on the client and the server? From
my understanding, you have to have a license for both ends of the G.729 call.
Are you passing this through one server to another and the call is being
rejected at the server level?
From: [EMAIL PROTECTED]
[mailto:[EMAIL
and
OZTell, my provider, use G729 as their main codec.
My box rejects connections from my provider due to
incompatible codecs and vice versa.
I'm waiting for them to get back to me on
this.
Clint.
- Original Message -
From:
Jason
Walker
To: 'Asterisk Users Mailing List
get back to me on
this.
Clint.
- Original Message -
From:
Jason
Walker
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Sunday, October 23, 2005 10:52
AM
Subject: [other] RE: [Asterisk-Users]
Unable to negotiate codec???
What co
Tom - do you end up with that phone shutting down with
an error on Windows XP? I downloaded the latest. After about 3 minutes on a
call, the other end can no longer hear me and then the phone just
dies.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
VileSent:
with one specific version of
asterisk ?
Whatever the problem is, it should not be there. Please help us find the
bug.
Joachim.
Jason Walker wrote:
Tom - do you end up with that phone shutting down with an error on
Windows XP? I downloaded the latest. After about 3 minutes on a call,
the other
Nope, I do not have that issue.
On 10/23/05, Jason
Walker [EMAIL PROTECTED] wrote:
Tom - do
you end up with that phone shutting down with an error on Windows XP? I
downloaded the latest. After about 3 minutes on a call, the other end can no
longer hear me and then the phone just dies
: [Asterisk-Users] iax softphone
I'm running it on sp2 myself, never had a crash with it so far.
Jason Walker wrote:
Are you running on XP SP2just curious? How about the version of *?
--
--
*From:* [EMAIL PROTECTED
I have not seen any
posts for awhile. Just testing.
thanks
___
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
When I run 'ps aux'
I get this:
root 964 0.0 0.4 47836
8280 ? S
00:02 0:00 asterisk -vvvg
-croot 965 0.0 0.4 47836
8280 ? S
00:02 0:00 asterisk -vvvg
-croot 967 0.0 0.4 47836
8280 ? S
00:02 0:00 asterisk -vvvg
-croot 975 0.0 0.4 47836
8280 ? S
00:02 0:00 asterisk -vvvg
-croot
What if you force a hangup between the two steps?
I have multiple destinations specified when my internal number is called at
work using similar syntax. All of the SIP and SCCP extensions dial based on
my setup - which again, is very similar to yours.
I do not use CVS HEAD on the production
As an FYI - here is the output of my TDM400P:
Module 0: Installed -- AUTO FXS/DPO
Module 1: Not installed
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Installed -- AUTO FXO (FCC mode)
I do not have newt installed on this machine, so zttool bombs. Just sending
this out as an example.
Correct - but is the context defined in voicemail.conf? As mickeymouse? Or
whatever...?
;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of FaberK
Sent: Saturday, October 15, 2005 6:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Have you tried the incominglimit parameter (or did she)?
I have found this to work pretty well when limiting the number of calls.
After monitoring the full log, I saw that incoming calls where
incrementing or decrementing the active call parameter for SIP agents. By
limiting the number of calls
I have 4 * servers interconnected with IAX trunks. Three are on a local LAN,
one is accessible over a VPN tunnel out of the office. The IAX peer status
(iax2 show peers from the CLI) will sometimes show upwards of 300ms.
Considering the lag and distance, I am not entirely surprised.
Anyway - my
2005, Jason Walker wrote:
I have 4 * servers interconnected with IAX trunks. Three are on a
local LAN, one is accessible over a VPN tunnel out of the office. The
IAX peer status
(iax2 show peers from the CLI) will sometimes show upwards of 300ms.
Considering the lag and distance, I am
You may have already tried this, but in the past whenever slips come into
the picture on my T1s, crimping a new end for the CAT5 cable seems to help.
We run T1s to a 110 block. Every once in awhile, the 110 needs to be
repunched.
I have found that slips can clear up when we rerun the
I would appreciate seeing the scripts as well. Nice job!
Desktophero at gmail.com
Thank you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Troy Swaine
Sent: Friday, October 07, 2005 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Has anyone used the DS3 card from Sangoma with Asterisk?
I have read many posts from users that the Sangoma cards have better echo
canceling and so forth. I guess I am just wondering if there are more
benefits to using this brand.
I currently am responsible for multiple Asterisk servers all
It looks like your * server is not able to see the destination
(presumably sip.uni.it).No route to
destination
-Original Message-From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]
On Behalf Of Fabio MontemaggioreSent: Friday, September 30, 2005 2:34
AMTo: asteriskSubject:
One key that I have found is the more RAM the better. I am not discounting
the CPU by any means and with the number of registrations you are talking
about, I have not set up a system for that many concurrent users.
I do have a 2x1.266 PIII w/ 2 Gigs of RAM that handles 75-85 concurrent SIP
I have run into a similar situation. One of our older faxes at the office
seems to not work with spandsp module. The newer faxes work just fine.
When I watch the logs, there appears to be communication from * requesting
the fax to slow down. When the fax machine does not respond, * seems to
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]
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 using a card for
timing (TE405P is what we pretty much use), I feel pretty comfortable with
FC1 and 1.0.9.
Are you using 1.0.9? Have you tried 1.2 beta?
-Original
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 using a card
for timing (TE405P is what we
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
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
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
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Denis
Girard
Sent: Wednesday, August 31, 2005 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd
Jason Walker
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
From voip-info.org:
Queue(queuename|options|optionalurl|announceoverride|timeout)
'optionalurl' allows you to send a URL to devices that support it.
Does anyone have details on the devices that support the
optionalurl method of the Queue application? I am wondering if there is
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
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,
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
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
The SEP file should be
SEPMACADDR.cnf.xml
You can also use XMLDefault.cnf.xml
These have worked for me w/ 7960.
What phone are you using?
Here is some more information for reference:
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx
-Original Message-
From:
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
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. ${EXTEN} would always be
720.
I don't follow your logic.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday,
(sipexten,${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
I currently have two channel groups in my zapata.conf file. I would like
one group to be immediate=yes and the other immediate=no
Does not seem to matter which way I go, the first entry in overrides my
explicit setting for the second group. I am running * 1.0.9 on FC1
[trunkgroups]
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
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:
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
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:
I have a conference set up through MeetMe and I
can record each call coming in with the Monitor command. What I would like to
move away from is having to then generate multiple files for the final output of
these calls.
On voip-info.org, there is an 'r' option to record
the conference.
Ummm, yes.
I am not quite following what your question is asking. However...I have a
PRI line pushing inbound calls to SIP users.
Can you expand on your question?
Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Equipe du
Royaume
Sent:
I have setup a TE110P with a different carrier. The pri
switch setting I used was national. I think this will work with NI1 or NI2.
Interestingly enough, I have to use this against a ATT 4ESS carrier
switch.
The number of digits outpulsed is usually a ten digit
number.
The version of the
What OS/distro are you running?
I experienced the same on Gentoo with the 2.6.xxx kernel. Switched to FC1
(2.4.xxx kernel) with the 1.0.7 CVS and have not had any issues.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Monday, June 13,
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