Thank you for the response, I will try downgrading.
-Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron
Wellsted
Sent: Tuesday, July 11, 2006 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Issues
Thanks, got it.
-Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick
Perez
Sent: Tuesday, July 11, 2006 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk stops abruptly
no.
just "l
Thank you for the quick response. I assume this change will require an
Asterisk reload?
Thanks!
-Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
aka Bret McDanel
Sent: Tuesday, July 11, 2006 8:43 AM
To: Asterisk Users Mailing List - Non
Asterisk 1.2.9.1 is the version I'm
on.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
BrummerSent: Tuesday, July 11, 2006 8:30 AMTo:
asterisk-users@lists.digium.comSubject: [asterisk-users] Issues with
making Transfers
Hello,
I am having a
problem with transfe
2
I'm not sure if the
SIP 500 error is relative to my issue. Any ideas on what could be causing
SIP transfers to hang or drop?
Thank
you,
Dan
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r/log/asterisk/messages log
file but it doesn't contain any information that I can use to help troubleshoot
the application crashing. Is there a way to put more debugging in the log
file?
Thank you for your
help,
Dan
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Using * 1.2.8
with some AAH2.0 dialplan logic.
Thx
Dan
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is a README in the tgz file that covers the steps
required.
Dan
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key to my questions is that I suspect you do need at least a
minimal QoS implimentation. A quick check on the circuit utilization
when the issue occurs can confirm this, or eliminate it.
Dan
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A few days ago, I started getting these errors on my Asterisk (1.2.9.1) console:
-- Executing Queue("Zap/1-1", "sales|tT|||3600") in new stack
-- Channel 0/2, span 1 got hangup
-- Channel 0/1, span 1 got hangup request
Jun 27 10:53:27 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring re
>We did it by comment out a number of lines in the code and then re-compiled
>just that module.
Thx Cullin for the reply, has anyone made a flow chart or end user
instructions for comedian mail? Jus trying not to reinvent the wheel if it's
already been done.
T
es, not that
it isn't out there.. I just cant' seem to find it.. Any pointers?
Thanks
Dan
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---Original Message-----
From: Dan Elder [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 1:47 PM
To: 'asterisk-users@lists.digium.com'
Subject: Dropped calls continued
Hi All... Well, I'm still experiencing LOTS of dropped calls since
installing the new (non pri) T1 here... I
Hi All... Well, I'm still experiencing LOTS of dropped calls since
installing the new (non pri) T1 here... I keep noticing a few things in the
logs when this happens, namely the "Wink/Flash" statements and the "Didn't
get a frame" messages...
Anyone got any ideas on if this is a telco issue, a wir
>This is *not* bizarre behaviour. Check your sip.conf file and add
>this line if it doesn't exist:
>progressinband=no
Thanks, seemed bizzare to me because it had been running for 3 months
without this setting then it suddenly started ringing after pickups.. Will
add that setting, I ended up swa
Hi all, using * 1.2.9.1 and this week all of the sudden calls keep ringing
even after they've been picked up... Here's one users summary:
When I pick up the phone, I hear a dial tone and I am able to dial out.
But for some odd reason, the receiving line picks up while the outgoing line
is still ri
I wrestled and wrestled with echo issues for months before finally just
following the guide here:
http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers
And with help from the list I was able to COMPLETELY remove all echo for
under $100.. Requires a tiny bit of simple soldering, but i
I'm am experiencing the same issue after upgrading from 1.2.0, still trying
to track down any info on it.
>w are running an asterisk server in connection with an octopus telephone
system. I have expired
>some random drops of zap channels bridged to SIP Telefones ( snom 190 ). A
>sterisk Messages
Re: my previous post about callid.. Seems to be working now after a complete
restart... Still unsure why it was borking though...
Thx all for your patience.
Dan
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To
extension as the
CID.. Any idea what would cause this? It WAS working on the 1.2.0 install
that I upgraded from last week.. No config files were changed during the
upgrade. Any ideas or pointers would be greatly appreciated!!
Thanks
Dan
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cing
server. Absolutely no issues with either channel. It would not be
too hard to add support to any of the RTP-based channels, but IAX
will not work with the code as it stands.
> Attilla
Dan
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Hey all, is it safe to run the asterisk-update.sh script that comes with AAH
to upgrade only the asterisk binaries? Doug has chimed in a few times saying
'upgrade' when I post problems, but Aah makes this really painful. I'm using
AAH 2.0 & am fighting a number of 'bugs' that only seem to be manife
Thanks Bill, will check this out, but from the bug report the problem linked
to below seems to happen when people are checking their vm as it's being
recorded. In the situation I'm looking at, the file will 'skip' to the end
even after it's been sitting in the inbox for hours. I'm going to do more
, and they play fine... This is *
1.2.0 -- anyone have any clues what might be causing this or has anyone had
any sort of similar occurance? It's driving both me & my users nuts!!!
Thanks in advance, as always!!!
Dan
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Thanks, will try this... I actually don't really want to delay incoming
calls before the attendant, but it seems to take about 7-10 seconds from the
time I dial until the AA picks up, without a ring, it just sounds odd, like
the call didn't go through...so I wanted to experiment with trying to add
Hi all, been searching & not finding an answer to this, although I'm
guessing it's absurdly simple... I just hooked up a T1 to our * box (1.2.0),
which had been using POTS lines via a channel bank.. Now when I call the new
T1 circuit, there are no rings, the Autoattendant just picks up right away..
display for them to be usefull in our environment.
Thanks again!
Dan
>Are you doing something funny with the CID on it's way to the phone?
>I've got a somewhat similar problem with an Aastra IP phone (yes, I did say
IP): it would NOT ring if the caller id
>>started with a
or 729, etc).
So it is not possible today, with code that is available, to send
a number of frames greater than the number required for 20ms of audio
in IAX.
I hope that makes sense.
> Thanks,
> Vij
Dan
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to give the IAX packets a priority
boost, but will not reduce bandwidth requirements.
> Thanks in advance,
> Vij
Dan
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ht
tly.
Any pointers would be greatly appreciated, I've searched the Wiki & the CID
faq's to no avail.
Thanks in advance
Dan
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rily for MeetMe, I do not have reinvite
enabled.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Neubauer
Sent: Friday, May 19, 2006 7:27 AM
To: Asterisk Users Mailing List
Subject: [Asterisk-Users] RTP Packetization
Hi all,
I need to be able to
If I hook up a cheap callerid enabled phone to the same port that the
PT350 was on (on the AB II), the CID IS displayed on the cheap phone..so it
seems that something is happening with the connection of the PT350 to
asterisk. Anyone used one of these handsets successfully with CID?
Thanks as always
27;ve
found doesn't have any mention of a setting to display CID...
Thx as always
Dan
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sly used skills.
All learning is good. If you cannot make money with what you've learned
about
VoIP while exploring Asterisk, then at least you've added experience
that might
make the next techology you work with easier to understand, and maybe
I believe you can accomplish this with a well crafted
dialplan.
If you did not have the restriction against out of tree
modules, I would
recommend an app that strores the conference details in a
database
and would allow just this kind of
control.
Dan
From: [EMAIL PROTECTED
vopts' (language 'en')
-- Playing 'vm-review' (language 'en')
-- Playing 'vm-repeat' (language 'en')
-- Playing 'vm-delete' (language 'en')
-- Playing 'vm-reachoper' (language 'en')
-- Playing &
r. Any ideas where I
can look to try to track this down?
Thanks!
Dan Elder wrote:
> Hey all, am running into a problem with * 1.2.1 recently. When we leave a
> voicemail for someone, occasionally when they check the vm, * doesn't play
> back the message that was recorded. I can see
end of the
session asking if you'd like to delete the message, even though it never
played. Any ideas what might be causing this? I've deleted & recreated the
VM boxes, but the issue continues across the company, any pointers on where
to look to track this down would be greatly appreciat
On 5/9/06, Dan Littlejohn <[EMAIL PROTECTED]> wrote:
On 5/9/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
> Mimmus wrote:
>
> > Where is the problem? Asterisk or ARI?
>
> Since Asterisk has no control over permissions, where the files are
> located or the
ame for A B C and D then they will
be able to see all the conversations.
Dan
512.791.0137
www.littlejohnconsulting.com
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Turns out there is a fault due to the new way that asterisk installs It isnt feasible to fix it until the asterisk installation method has settled down a bit.
Does anyone know a time scale on this?
Dan Journo
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It has been approved. We started out trying to use
CVS on SourceForge, but
it appears that there have been major issues with CVS, so
we just switched to
SVN.
We need to
checkin a baseline, and start integrating patches.
Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
before I moved it
to the asterisk test machine.
Thanks!
-Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
LopezSent: Monday, May 01, 2006 11:53 AMTo: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users]
Problems with zaptel and
TECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
BrummerSent: Monday, May 01, 2006 10:41 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Problems
with zaptel and TE210P
Hello,
I'm just starting
out with asterisk and I'm playing around with the system. Currently I have
ough=yes
[default]
exten =>
123,1,Answer()
exten =>
123,2,Playback(hello-world)
exten =>
123,3,Hangup()
exten =>
_9NXX,1,Dial(Zap/g1)
Any ideas?
Thank you in advance, your help is greatly appreciated.
-Dan
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ground, and it taking up space.
If anyone is interested, please contact me off list.
Thanks,
Dan
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The following occurs during make asterisk-addons.
I'm ok with asterisk but debugging things like this isnt my strong point.
Can anyone give me a pointer?
Thanks
Dan Journo
[EMAIL PROTECTED] src]# cd asterisk-addons[EMAIL PROTECTED] asterisk-addons]# makemake -C format_mp3 allm
or should I not even
try?
Thank you in
advance, I greatly appreciate it.
-Dan
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Seems like other postings tend to think that saving recordings as
files and not as blobs in the database are a more reliable way to go.
Opinions on this? Looking at supporting it for ARI and judging
interest.
Dan
512.791.0137
www.littlejohnconsulting.com
Is there an easy fix?- Dan L.On 4/25/06, Philip Edelbrock <[EMAIL PROTECTED]> wrote:
I experienced this today. Doing a 'show channels' in Asterisk showed aZap line perpetually ringing the sip phone even though the sip phone wasreset a few times. Doing a 'soft hangup' on
(additional language support is high on my
wish list).
Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben
QSent: Thursday, April 13, 2006 1:24 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] web meetme
> Dan Austin wrote:
>> Chan_h323 did not work with CCM, and a query/bug report was
dismissed,
>> basically stating that Cisco was F'd up and the channel would not be
>> updated to work with it unless funded. (fair, but not helpful)
> chan_h323 most certainly
e here that might back the concept. I also love my S518, and
would love it more if it was available as an option on the FXO module
for the A200.
Then I wouldn't have to choose between internet connectivity or POTS
connectivity on SFF systems that I like to use for PBX/f
On 04/06/06 04:41 Dan Austin said the following:
> Chan_ooh323 just worked. The code is, to a infrequent programmer,
> easy to read, extend and fix bugs.
Dinesh wrote:
> ok, i'm not getting into a my H323 is better than yours argument, but
> we've been struggling to g
t think it does, so I wonder
where it came from).
Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jordan
NovakSent: Wednesday, April 05, 2006 4:26 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] WebMeetme
Problem Please help!!!
I am running
ead, extend and fix bugs.
So for me chan_ooh323 is a 'better' H.323 channel driver.
Dan
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one supplier/service.
Good luck getting back online.
Dan
On 05/04/06, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
>>> Well, I wake up this morning, and aussievoip isn't>> up. I ring godaddy,
>> who _were_ hosting it, and they say that the>> machine's
Hi,
SHOWCHANINFO outputs no data in the following line:-
exten => 1571,2,VoiceMailMain(${SIPCHANINFO(peername)[EMAIL PROTECTED])
So that command becomes:-
exten => 1571,2,VoiceMailMain(@incoming)
Can anyone help?
Thanks
Dan
-
Lookup the realtime users db and read the MailBox column for that buddy.
If the mailbox column is empty, play a message saying "Sorry, no one is available."
If the column has data in it, do the following:-
exten => _11,3,VoiceMail(MailBoxID)
Many thanks
Dan Jou
I'd be happy to help, but you'll need to give me a bit more
to go on.
What do you see when you try to connect to the web
page?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jordan
NovakSent: Monday, April 03, 2006 1:26 PMTo:
asterisk-users@lists.digium.co
on April.
>> Dan Austin <[EMAIL PROTECTED]> wrote:
>> Sorry for the late reply, I was away on vacation.
>> Version 1.2 was created by Areski and I extended it to include the
>> scheduling functions. I guess I should get an account
Does anyone know of an affordable sip component or plugin that will allow me to build a voip softphone and customise it with branding etc?
Thanks
Dan
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Check out bugid 5162 on Mantis. It allowed per peer/user packetization
settings. It is in need of much love, but as-is should not be too
far from applying to the 1.2.X series.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Tuesday
any pointers to where this list is? I dont see it on the sourceforge pages.
Hans Witvliet wrote:
> aah-handbook (version 1.6) doesn't spill a single character about bri
> and "tfot" doesn't spill much paper of the subject either ;-(
>
> Any suggestions/pointers
>
> Hans
>
You may want to try t
review it I will post
an update to 2.1. If you'd like I can notify you
before I release it and we can
work on getting your translations in.
Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben
QSent: Thursday, March 23, 2006 7:35 AMTo: Asterisk
Users Ma
> I've got both the A200d and TDM04b in the same box with fc3 and all
> works very well. I'd suspect a config issue in your box of some sort.
> I didn't have to do anything special.
This is my first FXO/FXS card, so a config issue is likely, but like
you said they're pretty straight forward.
In
with kewlstart, but nothing
changed,
which makes sense.
Thanks,
Dan
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I found a solution... I just has to enter an Answer
line and now it behaves as I wanted. Here is the
working code:
[inbound]
exten => 1234567,1,Set(GROUP()=limit)
exten => 1234567,2,GotoIf($[${GROUP_COUNT()}>2]?103)
exten => 1234567,3,Dial(Zap/5&Zap/6,25,tT)
exten => 1234567,4,Voicemail,u110
exten
I want to limit the number of simultaneous incoming
calls that my IAX DID can accept to, say, 2. The IAX
DID provider sets no limit.
The code below does work, but when the limit is in
effect, new callers hear a "call cannot be completed
as dialed.." message instead of a busy signal. Maybe
this is
Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on
ould transfer fine)... so, no idea why
this is mucked up, but the above hint seems to have resolved my issue.
jus fyi...
Dan
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Hey all, an odd update to my previous note about not being able to transfer if
I'm the 'caller'... I set the dial command (in amp) to "T" only... now, If I
call into the pbx from outside and reach an extension, I CAN hit # on the
calling phone (outside, from-pstn) and get the 'transfer' message
Hey all, posted this the other day, but re-read it & realized I didn't give
enough info to be useful.. so I thought I'd try again.. I'm using AAH 2.0 (*
1.2.0) and am unable to transfer a call when I initiate the outgoing call. In
AMPs general settings, I've tried changing the Dial command using
Hi all, something odd is goin on w/my AAh2.0 install.. in my 'dial commands'
section, I have Tt - but if I try to transfer a call I originated, the # key
(attended transfer) nothing happens. I can transfer the call if its coming in,
but not if I made the call... the dial commands seem to be set,
I'm using Sipura SPA2002s which are basically the same as PAP2s.
To get this to work, I modified max_retires in chan_iax2.c from 4 to 10
and I haven't gotten any disconnects in 2 days. Does anyone know of any
other issues this my cause?
Dan
-Original Message-
From: [EMAIL
running asterisk
1.2.5.
If you have any ideas I would really appreciate some assistance. Thanks in
advance.
Dan
Disclaimer: This message is intended only for the use of the noted recipient(s) and may contain information that is privileged, proprietary and confidential.
If you
>It's in zapata.conf
is that
transfer=yes
& if I set this to no, does this keep the # transfer functionality that is
setup w/AAH?
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Is there an easy way to disable flash transfers? I'd prefer the users hit # to
transfer, since some users are hanging up a call, then dialing another one
without giving the handset enough time to actually hangup the call, so it
appears that they are transfering the 'ended' call to the new number
Is there some way I can follow this list from a newsgroup??
Is this the same as the gmane group gmane.comp.telephony.pbx.asterisk.user
??
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ve to answer somehow; I've read all through
TFOT and see nothing relevant to this issue. It's silly to spend $15000 on
a G723 license just so I can play back menu messages from Asterisk (since the
actual call decoding is done by the external boxes, which have already paid the
licensi
w AJAX enabled and that the voicemail
and call monitor pages self update.
You can take a look at it here
www.littlejohnconsulting.com/ari
and it has been checked into FreePBX svn.
Dan
www.littlejohnconsulting.com
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I have a hardware FXO/FXS which handle my voip calls, and they support
G723 internally. Asterisk hands off these calls just fine, and everything
works, as long as I don't want PBX menues available... The
problem is, once I want it to return messages, it will only return them as
GSM... whic
We have an external FXO/FXS, and use Asterisk as a call router. We
want to use G723 for the actual phone calls, because we have limited bandwidth
on our return direction. This has been working fine so far.
However, now we want to set up Asterisk to handle PBX menues and accept
extention
7;m curious how people have found it, and if there is an Asterisk
related news site that I missed.
Dan
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Dan Elder wrote:
> Hey all, I've been seeing this repeatedly and am wondering if anyone has a
> clue what's causing it..
>Could someone have managed to transfer one outside call to another?
Definately a possiblity, thanks for the tip! I'll also check out the disco
Hey all, I've been seeing this repeatedly and am wondering if anyone has a clue
what's causing it.. at least once a day I see two zap fxo channels being
bridged, and hanging..now, these two channels should never bridge, but they
keep doing it.. any leads on where to look for what's causing it? h
As a follow on from my last email, it appears that Asterisk restarts the player application if the process terminates.
Does anyone know a way to stop that?
Thanks
Dan
On 27/02/06, Dan Journo <[EMAIL PROTECTED]> wrote:
Hi guys,
Matt gave the advice below for a way to cause MoH to rewi
this?
Thanks
Dan Journo
www.TextOver.com
On 03/02/06, Matt Riddell (IT) <[EMAIL PROTECTED]> wrote:
Dan Journo wrote:> Ok, i feel like im getting somewhere but i need a little help.>> Asterisk displays this when its loading:-
> [res_musiconhold.so] => (Music On Hold Resource)>
sentence..."i am sick and
tired of my DIAX"?
There is something wrong with the application?
If you have any bugs or feature requests to report, please do not
hesitate to send me a mail directly.
Thank you in advance and best regards,
Dan
_
hNw2QfACbBybe
> 7COlKpOrWR92IQWJt1h6kDs=
> =2n+0
> -END PGP SIGNATURE-
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>
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> To UNSUBSCRIBE or update options visit:
>http://lists
sk-users
>
Hi Zach:
Sorry for the slow response, but you can get ARI here:
http://www.littlejohnconsulting.com/ari
There are instructions for setting it up there as well.
Dan
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Ast
Hello,
I didn’t exactly find what the
problem was but I built a new Asterisk server, copied the conf files over from
the original server and now the phones work fine.
Thanks, Dan
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Peters
Sent: Friday
goes on hold. C cannot flash back over to pickup As call. B does get
put on hold but the call to A never comes off hold. These tests are internal
on the hard and soft phones.
Thank you,
Dan Peters
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ou are really doing a GREAT job !
> Asterisk was really lacking this application part !
>
> Thanks again,
>
> And all the best !
>
>
> Jean-Marc
>
> On 2/17/06, Dan Littlejohn <[EMAIL PROTECTED]> wrote:
> >
> > ARI (Asterisk Recording Interface) has reac
>How is your echo can the issue?
>Did you disable the echo can and solve the DTMF issue?
I actually think my echo can had gotten into some odd state, a restart of the
tellabs board fixed the issue.
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support it.
Loaded into AMP CVS and also here:
www.littlejohnconsulting.com?q=ari
If you have a chance, take a look. Comments and suggestions are welcome.
Dan
512.791.0137
www.littlejohnconsulting.com
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s for
improvements.
Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben
QSent: Wednesday, February 15, 2006 6:40 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] [Announce] Web-MeetMe v2.0.0
It works!I hadn't pu
piled, but not
linked to the MySQL
libraries.
Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben
QSent: Wednesday, February 15, 2006 4:27 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] [Announce] Web-MeetMe v
Hi all, I'm getting some noise gate like effects on our sip lines & I think I
need to disable silence supression, I'm searching docs & not finding where this
can be set, does * have a setting to turn this off? basically what's happening
is when we stop talking, the other end hears total silence,
The error looks like a problem with the MySQL libraries on
your system. I have not
tested it against 1.2.4, but do have it running on SVN 7668
and have had it running
on 1.2.0
I can try 1.2.4 next week if you are not able to resolve it
by them.
Dan
From: [EMAIL PROTECTED
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