this, but I want the former user of that phone to be unlogged
> immediately.
This needs to come from the phone - your phone should have a setting
something like 'unregister on reboot' . Turn this on.
Peter
--
Peter Bowyer
Email: [EMAI
On Wed, 4 May 2005, Alex Mack wrote:
> So I'm already doing ECT by using the bristuff'ed version of *?
I have no idea. We use PRI only, not BRI. Hopefully it is in the
documentation for bristuff.
Peter
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On Wed, 4 May 2005, Andreas Sikkema wrote:
> As far as I know, Asterisk doesn't support QSIG. Do you
> _have to_ use QSIG?
I think there is q.sig support in libpri. It may be avialable to bristuff
as well.
Peter
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ing the
> link to the PBX with too many (in my case 4) forwarded calls.
>
> Does anyone have experiences with this? Are there references for Q-SIG
> out there?
I think at least some of the standards are published at
http://www.ecma-international.org/activities/Communications/QSIG_page.htm
out the new Grandstream GXP-2000. I've been testing these,
they're much better than the BT-100s.
Peter
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low-to-high
and the cpe end to hunt high-to-low.
Finally, even on isdn you have end devices (phones) which may themselves
be prone to the human equivalent of glare - picking up the handset before
the ring is heared. Some phones allow the user to request an outside line
by pressing a button to
/R2 support. See
http://www.soft-switch.org/unicall/installing-mfcr2.html.
Peter
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Doesn't Voicemail already do this? I just tried with mine and pressing *
while hearing the greeting dropped me into VoicemailMain for that box.
Peter.
On Mon, 2005-05-02 at 19:48 -0500, Lenwood S Sawyer III wrote:
> All,
>
> Can a user dial in from the outside on a DID and then
on't know if it fits well.
If someone is willing to communicate, it would be every appriciated.
Regards.
-Peter
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he keys on my
> dev. box would cause a conflict (also was pretty clear that I wanted
> to be in compliance with their license agreement) and the lady said
> there was no problem and leaving the old keys on the dev. box would
> not cause a conflict.
>
> On 5/2/05, Peter <[EMAIL
Hi all.
Dopes someone know how I can move a key license of the g729
codec from one to another machine?
Find nothing usefull @ the wiki.
Thnx 4 help in advance.
Regards.
-Peter
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to the calling party. Both parameters
can be used alone.\n"
Peter
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rent members of the group using the Dial(chan&chan&chan&...) syntax.
Peter
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ne possible cause is in the Type
Of Number (TON) handling in Asterisk. What is the prilocaldialplan set to
for the link to the Eicon PRI? Are you using any of the "nationalprefix"
or similar options?
Peter
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card and not a non-isdn T1 card.
If the KX-TD1232 uses BRI CO lines they can be used instead. It may be
hard to obtain BRI cards in the USA.
Peter
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n debug log. For a PRI link the command would be "pri intense
debug span X". I assume there is a similar command for a BRI. It should be
possible to find the problem from that log.
Peter
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motherboard
will distribute the pci busses across the cpus.
Read http://www.samag.com/documents/s=9408/sam0411b/0411b.htm for good and
bad examples and a list of things to watch out for when purchasing an
Opteron system.
Peter
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The incommingmsn line is not the issue for me as I just use * its the msn line.
I need to have an entry in there for each of the msn numbers that I want to
dial out on, so with 10 msn numbers at 10 digits each and 9 commas between them
its a bit longer than 80 chars.
Peter
> -Origi
agi scripts that fill the
> gap. One of them finds a hardcoded IP in a hash of extensions, the other
> sends the callerid information in YAC format.
>
> Email me if you want a copy.
>
> Adam.
>
> p.s. CC adam@ to make sure I see it if you reply.
I'd appreciate a cop
GTON was not populated in -stable. Tha patch was only added to
-head.
It is not that hard to add, I can send you our old patch if you want it.
Peter
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se functions) but with quite a
few additional patches of our own. there is currenctly a showstopper bug
where the dtmf-detecting dsp is disabled on outbound call legs. Bad if you
need #-transfers.
Peter
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be these kinds of hacks.
Peter
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control subclass '17'
Apr 26 11:21:37 WARNING[20842] file.c: Unexpected control subclass '17'
Apr 26 11:22:14 WARNING[20842] file.c: Unexpected control subclass '17'
What does it mean ? Should I do something about it ?
TIA
Peter De Schrijver
_
hy but it just works.
In the first line you passed th "r" in the argument reserved for the
timeout value. Th options field in Dial is the third argument, not the
second. So, you had a timeout of "r" seconds (invalid) and no ringback
option.
Peter
__
I modified the source code as I have 10 msn numbers here at home, I will try to
make a diff of the changes.
Peter
> -Original Message-
> From: Stefan Helbing [mailto:[EMAIL PROTECTED]
> Sent: 22 April 2005 16:40
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-U
Joseph Gutowski <[EMAIL PROTECTED]> wrote:
[...]
> I wasn't suggesting Asterisk should magically be able to pick up the
> call before it rings at all, just that if my old roommate could
> manage to dive across the room and pick up half way through the
> first ring 99% of the time, surely a computer
On 23/04/05, Michael DiMartino <[EMAIL PROTECTED]> wrote:
> Peter thanks for the response.
> I put the user name and password in but i still get the same error.
>
> ;Extentions at telx-nyc
> exten => _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
>
>
dientials you need to authenticate
with the other server.
The username/secret in iax2.conf is for inbound, not for outbound calls.
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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y?
Like someone else said - have you tried to define a group and call
"Zap/g1/.."? The message above means that asterisk does not consider
the channel Zap/1 to be available for one reason or another.
Peter
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A
; since that tells the
switch to interpret the dialstring the same way it would be done on a pots
phone.
Peter
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On 22/04/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
>
> On Fri, 22 Apr 2005, Peter Bowyer wrote:
>
> > On 22/04/05, Ian Hailey <[EMAIL PROTECTED]> wrote:
> > > Hello everyone,
> > >
> > > I am trying to receive DTMF commands on
ere which might be adding to your
problems - mail vs main - but the bigger problem is that you are
sending calls to a context called 'from-ask-main', but that context
doesn't exist in your extensions.conf. You have one called 'from-sip'
which is where you probably could send t
Gavin Hamill <[EMAIL PROTECTED]> wrote:
> On Friday 22 April 2005 12:07, Peter Corlett wrote:
[...]
> In the UK it's entirely possible - the CallerID info comes through
> as encoded data before the first ring has taken place :)
> Polarity change, a burst of V23 data, then t
Joseph Gutowski <[EMAIL PROTECTED]> wrote:
[...]
> Either way, the best I've ever managed on the X100P's was 1 ring
> before Asterisk picks up and starts doing its thing.
Well, when you think about it, it's hardly going to pick up after zero
rings, is it? :)
--
Beer is proof that God loves us an
s anyone
> have any idea if these tones are on purpose muted by the service
> providers or any other reason why it does not work?
I'm not aware of the detailed reason, but DTMF into Asterisk from
Sipgate won't work. This path is well-trodden...
http://www.voipuser.org/forum_topic_844.ht
the size of any jitterbuffer in the Adit. That may
make the echo latency lowe enough.
Peter
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ler
So the call is immediately hung up...
What am I doing wrong ?
This can't be such an exotic setup ?!
TIA
Peter
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To UN
r server?
Many similar systems (webmail, bulletin boards etc) are configurable
to use a local MUA (/sbin/sendmail etc) or talk SMTP directly to an
MTA, either locally or remote. Asterisk voicemail unfortunately is not
one of those systems (AFAICT) - you're stuck with having to use a
local MUA.
Have you tried using a CSU/DSU between each of the T1's on that system?
The problem could be a voltage issue on any of the 7 T1's running into
that box.
If you have the 7 CSU/DSU's it might be worth while to try running all
7. If not, try 1 at a time and see if the CSU/DSU notices any problems
upgrades free? or pay-as-you-upgrade?
* Does the device use a standard SIP (i.e. fully rfc compliant)?
Thank you very much!
Peter Hoppe
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Stephen,
It would be very kind if you provided the make and model of the fxs->fxo
converter. Also, what you mean by 'failed' (which symptoms - no ring
tone? No dial tone? etc.). Maybe also some more specific information
about your setup? This would help.
Thanks very much
Peter
y in fact usable with *?
You need the 'third-party' version. http://www.virbiage.com/firefly/download/
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Aster
Cell Phone TX" <214xxx>, on Tuesday, April 19,
> 2005 at 01:51:25 AM so you might
>
voicemail.conf
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Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
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Aster
er. You will in turn transmit the alarm state blindly to the other
side that sees it as a remote alarm.
So, most probably you are not hearing the remote end correctly. Recovering
I assume means that it hears some frames correctly, but not enough to kick
out of red alarm state.
Peter
___
Chris Hills <[EMAIL PROTECTED]> wrote:
[...]
> There are some providers who can terminate some, but not all, 1800
> numbers for free. (If they could terminate all 1800 numbers for
> free, then we'd use them!)
> I don't understand - I thought all 1800 numbers were free?
They're not like UK 0800s -
is is a problem with
> the drivers...
In HEAD this is accomplished through the variable PRI_CAUSE and the Hangup
application. This is the most generic way. Read more in the wiki or
search the mail archives.
Additionally the behviour of the Busy and Congestion applications can be
changed by
P, IAX, or otherwise) and be connected to the parked caller.
Since almost all pbx:es are hybrid systems most of them allow what the OP
wanted. In fact, most of it can be implemented using Asterisk and
SuperValetParking, contexts etc.
Something like supervaletparking in the standard Asterisk w
"r" on internal extension (going to a pbx
over pri or to some sip phones).
Peter
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ould legally
be ok. I am experimenting with one unit at the moment, and am smacked by
the literally hundreds of options it has. But I heard good reports about
that one, so I expect it to work well in our setting.
Hi Peter, I'm not sure how you are getting PSTN lines into your * box, but i
the Mediatrix voip gateways
1124 and 1204 which seem to use non standard SIP and have
pay-as-you-upgrade)?
Thank you very much for your consideration!
Peter Hoppe
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is speaking.
Peter
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e support is _way_ better.
Of course, if you are not familiar with the problem space for which you
are purchasing a solution then resellers can add a lot of value.
Peter
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<[EMAIL PROTECTED]> wrote:
> Johannes,
> I would be curious to know if there is a solution for this. Another
> solution is that you buy a "call meter". Which is a small box that
> can be placed in front of phone phone and that can display costs.
> FXS--> call meter --> analog phone
> This call me
ta.conf means, that asterisk is the slave,
> and "signalling=pri_net" means master, or do you mean something different?
I think the poster above meant T1/E1 timing (clocking) source, not isdn
net/cpe. The timing is configured in the span line of zapte
. When someone is sending a yellow
alarm they are usually considering the link to be down. Are you sure
there are any calls left active by the remote side when the yellow alarm
clears?
Peter
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ol reports
> "Yellow Alarm"
A yellow alarm means the remote end is sensing some error condition. Try
looking for an error message at the remote end. It may be as easy as a
broken cable (where the Hipath does not hear the Asterisk box).
Peter
_
digits in front. Since you start with an esacpe
digit almost anything can happen. In your case the telecom operator may
have interpeted your number as +38 x.
Peter
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On Apr 5, 2005 7:45 AM, Matt Riddell <[EMAIL PROTECTED]> wrote:
> Peter Bowyer wrote:
> >>exten 456,1,Background(Please-set-time-mmddhhmm)
> >>exten _.,1,System (date ${EXTEN})
> >>
> >>If I dial 456 I get the message, so I type 04021305 (2nd April, 13:
zero. Any ideas?
>
> exten 456,1,Background(Please-set-time-mmddhhmm)
> exten _.,1,System (date ${EXTEN})
>
> If I dial 456 I get the message, so I type 04021305 (2nd April, 13:05).
>
> On the console Asterisk reports the command Dial 04021305 exits non-zero.
You need 'Rea
On Mon, 4 Apr 2005, Tobias Jönsson wrote:
> On Thu, 31 Mar 2005, Peter Svensson wrote:
> > It would not be very hard to add both features to libpri. Libpri already
> > has a function to decode and dump the time/date information. If I
> > remember correctly the time/date
I am hoping someone in the * community has come across this problem before.
Problem:
Person SIP Phone A (SIPA)
Person SIP Phone B (SIPB)
SIP Phone C (SIPC PSTN Line)
SIPA calls a billable phone number via SIPC
exten => _123456/_1XX,1,SetAccount(${ACCOUNTCODE_COMPANYZ})
exten => _123456/_1X
ence with the inner
workings of configuration.
If there is someone whom has experienced this problem before or
something simular, I would be interested in knowing how you managed to
resolve it.
- Peter
Info:
--
Asterisk: v1.0.7
SIPA & B: Polycom SoundPoint IP 300
SIPC:
t? I would appreciate any comments!
Thanks in advance
Christian Peter
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resentative.
>
> With AMP I managed to set up a group which rings with an incoming POTS call.
> With AMP also, I have also managed to create a Digital Receptionist BUT this
> requires caller input, which is not what I need :(
Woulden't a queue with &q
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Squeaking / chirping on ZAP Digium TDM400P
host name permanent
either:
hostname >> /etc/HOSTNAME
or
echo "" >> /etc/HOSTNAME
On Fri, 1 Apr 2005, Kellner, Peter wrote:
> It's an HP
hanks, -Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Friday, April 01, 2005 3:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Squeaking / chirping on ZAP Digium TDM400P
Ke
Here is my printout below. It looks to me like it is sharing with USB.
I don't seem to have a way in my bios to turn off USB though and nothing
is plugged into it. Could that be a problem? Also, are the other
things mentioned all part of Asterisk?
Thanks, -Peter
CPU
ug the incoming line to a
PSTN. I'm the only conversation on this hardware and it is a 2.2Ghz P4
with 512Meg RAM.
Any ideas on what or how to look for this problem?
Thanks,
-Peter
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htt
Ping runs as a low priority service so it is not realistic to measure
response time using ping.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Johnathan
Corgan
Sent: Thursday, March 31, 2005 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discus
rectly the time/date IE should be added to the SETUP messages. I have
been thinking about adding it, but have not had the time.
Peter
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hat you're looking for.
http://www.voip-info.org/wiki-Asterisk+cmd+DISA
Peter
--
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Email: [EMAIL PROTECTED]
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On Thu, 31 Mar 2005, Joe Presto wrote:
> Peter, thanks. This would be a less than optimal solution for me, as I
> wouldn't be able to pass the caller id of the orig caller (which I could do
> via IAX), nor would I be able to announce the caller ID after the call so I
> could pre
34234234)
will dial 012345678 and 0234234234, both of which are required to send a
dtmf to be considered answered.
Peter
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To
dn is the preferred signalling,
but RBS can be used if it is available. Stay away from analogue links
since they will introduce two additional hybrids (2- to 4-wire
conversions) and the signalling is still not very precise.
We have hooked up our Panasoni
ld is correct. I would also suggest that the carriers
would also not want to allow this as it would impact on their interconnect
commercials.
Peter
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ated.
Regards
Peter
Here are some of the settings.
Sip.conf for the extension:
[9001]
type=friend ; either "friend" (peer+user), "peer"
or "user"
context=extensions
secret=9001
fromuser=Cisco ; overrides the callerid, e.g. required
100. Nice, relativly cheap, fast
and with a small footprint.
Peter
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t? I would appreciate any comments!
Thanks in advance
Christian Peter
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> Anybody using a Sipura 3000 for FXO with Asterisk?
>
> Mine is working except for one small nit...
>
> When a call comes in from the PSTN, the Sipura answers it and then
> passes it on to Asterisk, which plays extension ring tone.
>
> I'd prefer for the POTS line to stay on-hook while the e
I reinstalled .7 again from scratch and this time it worked. I have no
explanation for why it did not work the first time. Sorry for the
inconvenience.
I am still having a problem getting my internal extensions to work. I'm
including again my list email hoping someone will respond.
:::
I've finally got @Home .7 up and working. I have a Digium TDM400P with
2 fxo and 2 fxs. All I've done so far after loading is to run
genzaptelconf -s -d ; verified that ztcfg -v shows all 4 ports and
logged into the AMP app and added 1 extension (200) to one of my fxs
ports.
When I pick up e
dback to the team working on [EMAIL PROTECTED] If I did something
wrong, I'd sure like to know what, and if not, when a fixed version is
available.
Thanks,
-Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dalon
Westergreen
Sent: Tuesday, Marc
). Again, what am I missing? I thought AMP would
modify my extensions.conf but doing a diff before and after shows it
does not. Where is amp setting the extensions to work and what could be
my problem.
Thanks,
-Peter
(also, when I type aah_help I g
Is your network equipment QoS aware? If so have you implemented a high
priority for VoIP (IAX, SIP, etc) and lower priority (SSH, HTTP, etc)
for other networking traffic?
Also it doesn't hurt to have a QoS policy on your Linux (if you are
using Linux) box where Asterisk is running too.
Plus your
We use the ANSWER, WAIT, BACKGROUND which works perfecting fine.
e.g.
exten => _123,1,Answer
exten => _123,2,Wait(1)
exten => _123,3,Background(welcome)
On Mon, 28 Mar 2005 15:50:33 -0800, Robert Goodyear <[EMAIL PROTECTED]> wrote:
>
> On Mar 28, 2005, at 3:22 PM, Noah Silverman wrote:
>
>
ideas why would be appreciated.
Thanks,
-Peter
I have a TDM400P with two FXS and two FXO's.
My extensions.conf
TRUNKMSD1=1 ; MSD digits to strip
(usually 1 or 0)
TRUNKMSD2=2 ; MSD digits to strip
(usually 1 or 0)
; l
e Hangup app after setting the PRI_CAUSE variable. This is the
general way of sending a specified disconnect code. See
http://www.voip-info.org/wiki-Asterisk+variable+PRI_CAUSE
* set the configuration option "priindication=oob". Thi
hotel room without DID) you need the patch
from
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0003405
* You could use the manager interface or a call file to place a call
that, when answered, places the new party in the conference
autmatically.
* Various scenarios based
message starts playing
> almost
> immediately, so if the called person takes 2 or 3 rings to pick up the phone,
> half the
> message has already been played.
You need answer supervision on your line. It is available on isdn lines
and s
ts that are nearly as good from any
manufacturer. If you must have the call center on softphones then get an
adapter and use telecom headsets.
Peter
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth
> Sent: Friday, Marc
Most US carriers have an email --> SMS gateway that can be used for
free. I don't know about UK carriers. That wouldn't need a PSTN
connection...
-Pete
Mark Charlton wrote:
Hi all
I have been googling and wiki-ing and have found a number of potential
solutions to my questions, but I don't want t
t did) mix in the
outgoing pcm data into the recorded stream. This generated the most awful
echo until we turned that mixer input down to zero. It may be worth
checking out.
Peter
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http
7;s a
> voicecall, and mail the fax if it's one.
And the answer is in the first hit on google for the recommended keywords.
Peter
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he asterisk box rather than the
> forwarding I think.
Since when was telnet able to open a udp port? Actually, a udp port is not
opened at all, though you get an icmp packet back if you send data to a
udp port where noone listens.
I think you have tcp and ud
nbridged option.
Peter
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thers. If the Asterisk echo canceller is not
enough you may consider an expensive inline echo canceller.
Peter
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Hello
We are running Asterisk CVS 22/12/04 and pwlib/oh323 pandora version to work
with our call agent.
Unfortunately **VERY** frequently, asterisk stops responding and goes to 99.9%
CPU. There is no debug output or other information that indicates there is a
problem...
Rather than continually
be a good idea to add a comment to the
bug report so the problem gets resolved.
Peter
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background knowledge about Asterisk. Alternativly you need to formulate a
more precise question which can be answered.
Peter
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To UN
tart with, without a
context specified in the DISA command, the number you dial will be run
from a context called 'disa', do you have one?
Otherwise, post the console output with plenty of verbosity and tell
us what you think should be happening.
Peter
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