For the time being try putting 212.41.253.181in hostname= line in ur sipconfig and it should work . Also check if you /etc/resolv.conf has correctdns list ( i guess it does bcoz OS canresolve) . Also check /etc/asterisk/dnsmgr.conf .
Here's example :[general]enable=yes ; enable creation of managed
At 08:48 AM 11/9/2006, you wrote:
Anyone having problems with voxee since last few days or is it just
me ? In peek hours i get LAGGED when i do a iax2 show peers or even
1000 ms latency . Most of time it is 20 ms or so but when i start
sending traffic to them latency increases to 1000 ms or
I'm running chan_capi on a number of systems in France, France Telecom
offer the possibility of having the caller's name, but say we must
configure for EuroISDN+. Google doesn't show much and the best I could
see was in Dutch.
Any Europeans solved this one?
Rgds
--
Dave Cotton [EMAIL
Hi,This is piece of cake for asterisk, but you need to do your script, or dialplan programing, asterisk has all the functions and applications to do it.But you need to get hands on it :)
On 11/10/06, Jeronimo Romero [EMAIL PROTECTED] wrote:
I'm running asterisk 1.2.8. I would like
Am Freitag, den 10.11.2006, 11:21 +0100 schrieb Dave Cotton:
I'm running chan_capi on a number of systems in France, France Telecom
offer the possibility of having the caller's name, but say we must
configure for EuroISDN+. Google doesn't show much and the best I could
see was in Dutch.
OhI'll see you :)On 09/11/06, Kristian Kielhofner [EMAIL PROTECTED] wrote:
Felipe Amaral wrote: Hi, There's anyone here who go to Estacao Voip in Brazil???
http://www.estacaovoip.com.br/ I was think to go Anyone here ?? -- Felipe Amaral
Vento Livre InternetFelipe,I will be there,
Hi!
We have an installation with WLAN SIP phones only. Sometimes we have
connection drops. What is the best way to debug if we have problems
with the WLAN or the SIP devices or the uplink to the IAX Provider.
TIA,
Mike
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Helo
My money is on the WLAN part of the equation. We actually dropped WLAN SIP
phones altogether, since they worked so poorly. Connection loss, bad audio
quality and low coverage range.
Just my 5 cents...
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Dear all,
I am looking for ip phone/ ATA that has built in VPN support. can any
one suggest me any brand or customize firmware ?
thanks
Salaque
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I've tried other phones and the issue does not happen. I've tried a
different IAX provider and it DOES happen... but only if the
jitterbuffer is on on the REMOTE side. I am currently working with
aastra to try to figure out if this is a phone or asterisk problem.
On 11/9/06, shadowym [EMAIL
For what it's worth, Apple's Mail application automatically embeds a tiny QuickTime Player interface in mail messages that contain audio attachments. The result looks like this:http://zachfine.com/blog/images/voicemail_sample_in_apple_mail.gifI'm sure it would not be too difficult to embed a small
Am Freitag, den 10.11.2006, 12:56 +0100 schrieb Mike Heininger:
Hi!
We have an installation with WLAN SIP phones only. Sometimes we have
connection drops. What is the best way to debug if we have problems
with the WLAN or the SIP devices or the uplink to the IAX Provider.
I'd go with
Anybody sucessfully got stable 1000Hz clock without Digium harware and kernel
2.6?
We need to consult some peoples how to clock asterisk stable with exactly 1000
Hz without much kernel/drives patching/tweaking.
Some test results we made so far:
2.6 with digium card - stable 1000 Hz.
2.6 with
Using Asterisk 1.2.12.1.
I have 4 queues running on a server with various handsets logged into them.
When a call comes in, asterisk tries forwarding the call to all
handsets, including ones that are in use (whereby it gets a BUSY HERE
response, which is all what you'd expect after all asterisk
Can you tell us how you do the testing ?
Zoa.
Anton Tinchev wrote:
Anybody sucessfully got stable 1000Hz clock without Digium harware and
kernel 2.6?
We need to consult some peoples how to clock asterisk stable with
exactly 1000 Hz without much kernel/drives patching/tweaking.
Some test
I'd go with parallel softphones on LAN-connected and/or WLAN-connected
PCs and see wether they have the same problem. That could rule out the
provider or confirm the WLAN or WLAN phone implementation make the
problem.
I think this is the next thing we will try next week.
TIA,
Mike
Zoa wrote:
Can you tell us how you do the testing ?
3-4 different ways. All gives same results, so test are pretty valid.
1. Interrupt counting inside the PC.
2. TDMoE packet counting on the switch.
3. External TDMoE equipment connected thru extreme network swich.
The card of the PC and the
I looked for a reference to do this for some time to replace the callout
feature in my old AVT voicemail.
I never found one, so I decided to dig in.
Here is my first run. It is in production, so unless I find a problem,
I am done.
Script set to run every 5 min. via cron.
This sets a lock
Hello all,
I am just wondering - how can I implement presence awareness in Asterisk?
I know there is the hint feature that might be useful (for someone) but
it is not exactly what I am looking for.
My idea is some fairly simple application running on user desktop and
having just 3-4 buttons like
Actually, while I was waiting for an answer, I figured out my problem.
If I have any further questions, however, I'll be sure to post. Thanks!
Jay
Dovid B wrote:
Post away.
- Original Message - From: Jay Moore [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Hi all,
[I'm new on this list. On other lists, I cringe at this type of query
because they sometimes end up in flame-wars. This is a serious request
because my time frame is about 2 months to identify, select, acquire,
install and setup our next PBX at a new office. I really would like it
to
n new stack
-- Executing Set("SIP/146-b78060b0", "CALLERID(all)=Mohamed
Samir - UNIX 146>") in new stack
-- Executing NoOp("SIP/146-b78060b0", "Using CallerID
"Mohamed Samir - UNIX" 146>") in new stack
-- Executing Macro("SIP/146-b7
Hello Jorge, and thanks for the answers, but:I don't understand what is a blind transfer and a supervised transfer.I mean, in the topology:- pstn line - norstar (ext 123) - ATA - (fxo zap/1) asterisk
An incoming call from the pstn line is forwarded by the norstar to extension 123 were asterisk
I have seen this before if the caller is on a cell phone with too high of an
audio delay.
There is a delay for them to hear the end of the prompt, and then a delay for
them sending the digits.
--
--
Steven
http://www.glimasoutheast.org
Jonathan Campbell [EMAIL PROTECTED] wrote in message
Am Freitag, den 10.11.2006, 14:35 +0100 schrieb Ondrej Valousek:
Hello all,
I am just wondering - how can I implement presence awareness in Asterisk?
I know there is the hint feature that might be useful (for someone) but
it is not exactly what I am looking for.
My idea is some fairly
Interesting!I think this can help for a start (but I don't know how to continue!!):[incoming]exten = s,1,Answer()exten = s,2,Backgroud(enter-ext)exten = _XXX,1,Playback(enter-name)exten = _XXX,2,Record(/tmp/prompt${EXTEN}:wav)
exten = _XXX,3,Dial(zap/1/${EXTEN})now, how to play the recorded
exten = _XXX,3,Dial(zap/1/${EXTEN},,A(somefile))
bails
Gustavo Berman wrote:
Interesting!
I think this can help for a start (but I don't know how to continue!!):
[incoming]
exten = s,1,Answer()
exten = s,2,Backgroud(enter-ext)
exten = _XXX,1,Playback(enter-name)
exten =
Hi Anselm,
Yes it looks promising.
somehow update the status in the Asterisk DB
and that's the problem - how can I access Asterisk DB remotely (in some
nice and elegant way)?
That's why I was more thinking about mysql - it is already running on my
* box and remote access is no problem.
Question
I would recommend you to record files with a uniqueid like var ${TIMESTAMP}[incoming]exten = s,1,Answer()exten = s,2,Backgroud(enter-ext)exten = _XXX,1,Playback(enter-name)exten = _XXX,2,Set(filename=${TIMESTAMP})
exten = _XXX,2,Record(/tmp/prompt${filename}:wav)exten =
Am Freitag, den 10.11.2006, 00:07 -0500 schrieb Jeronimo Romero:
I’m running asterisk 1.2.8. I would like PSTN inbound calls to do the
following:
1-once PSTN callers enter their desired extension; they have to record
their name
2-recording then announces that it is trying to locate
Am Freitag, den 10.11.2006, 16:33 +0100 schrieb Ondrej Valousek:
Hi Anselm,
Yes it looks promising.
somehow update the status in the Asterisk DB
and that's the problem - how can I access Asterisk DB remotely (in some
nice and elegant way)?
That's why I was more thinking about mysql - it
I am looking for ip phone/ ATA that has built in VPN support. can any
one suggest me any brand or customize firmware ?
I think the Zultys 4x5's are supposed to have built in IPSec VPN
support. Zultys was rumoured to be going out of business, though (not
sure if that's really true).
- Noah
Hi all,
Does anyone know if the Mitel phone features a webintreface for configuring the
phone?
Many thanks,
Christian
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Mohamed A. Gombolaty wrote:
Dear All,
I have an asterisk server version 1.2.12.1 along with trixbox and I
am having this nasty problem, I have a TE200P and have an E1 pri
attached to it and zttool says it's OK, I have configured the whole
31 channels into one group as follow:
On Fri, Nov 10, 2006 at 12:38:51AM +0100, Christian wrote:
Hi all,
Since i cant get latet beta of zaptel installed on the latest test
version of Debian with kernel 2.6.17-2-686 can someone who is using
debian give me some tips on how to get it working and installed?
Many thanks,
Christian
On 9 Nov 2006, at 16:16, Ira wrote:
At 05:00 AM 11/9/2006, you wrote:
Several motherboard manufactures in the last 3-4 years have had
capacitor problems, some reached the point of leaking others began
to cause problems on the machine after they began to swell. Both
Dell and IBM have
On Thu, Nov 09, 2006 at 05:44:23PM -0800, Darryl Dunkin wrote:
After running 'make install', do a 'depmod -a'.
Then check /lib/modules for the file:
find /lib/modules | grep zaptel
Be sure the path /lib/modules/kernel/extra/zaptel.ko matches up with
your currently running kernel (from
On Thu, Nov 09, 2006 at 05:25:02PM -0600, Eric ManxPower Wieling wrote:
Julian Varanini wrote:
Hi,
Can someone walk me through compiling and loading the Zaptel 1.2.10 driver
for Mandriva 2006 kernel 2.6.12-12? When I compile and attempt a modprobe
I get module zaptel not found
You
I have noticed it too and do not use them anymore..
Jon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ira
Sent: Thursday, November 09, 2006 11:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voxee lag
Is there anyway to get CDR to show just the answered calls. Not by
exporting to a spreadsheet and editing. We have ring groups and queues and
CDR shows everything as calls received. Even if it's multiple extensions
ringing it shows them as multiple calls received. This seems kind of goofy.
Tzafrir Cohen wrote:
On Thu, Nov 09, 2006 at 05:25:02PM -0600, Eric ManxPower Wieling wrote:
Is such horror normally needed with Mandrake? Doesn't Mandrake provide
working kernel headers linked from /lib/modules/`uname -r`/build ?
I have no issues with Mandriva 2006 or 2007 on compiling
Hi everybody,
I have this issue:
I need to automatically park an incoming call, play a welcome prompt and
then connect to some extension but under extension user's command.
I was thinking to use a small database to comunicate between asterisk and
the main application.
Has anybody had this kind of
shadowym wrote:
Is there anyway to get CDR to show just the answered calls. Not by
exporting to a spreadsheet and editing. We have ring groups and queues and
CDR shows everything as calls received. Even if it's multiple extensions
ringing it shows them as multiple calls received. This seems
Doug Lytle wrote:
Tzafrir Cohen wrote:
On Thu, Nov 09, 2006 at 05:25:02PM -0600, Eric ManxPower Wieling wrote:
Is such horror normally needed with Mandrake? Doesn't Mandrake provide
working kernel headers linked from /lib/modules/`uname -r`/build ?
I have no issues with Mandriva 2006 or
Mani,
I've gotten the same result both dialing from a gtalk client to SIP, as
well as an SIP call to gtalk. You can run a jabber debug before the
call is placed to see more debug info on what's causing the crash. With
the module in Beta, I believe it's just a bug that needs to be worked
out.
I have had no success in getting the voicemail working on Asterisk
1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried with or
without ztdummy device, renice -20 on asterisk process and even real-time
priority on the host Windows XP box for the vmware process. I am
Ciao Ondrej,
That's why I was more thinking about mysql - it is already running on
my * box and remote access is no problem.
Question is, if I could do the same trick you did with Asterisk DB
with Mysql.
Of course you can. In asterisk-addons there's the app MYSQL(), that
does exactly what
Hi All,
Thanks for your replies and help, I have this working now, TNT 11.0.6
and Asterisk 1.2.9.1, passing TNT SIP calls to Asterisk just fine.
Working through the solution was extremely painful, took a week in the
lab to figure out that I had my head shoved so far up my ass, I was
eating lunch
On Fri, Nov 10, 2006 at 03:28:09PM +0200, Anton Tinchev wrote:
Zoa wrote:
Can you tell us how you do the testing ?
3-4 different ways. All gives same results, so test are pretty valid.
1. Interrupt counting inside the PC.
2. TDMoE packet counting on the switch.
3. External TDMoE
show applications in the CLI is your friend. Look for parkandannounce
On 11/10/06, Mauro Zanin [EMAIL PROTECTED] wrote:
Hi everybody,
I have this issue:
I need to automatically park an incoming call, play a welcome prompt and
then connect to some extension but under extension user's command.
I
I'm running sippeers and sipusers in my extconfig, and everything runs
perfectly when a client is registered (ex. registers to port 1000), but
when it re-registers the client is set to port 5060. This behavior does
not take place if I use the static files.
Both in my sip_buddies table for db,
On Fri, Nov 10, 2006 at 03:04:46PM +0200, Anton Tinchev wrote:
Anybody sucessfully got stable 1000Hz clock without Digium harware and
kernel 2.6?
We need to consult some peoples how to clock asterisk stable with exactly
1000 Hz without much kernel/drives patching/tweaking.
Some test
Hi,
How can I use a command return code in my dialplan?
Example, I want to use the system command to run a perl script. This
script exists with a code that I need to use in my dialplan. But I can
figure out how to extract this value.
Thanks for any pointers.
Andre Courchesne
if anyone has one-way audio issues with iax over jittery connection,
please look at bug report, what I created yesterday and report your
experiences,
I think this is one of the most serious bug, that must be identified and
resolved before 1.4 will be released, thanks
Hi there!
I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris
20-20 PBX. More less everything went fine, but the problem I have now
is when dialing to the Harris PBX, it seems to pick up my call as
soon as it reaches it.
For example if from the Asterisk outgoing folder I
Hi,
did anyone managed to get chan-capi and app_pppd to work? Incoming
call is accepted, pppd started, but no data transfered to pppd.
I used app_pppd-060822.tgz, chan-capi 0.7.1, asterisk 1.2.13.
Error messages:
chan_capi.c:918 local_queue_frame: Could not write to pipe for ISDN1#0
Alyed Tzompa wrote:
Hi there!
I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris
20-20 PBX. More less everything went fine, but the problem I have now
is when dialing to the Harris PBX, it seems to pick up my call as
soon as it reaches it.
For example if from
Thomas Kenyon wrote:
On a slightly different tone, has anyone written a queue viewer that
runs as a daemon and serves the pages to the viewer rather than creates
a manager login/logout event every few seconds? (If not I'll write one
myself, but worth checking first).
Just written one, so
I have used WIP300, hitachi 5000 wireless phones on
asterisk and have had good success.
However, I am looking for a WIFI phone with integrated
belt clip. Has anyone found any?
I have tried after market clips and holders and those just
don't work.
THanks for sharing if someone has found
On Fri, Nov 10, 2006 at 03:45:18PM -0500, Jerry Geis wrote:
I have used WIP300, hitachi 5000 wireless phones on
asterisk and have had good success.
However, I am looking for a WIFI phone with integrated
belt clip. Has anyone found any?
I have tried after market clips and holders and those
I'm preparing to deploy a small number of Grandstream BT101's and
GXP2000's to a remote location (which I won't have access to). I'd
like to have them pull a config file from my server - I'm almost
there...
The phones are looking for the config file on my webserver which is
good. I
snip
Anton wrote:
2.6 with kernel clock - needs kernel recompiling and
work stable with switched off kernel Preemption. Long
time tests in progress now.
How are you doing this? I saw a couple developers talk
about rewriting ztdummy yo use the new hi-res kernel timers
(kernel2.6.17), but did
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Same here - wrote an email to support. They claim that their servers are fine and will get back to me in a day or two...On 11/10/06, Jonathan Borden
[EMAIL PROTECTED] wrote:I have noticed it too and do not use them anymore..
Jon-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL
options visit:
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Despite of monitor-join being equal yes, I get individual -in and
-out files for queue calls. My box runs Asterisk 1.2.10 and I've set
up real-time queues. Does anybody have any idea of what is going on?
Thanks in advance.
Carlos.
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i am sure this came up before
but all my searches are not resulting in anything usefull
trying to setup a grandstream phone
to connect to an asterisk server
now i am outside the network (home)
on my side
settings on the phone seem to be correct
id and password, astersik server ip, port
in
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On Nov 10, 2006, at 11:12 PM, Christian wrote:
Hi,
But what is the problem, why doesnt it install?
I am a little new to this so still learning.
Many thanks,
Christian
Use the latest 1.4 svn version instead of beta2
That will probably fix your problem.
I installed latest 1.4 svn checkout on
Everyone here is saying how it would be so great to have native
desktop/outlook/exchange/etc support, but seriously, do you really think M$
is going to develop these products to use with the open source market?
They're going to want to try monopolizing it and creating an environment
where you
On 10 Nov 2006, at 21:51, Rajeev Natarajan wrote:
Same here - wrote an email to support. They claim that their
servers are fine and will get back to me in a day or two...
Now there is a definitive case of a 'lagged' communication channel!
:-)
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
Steve Davies wrote:
*bump*
No suggestions at-all? Does anyone use this facility in a similar way
and NOT have problems?
Check the gain on your ISDN interface. The monitor command doesn't
modify the volume by default. Have you tested calls via IAX to your cell?
Leo
Nope I don't think for a moment they are going to encourage us to
integrate by making it easy.
This is why we need to develop more and more features (like the weather
app - you know you can ftp to text to voice any file right?)
The more features the more reason people will want to go with
a) Can we agree to stop using M$? Yeesh.b) The only trick to getting Asterisk to work is that Exchange 2007 is using SIP over TCP instead of SIP over UDP. Interestingly (and I just found this out myself) a product called Express SIP Router automagically translates UDP to/from TCP.
See this article
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Add a subject next time.
Are you behind a firewall where the Asterisk server is located? Have
forward ports 5060 and 1 - 2 UDP to the asterisk server?
On 11/10/06, Stas Khromoy [EMAIL PROTECTED] wrote:
i am sure this came up before
but all my searches are not resulting in anything
I am surprised that you have had good success perhaps you haven't done proper testing?On 11/10/06, Jerry Geis
[EMAIL PROTECTED] wrote:I have used WIP300, hitachi 5000 wireless phones on
asterisk and have had good success.However, I am looking for a WIFI phone with integratedbelt clip. Has anyone
Hello,
This is not exactly an Asterisk question, but I was encouraged to seek
advice here anyway. The kindness of the * open source community is
legendary :)
I am getting going with an Asterisk 1.2 box, and I'm having trouble
getting good quality transmit sound using handsets with VoIP phones.
Greetings,
Has anyone noticed that attempting to place a call from the Placed
Calls list on a Polycom IP501 by pressing the 'Dial' softkey sometimes
simply returns the phone to the idle screen? It is not related to the
number being dialed, as we have observed two entries for the same
number,
Add me to the list. Not only lagged, but also failures to register. AND, apparantly Paypal won't automatically authorize payments to them anymore. I'm not recharging my account anymore.
On 11/10/06, Tim Panton [EMAIL PROTECTED] wrote:
On 10 Nov 2006, at 21:51, Rajeev Natarajan wrote: Same here -
Hello if someone found some method to authentificate to asterisk with nokia push to talk clients please send me all your documentations and the tests results, I really need this for a project of main and i wanna dig deeper to solve this mister. Thank you guys for your cooperation. Alex i put you
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I'm trying to get app_swift (v0.9.1 from http://www.loopfree.net/app_swift/)
working, but it's having issues (see below). I'm running 1.4.0beta3 on FC6.
Any thoughts?
*CLI -- Executing [EMAIL PROTECTED]:1] Answer(SIP/spa3k-fxs-08e884b0,
) in new stack
-- Executing [EMAIL
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I want to add some sound filed on demand during a phone call only
possible on some extension numbers.
I get many phone calls from local companies, but don't understand
Chinese! I would like to record the call, but also ask the caller some
questions, which should be added into the call with
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