Hello everyone, I am pulling my hair here because a carrier threw me curve
early today.
They want to send calls to my asterisk server using SIP. Then they said
that their gateways don't have to register with my server, that all they have
to do is send a prefix for validation. Whereas
Hi,
Damon Estep wrote:
Here is the setup; analog phone Linksys ata asterisk sip
provider sonus GSX 9000 PSTN called party.
The caller on the analog phone connected to the ATA hears no echo at all.
The called party has a slight echo of their voice.
All of the Zapata.conf echotraining,
What they're asking you to do is quite insecure to be doing over public IP. At the very least, you should confirm that there is a static IP that these calls will be coming from and only accept calls from that IP, but that's still not quite as secure as digest authentication that would be available
I suppose the question is now whether you would recommend buying one
later,
PaulH
- Original Message -
From: Rod Bacon [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, September 13, 2005 5:22 PM
Subject:
On Mon Sep 12 10:32:47 CDT 2005 John Hill wrote:
I have 1 x100p. Caller id works fine with the beta1 release. Cvshead
releases fail with a combination of checksum and ss_thread errors?
I'm concerned when beta2 or the 1.2 release comes out it will not
work.
I have been through the configs I
Brave is the person that wants to use 3 Fritz cards in one box
Go with the Jurgens 8 bri or 2 quad Brior bri-e1 chan bank...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christoph
Eicke
Sent: Tuesday, 13 September 2005 6:11 PM
To: [EMAIL
Agreed - ATP are always good to deal with.
PaulH
- Original Message -
From: Callum McGillivray [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 14, 2005 12:41 PM
Subject: Re: [Asterisk-Users] Digium
On Tue, 13 Sep 2005, trixter http://www.0xdecafbad.com wrote:
On Wed, 2005-09-14 at 07:01 +0200, Remco Barende wrote:
Hi!
I'm considering to buy a GSM bridge to save on GSM calls. Right now they
are offering subscriptions with 200 minutes each month for almost nothing,
however the 400 minutes
On Wed, 2005-09-14 at 09:30 +0200, Remco Barende wrote:
Thanks for the tip. I was actually thinking in the direction of putting
the asterisk calling card application to use. I've never used it and
wonder if it is at all possible to use it from within the dial plan
instead of normally from
Hi!
My asterisk (1.0.7) is connected to a Nortel pbx with Digium E100P card, both
side are ETSI EuroISDN. I would like to reject an incomming call with cause code
34, but the Nortel PBX gets the value of 31 instead of 34. It seems to work on
the asterisk side:
Protocol Discriminator: Q.931
Probably easiest to set a variable to the number to be called and then
jump to an extension to do whatever you want to do?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruno
Voigt
Sent: 13 September 2005 23:37
To: asterisk-users@lists.digium.com
Hi :)
I hope someone can help me (google cannot):
My little asterisk receives calls via h323 from PSTN. I connected a Sipura
phone to my asterisk. oh323 is installed and calls go into the right context
but immediately after the phone is picked up a hangup is signalled and the
call ends :(
Hello,
I want to use call restriction option. For example, there
are 3 registered numbers that 100,200 and 300.I want 100 to call 200 but not
300, btw 300 can call both 100 and 200. How can i configure this?
Thanks.
Erdem HAKI
___
finally I did it - I put some of the vars in (double)quotes - this didn't
work
even if there's a space inside, the vars need not to be kept inside
(double)quotes...
You probably want to use 'database put' for changing incoming CID
http://voip-info.org/tiki-index.php?page=database%20put
you really should read about the concept of a context in extension.conf,
that will answer your question and is also a basic key to understanding
Asterisk.
http://www.voip-info.org is your friend.
Christoph
On Wednesday 14 September 2005 10:47, Erdem HAKİ wrote:
Hello,
I want to use call
Richard Kashdan wrote:
On Mon Sep 12 10:32:47 CDT 2005 John Hill wrote:
I am having the identical problem. I use the CVSHEAD Asterisk and do an
update every couple of weeks or so. I did one last week and the caller
id quit working on my two lines that have x100p cards. I didn't make
Well, a SIP authorization does not require a registration (in
fact, registration should be primarily used to inform a registrar about
thewhereabouts of a UA with dynamic IP address in order to handle incoming
calls_for_ that UA).
CS can just createfor his Asteriska "type=user"
entry in
Just to answer my own query, I needed to set the devices to dtmfmode=inband
in my sip.conf, and on the 7960 set Sip configuration - Out of Band DTMF -
none
The benefits of a good nights sleep :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mat
Hello all !
Can anyone recommend me ATA device that REALLY has T.38 built in.
So far I have heard of Telco Systems Access201, which seems to be
impossible to bye in Europe (all resselers are droped Telco systems ATAs for
some reason (tried in Germany and in UK so far)), and I have heard that
answer: SetCIDNum(0${CALLERIDNUM:2:20})
shows 20 digits of the number but strips the first 2, additionally a 0 is added
at the beginning.
yes, it is basic - but is it thoroughly documented somewhere? i'm sure that
there are lots of other syntax possibilities...
On Sat, 10 Sep 2005 11:58:32
On Tue, 2005-09-13 at 09:31 -0700, canuck15 wrote:
I don't recommend anyone use free dyndns via router support. If you reboot
your router more than once or twice in a month or have a power outage or
whatever dyndns stops updating the IP automatically and will cancel your
account for too much
Hi,
QueueMetrics version 0.9.5 rc 2, out today, does the trick and allows
agent pause monitoring (together with the rest of the stuff).
See http://queuemetrics.loway.it
Thanks
l.
In data Wed, 14 Sep 2005 07:28:51 +0200, Callum McGillivray
[EMAIL PROTECTED] ha scritto:
Hey Kevin,
take a look into the wiki...
http://www.voip-info.org/wiki-Asterisk+variables
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi everybody,
My Sipura device registers on an Asterisk server and works fine. Its
default registration time out value is 3600s. But I've noticed that once
in a while it stops receiving calls but dial out works fine. To solve
this problem I've to change registration time out value to 10s. Why is
It's very possible your firewall is closing the connection. When you
try to make a call it forces the phone to re-register. Are you using
STUN?
On 9/14/05, Zeeshan [EMAIL PROTECTED] wrote:
Hi everybody,
My Sipura device registers on an Asterisk server and works fine. Its
default
Nenad Radosavljevic wrote:
Hello all !
Can anyone recommend me ATA device that REALLY has T.38 built in.
So far I have heard of Telco Systems Access201, which seems to be
impossible to bye in Europe (all resselers are droped Telco systems
ATAs for some reason (tried in Germany and in UK so
Can anyone recommend me ATA device that REALLY has T.38 built in.
While I have not tested it myself (one just arrive for me try out), I
have been told that the Mediatrix products have a working T38
implementation. Of course my suggestion would be check with the
provider tho you plan to use the
The newest 2100 firmware has T.38.
What about other Sipura products like SPA-1001 and SPA-2002 ?
Does it really have to be the one with broadband functionality integrated ?
Thanks,
Ivan
___
--Bandwidth and Colocation sponsored by Easynews.com --
The MOSA 3700 family from
Vodtel have working T.38. They come from 2 to 16 ports. Can be bought on www.bobascom.com
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moody
Sent: den 14 september 2005 14:22
To: Asterisk
Users Mailing List - Non-Commercial
Kevin Bockman wrote:
I'm having some problems getting TDMoE setup for the 1st time. I have a
TE405P installed in the main server with an ethernet cross-connection
to the secondary machine.
(Yes, I know about IAX2 but I want to use TDMoE to simulate using T1s.)
I'm using -HEAD from
I'm wondering if anyone can recommend one over the other. I'm mostly
interested in running open source solutions, so I would prefer if
your recommendations are within the open source arena.
Basically, I contemplated the idea of using SER as a NAT Helper and
possibly as a SIP server for a
Hi.
I'm dialling two numbers - one that's unobtainable, one that's busy.
${DIALSTATUS} is coming back ANSWER each time right before the channels hang up.
Am using the following dialplan macro to dial out.
[macro-advdial]
exten = s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum
Hi,
Thank you very much for your suggestion this was what i nedded.
Best Regards
Accursio Avona
The question is, how can i indicate the marked user?
A quick search of the archives reveals:
Example:
meetme.conf
conf = 1000
extensions.conf
; ** Normal users enter the conference
I currently have:Telco-PRI Panasonic DBS576 PBX EM wink
T1 Asterisk.
I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk
extensions over the T1.
I do not get DID nor CID on the Asterisk, so I want to use PRI between the
PBXs.
I do not want to pay for
I would be most interested in seeing some TNT/APX configurations and
corrosponding SIP configurations for Asterisk.
Right now, I'm using call routes and switching off a T1/PRI to my
asterisk box, and would love to change that to pure SIP if possible.
The only caveat is that my TNT boxes are
I can confirm that sipura spa-2100 has t.38 suppurt from firmware 3.2.1
and it seems to work fine in our test with some t.38 providers.
Bye
Rosario
- Original Message -
From: Nenad Radosavljevic [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 14, 2005
If you have a linux box, then u can try sip-nat-helper for netfilter.
Cheers.
Mensaje citado por: Waldo Rubinstein [EMAIL PROTECTED]:
I\'m wondering if anyone can recommend one over the other. I\'m mostly
interested in running open source solutions, so I would prefer if
your recommendations
ah, i see. didn't stumble over this yet, thanks!
On Wed, 14 Sep 2005 13:48:37 +0200
[EMAIL PROTECTED] wrote:
take a look into the wiki...
http://www.voip-info.org/wiki-Asterisk+variables
___
--Bandwidth and Colocation sponsored by Easynews.com --
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Doug Lytle
Sent: Wednesday, September 14, 2005 4:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Callerid fails in any release
after beta1 fails
On Monday 12 September 2005 01:56, Ronald Wiplinger wrote:
I would like to know if somebody else experienced that:
sip show peers will always drop the first character of the 11th line.
while sip show peers like [0-9,a-z] will not drop any character.
Can anybody test this, please?
I think STUN is quite widely supported by hardphones. I'd be interested
to know if STUN is a magic fix to SIP NAT - I've a feeling that its not.
Derek
Waldo Rubinstein wrote:
I'm wondering if anyone can recommend one over the other. I'm mostly
interested in running open source solutions,
On Wed, 14 Sep 2005, Dinesh wrote:
I was wondering if its possible to hook up eyebeam with video support to
[EMAIL PROTECTED]
Yes.
But eyebeam's video support is pretty rudimentary. it doesn't show
inbound video at all until you start sending yours. people with eyebeam
but no camera
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example 100
102 etc. but when i truy to go outside with the 9 before the call rings
in the first extensions (100). this is a asterisk problem? or a pbx
It could potentially be both. I would look at your extensions.conf first
though. What does the extension entry for that context look like.
For instance I have an entry in my extensions.conf for dialing outside
lines (outside being from asterisk to my PBX and then onto the outside
world from
unless you show us some config files, I doubt that anybody can help you...
On Wednesday 14 September 2005 16:46, Pablo Allietti wrote:
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example 100
102
On 9/2/05, Chris A. Icide [EMAIL PROTECTED] wrote:
Dana Olson wrote:
Chris,
Thanks for the reply.
I checked those settings, and they were commented out, so I uncommented
them. I assumed you meant rtnoupdate=yes, so that's what I put, but
that didn't work. I tried rtnoupdate=no, and
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Troy Settle
Sent: Wednesday, September 14, 2005 7:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MAX PRI for single server (was:Not
Hi,
I've setup a queue with 3 sip members.
I've tried with random and roundrobin and different timeout settings in
musiconhold.conf
Always after the second Nobody picked up in 15000ms I get
Exiting on time-out cycle
Stopped music on hold on CAPI/contr1/s-0
Where can I increase this timeout?
como'n folks.. ...
Well, as I told earlier.. my asterisk was running great with one fxo and
one
fxs module of a TDM400P
All i tried last night to run asterisk with non-root
I must did something wrong while I was trying to do that
FXO module on channel # 1
FXS module on channel # 4
Hi !
Asterisk sends a RELASE COMPLETE with cause code 34. It seems that
Nortel expects a RELEASE message in this state. The conversion
is done in the protocol engine of the MSDL.
Why would you want the cause code 34 to be sent ? Do you need a
special rerouting on the Nortel side ?
Would it be a
Hi,
I have some experience in sending SMSs using smsclient.
I call the german Vodafone SMSC (01722278020),
and smsclient takes approx 20 secs to send a SMS.
The hardware is an Sedlbauer ISDN card.
Now, I want to do the same using asterisk and a digium PRI card.
I dialed using the manager with:
On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote:
ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in
the pbx. and all incomming calls go to 100. thats the problem i will
try to solve this.
It could potentially be both. I would look at your extensions.conf
I have 2 servers that I use to talk from one place to another place. One
of them, Server A registers with the other one, Server B. There are
many cases the registration drops out and then works again after some
time. The internet connection between them is not so great, which could be
suspected.
On Wed, 2005-09-14 at 08:10 -0700, Naren Koka wrote:
I have 2 servers that I use to talk from one place to another place. One
of them, Server A registers with the other one, Server B. There are
many cases the registration drops out and then works again after some
time. The internet connection
I've been evaluating asterisk for quite some time now and am attempting
to create services on it. The system is simple right now. asterisk
seems to look up atleast every week if not more. I am running asterisk
1.0.9 and would like to find similiar experiences of long term
stability.
I attempted
John Hill wrote:
I deleted all modules and did a make install of the beta1 source using the
cvshead of zaptel and libpri.
Caller id then works fine?
Something has changed in the asterisk code that is not seeing callerid from
of my x101p.
I was thinking about doing a fresh install this
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Doug Lytle
Sent: Wednesday, September 14, 2005 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Callerid fails in any release
after beta1 fails
Kevin P. Fleming wrote:
Andy Howell wrote:
I have a weird problem in which my digium card stops answering. After
running for a couple days, incoming calls are not seen. Running asterisk
-r shows no incoming calls. Restarting Asterisk does not help. After a
reboot it is fine.
This problem
Leonardo Gomes Figueira wrote:
Hi,
Andy Howell wrote:
I have a weird problem in which my digium card stops answering. After
running for a couple days, incoming calls are not seen. Running asterisk
-r shows no incoming calls. Restarting Asterisk does not help. After a
reboot it is fine.
Any
On Tue, Sep 13, 2005 at 12:01:21PM -0800, Mojo with Horan Company, LLC wrote:
hisax seems to be a loadable module for an ISDN card. if:
# lsmod | grep hisax
prints any output, try
# rmmod hisax; modprobe zaptel
What information does kudzu use? Why doesn't it know that those PCI IDs
Title: TE110P - [EMAIL PROTECTED] Install Problems
I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and
i guess is usefull a neighcompany context, where you will allow users
to call other companies, using a company prefix. I need more info about
your real dial patterns in order to suggest something more specific.
best regards
On 9/13/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I have to design
Andy Howell wrote:
Its 1.0.9, as part of [EMAIL PROTECTED] 1.3
Then I would suggest upgrading to 1.0.9.1 or the just-released 1.0.9.2.
___
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Asterisk-Users mailing list
mmm actually i think that is a functionality most VoIP phones provide,
you dont need do anything, just press transfer in your VoIP phone and
the dial the extension you want to transfer to.On 9/13/05, PJ Santos [EMAIL PROTECTED] wrote:
Hi All,
I need help to create one IVR Menu, when a say Welcome
- Original Message -
From: Rosario Pingaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 14, 2005 9:07 PM
Subject: Re: [Asterisk-Users] T.38 ATA
I can confirm that sipura spa-2100 has t.38 suppurt
Well I don't know how you could measure long term stability at the
moment since 1.0.9 has only been out for about 2 months, but I can offer
some insight on older versions.
We have one 1.0.3 box that has been up and running for 27+ weeks without
an issue. It is running SIP for ~ 30 phones and
I need create one configuration to provide one Interactive Voice Response.
I read any docs about this.
So, if you have one sample, please post.
Thanks.
Paulo Santos.
Brasil-RJMoises Silva [EMAIL PROTECTED] escreveu:
mmm actually i think that is a functionality most VoIP phones provide, you
Hi !
First of all thank you all for fast response on matter of T.38 capable ATAs.
I have asked a UK VoIP suplier to check with manufacterers of various ATAs
they sell, do they support T.38 and here is what they/I have got as a
result:
1. Sipura SPA-2100 only and with firmware 3.2.1 is T.38
Hello, all!
I'm looking at the wiki page and info on the mailing list and I'm getting
conflicting info...
I am using the manager API from the telnet CLI and I am testing creating calls
with it. I login with events: on and I can originate calls just fine.
However, when I set ActionID on an
Disclaimer: Not a troll
I'm curious as to this obsession with uptime is. All of the posts of this
type are along the lines of After X days, Y thing does not work but if I
reload or reboot, it's OK - so why not cron a reboot? Is it considered bad
form or something like that? I reboot every night
This is a sample that I built as part of our * pilot here - it
demonstrates the various things you can do with an auto-attendant type
of system. Is this the kind of thing you are looking for?
[info-line]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten =
I just experienced something I'd rather not experience again.
Using a SPA-841 SIP phone connected to our Asterisk server, someone
dialed their own extension, answered, and then transferred the call
using the phone's XFER soft key. This does a SIP REFER.
Now, the phone has dropped out of the
LOL - Congrats!
$30 down...
Let's see... how much to go?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Monday, September 12, 2005 1:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users]
I would recommend it for tech-savvy people right now. It's a bit klunky
in the interface, but the phone functions (dial, receive, cid) work
great and the sound is clear in both directions. the setup through the
phone interface is a little repetitive and slow (albeit probably great
for avid
Hello. Im trying to get Fax-to-email working.
I've installed Rx and txfax, spanDSP and every package needed. I've done
everything on this page (altough, some bash-scripting problems):
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fax+to+email
anyway, when i try to send an fax, i get
How is this insecure? Most large business
and wholesale providers use only IP authentication, relying on a session border
controller to do the authentication work resulting in great scalability on the
softswitch (since it does not have to act as a proxy as well).
If they know your IP,
We have Asterisk 1.0 (CVS-v1-0-12/28/04-03:08:11 built by [EMAIL PROTECTED] on a
i686 running Linux) and as a safe countermeasure we do a cron reboot every
week. On four different locations.
No more crashes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Hi All,
I have a small setup with 2 SPA3000 1 SPA2001 and 1 Polycom 301
The Polycom misses 1 out of 2 dialout calls, this is the full log from a
call which didn't go through.
303091 Sep 14 10:45:15 VERBOSE[15427]: -- SIP/pstn_2-1f35 answered
SIP/200-0db1
303092 Sep 14 10:45:15
Posso ajudar?
Fábio Sakai
DGX -
Digital Express
Suporte CosmoCall
[EMAIL PROTECTED]
+55 11 3049.8109
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de PJ Santos
Enviada em: quarta-feira, 14 de
setembro de 2005 13:16
Para: [EMAIL PROTECTED];
Asterisk Users
We have another box that is running 1.0.7 with H.323 to an H.323
gatekeeper and it is just acting as voicemail for a Cisco Call
Manager.
It crashes at least 1-2 times per week. Starting asterisk again
brings it back up. I don't know why it happens and I have been unable
to get anything
Thisisverybasicprogrammingandisexplainedinatutorials
is you have a sip phone you will have a
transfer or flash button so you can transfer any call to
another
a ivr menu is very simple to
exten =
s,1,Answer
exten =
s,2,Background(audiofile..)
exten =
1,1,Dial(sip/100)
exten
Hello. Im trying to get Fax-to-email working.
I've installed Rx and txfax, spanDSP and every package needed. I've done
everything on this page (altough, some bash-scripting problems):
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fax+to+email
anyway, when i try to send an fax, i get
In queues.conf
; How long do we let the phone ring before we consider this a timeout...
;
timeout = 15
But this is just the function how long the phones will ring you should not
set this option to long or your phone will stop ringing if a timeout is set
in your phone
But when the line
Tony Mountifield ha scritto:
It works for me (using CVS HEAD, but I'm sure it's worked in the past for
me on Stable too). I think there must be some other reason it's not working
for you.
Just done a little test for it, as follows...
My extensions.conf:
[vartest]
exten =
i think you need a restart, then:
[your-local-extension-context]
exten = _,1,Gotoif([${CALLERIDNUM}=${EXTEN}]?2:4)
exten = _,2,Playback(you-are-a-frigging-idiot-stop-that)
exten = _,3,System(/etc/asterisk/email-administrator-moronic-behavior
${CALLERIDNUM})
exten =
about spa-2100, the t38 stream is on UDPTL and so asterisk passthrough
doesn't work.
- Original Message -
From: Nenad Radosavljevic [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 14, 2005 12:16 PM
Subject: [Asterisk-Users] Re: T.38 ATA
Hi !
Hi Andres -
I have a small setup with 2 SPA3000 1 SPA2001 and 1 Polycom 301
The Polycom misses 1 out of 2 dialout calls, this is the full log
from a
call which didn't go through.
303094 Sep 14 10:45:15 DEBUG[15073]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 2: Found
On Wed, 2005-09-14 at 19:15 +0200, Arne Morten Johansen wrote:
Hello. Im trying to get Fax-to-email working.
Didn't I see that exact same message exactly 29 minutes ago?
That's the best way _not_ to get an aswer on this list.
--
Dave Cotton [EMAIL PROTECTED]
lol that was me ironic that I just hijacked this thread and said that
reboots are not a bad thing! It's true I do have 30 IAX/SIP boxen that I
don't reboot, they are all slave servers to the IAX/SIP/PRI master
server, which I *do* reboot every night. The 30 boxen I did by cloning a
single hdd and
This is not a siemens pbx problem you set the
pridialplan = to national and that adds a number to the outgoing call or
something just use
Pridialplan = local
prilocaldialplan = local
and it should work
I tried to open the file kds again and now it showed me your configuration
:) don't know why
On Wed, Sep 14, 2005 at 10:45:36AM -0500, Robert Wagner wrote:
hi te110p is a et1 card. your sigfnalling is wrong i think
i have the same card and is work with this conf
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone= us
defaultzone = us
Hello asterisk-users,
Just curious if anyone has the indications for Ireland, tried
googling for it to no avail.
Sean
--
+---+
|VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie |
|GPG Key http://thecivvie.fastmail.fm/thecivvie.asc |
Have a customer with a fairly large scale project that needs to get
done, yesterday. Not sure how they thought they would be able to
complete this internally, but they have basically a week or so to pull
this off. Here is a list of requirements, if someone is interested in
taking this on,
Anyone know how to fix this?
gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff
In file included from app_rxfax.c:14:
/usr/include/asterisk/lock.h: In function `ast_mutex_init':
/usr/include/asterisk/lock.h:302: error:
`PTHREAD_MUTEX_RECURSIVE' undeclared (first use
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
We have another box that is running 1.0.7 with H.323 to an H.323
gatekeeper and it is just acting as voicemail for a Cisco Call Manager.
It crashes at least 1-2 times per week. Starting asterisk again
brings it
I took a recording that was in 41k sampled mono wav. Did the sox
file.wav -r 8000 file.gsm resample ql
took an 8K record in wave did sox file2.wav file2.gsm
Both of them have introduced a hissing noise.
If I play the wave files they sound fine.
How do I remove or reduce the hiss.
Jerry
I am game.
What do you need from me???
Locked, loaded and ready to GO!!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory
Andrews
Sent: Wednesday, September 14, 2005 2:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL
I have been testing the ASTCC and have notice that when the caller hangs up
the line while the balance is being played back the sub savedata() is not
being called because the asterisk terminates the AGI and the rest of the
script does not get executed thus never returning:
AGI Script astcc.agi
On Wed, Sep 14, 2005 at 07:52:26PM +0200, Sander wrote:
This is not a siemens pbx problem you set the
pridialplan = to national and that adds a number to the outgoing call or
something just use
Pridialplan = local
prilocaldialplan = local
and it should work
no uuuaaa the same
I've had an spa3k in service here at the house for a while now. After
some initial wrangling, it's been working okay. I've had to reboot it a
couple times and have noticed something rather annoying though.
My setup is pretty simple and, dare I say, common. I have the SPA-3000
inline between my
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