Mark,
What you are doing is indeed wonderful and I applaud you for your efforts!
I will be sending in my details in an email soon. By the way if you
(or any other non profit organisation for the matter) are interested
in similar efforts going on in Asia do drop me an email as I am based
in
Try playing with faststart .
Moises Silva wrote:
Hi Carlos. I have never used H323. But im interested in your problem.
Have you tried to use de 'sip debug' 'iax2 debug' commands? and check
the console with a high verbosity level? could you post any warning or
relevant output when the call is
Could you go with some details? What was better performance, stability?
All our user info is in MS SQL and we have billing based on it, so it won't
be easy to move to mysql.
I.N.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shamsul Arefin
Sent:
On 20:00, Mon 13 Jun 05, Frank Cases wrote:
current setup
SIP phone 192.168.1.30 -- linksys wrt54g sveasoft -- INTERNET --
(xl0) Firewall (xl2:172.16.0.50)-- (em1:172.16.0.101) Asterisk
problem is RTP stream not oging trouhg from * to sip and vice versa.
#1 and asterusk is pushing
If you do a sip show peers I think you will see that your PAP2 setup
registers its port with * as being 5060 on line 1 and 5061 on line 2,
but it stills calls port 5060 on asterisk when it makes the
registration.
I think * is actually listening on the first configured port.
You might get the
Forget about MS SQL, odbc drivers that run on linux to talk to MS SQL
stink Odbc in general stinks.
You might be able to get MS SQL DTS (data transformation services) to
link to the mysql database and present the data as it were in your ms
sql database.
There is a pretty good odbc 3.51 mysql
I just ran a couple of test with CVS Head
Port=5060
Port=5061
Result = chan_sip reports listening on 5060
Port=5061
Port=5060
Result = chan_sip reports listening on 5060 (ignoring port=?)
Port=5061
Result = chan_sip STILL reports listening on 5060
Bindport=5061
Result = chan_sip reports
Thanks for info . How do you integrate * specific data in mysql with data
from MSSQL? App is running on .NET, in this case it will need to have
assess to both DBs and update them simultaneously. Sorry, I'm not a DB
admin.
I.N.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi,
What about using IPTABLES DNAT stuff in order to map all incoming
5061 traffic to 5060 port ? That may work.
On 6/14/05, Damon Estep [EMAIL PROTECTED] wrote:
Conclusion - asterisk only listens on one port, and ignores the second
port= or bindport=
--
Juanjo sin .sig :(
If your app is .net get the .net provider for mysql and give it to your
dba/programmer with the docs, he/she will figure it out. No different
than talking to ms sql with .net except you reference a different data
provider.
-Original Message-
From: [EMAIL PROTECTED]
You must have missed the part where Prepaid got upset when I suggested
workarounds :)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Juan J. Sierralta P.
Sent: Tuesday, June 14, 2005 12:46 AM
To: Asterisk Users Mailing List -
Hi,
i try again to ask this.
When i use the *8 for the call pickup the call i fetch is directly
connected and i can't see the callers number.
What i want is that the call in the first only rings at my phone and in the
second i can see the callers number before i am connected.
I am using a
Hi,
I just noted that :(
Anyway maybe is not so hard to modifiy the code to handle
multiple ports, I believe this would give a cleaner solution, only if
Prepaid can code ;)
BTW: Maybe everybody has this problem but sometimes when I post I
receive a message from Mr. Zimmermann who seems
Title: Message
Hi,
I have
set autocreatepeer=no and it behaves just the same.
It
seems that the default value is no, or Asterisk does not understand this
property.
In
which version of Asterisk was this property introduced? I use
1.0.5.
Since
this didn't help, can help with another
Stojan Sljivic - GDS wrote:
Hi,
I have set autocreatepeer=no and it behaves just the same.
It seems that the default value is no, or Asterisk does not understand
this property.
In which version of Asterisk was this property introduced? I use 1.0.5.
autocreatepeer is off by default and
Hi Olle,
Do you have any idea why is Asterisk behaving like this?
Did you tested this with your Asterisk?
Regards,
Stojan Sljivic
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Olle E. Johansson
Sent: Tuesday, June 14, 2005 9:53
To: Asterisk
Stojan,
You have to check what extensions you have enabled in the context
specified in the sip.conf [general] section. All of those will be
reachable without authentication by anyone.
If there's no context= setting, Asterisk defaults to the context
named [default] in your dialplan. Any extensions
Before change OS try to do next steps:
first, stop asterisk. Second, you must do ztcfg -s to shutdown
span. Unload modules, load modules if you need and do ztcfg -vv again.
Start asterisk
Regards
Srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En
Hi,
priindication = passthrough in zapata.conf is your friend. :)
You need to BRIstuff your * though...
best regards
Klaus
Am Freitag, den 10.06.2005, 12:17 +0200 schrieb Diego Ercolani:
Hello all,
I've a little clue with zaphfc used to connect to a BRI linethat probably can
be a
Hi,
I'm facing something
strange but maybe I haven't the right solution.
What I want ot do is
:
Someone from outside
call my phone number, I check some informations using an IVR script and then I
want to transfer the callto an externalphone number. The point
is that when I'm doing
Hi
We're having some problems with max retries exceeded errors using IAX2 which
causes dropped calls. Sometimes happens with Firefly softphone, now 1.9.9 (the
current one) but has also happened with a hardphone we use (IAXtel). This is
just for the internal connection between our desktops and
Hi all,
I am looking for a sample configuration for chan_h323 or chan_oh323
that enables a static CLID for all incoming H.323 calls with a prefix.
I have configured H323 channel and registered it to a GK as a GW.
There is no call issues for the moment. However, I would like to have
incoming
I'm trying to make H.323 trunk between AVAYAAsterisk. But call from
AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started.
Does any one use AVAYA and h.323 channel?
Thanks Bob.
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Hi,
I have this error, I have a digium card TE110P Tiger3xx
When I'm load the dirvers by this command modprobe wcte11xp I got this
error
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
1 error(s) detected
FATAL: Error running install command
hi,
On Tue, 2005-06-14 at 11:35 +0200, Yousef Herzallah wrote:
/etc/zaptel.conf
span=1,1,0,esf,b8zs
bchan=1-15,17-31# set this to 1-15,17-31 for E1
dchan=16# set this to 16 for E1
defaultzone=it
loadzone=it
for an E1 in italy you should use ccs,hdb3 and
Hi,
I want to connect via SIP to opentelecoms.org.
They sent my the login data and gateway settings
for XLite, supporting NAT and STUN.
My asterisk box has a permanent IP address without NAT.
Does anyone know, which gateway addresse to use?
register = userid:[EMAIL PROTECTED]/userid
does work
and also
On Tue, 2005-06-14 at 11:35 +0200, Yousef Herzallah wrote:
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
seems you have udev installed, see README.udev in zaptel src dir
(and use zaptel.init to start zaptel)
matteo.
--
Hi,
I would like to setup Asterisk to route incoming calls to ZAP on
my TDM400P to SIP phones. What is the best dial plan to use.
We are currently able to route outgoing calls to PSTN from SIP
to ZAP.
Thanks in advance.
Walid
___
Asterisk-Users
On Tuesday 14 Jun 2005 09:16, David Masure wrote:
Hi,
I'm facing something strange but maybe I haven't the right solution.
What I want ot do is :
Someone from outside call my phone number, I check some informations
using an IVR script and then I want to transfer the call to an external
On Tuesday 14 Jun 2005 11:45, Bob Goddard wrote:
On Tuesday 14 Jun 2005 09:16, David Masure wrote:
Hi,
I'm facing something strange but maybe I haven't the right solution.
What I want ot do is :
Someone from outside call my phone number, I check some informations
using an IVR
I have this setup:
SIP Phone (xten) - Linksys - Internet - PIX - Asterisk
I can get 5060 working with no prob (PIX hasa helperbuilt in) but I need to forward RTP 8000 from my linksys to my SIP phone. Is there anyway around the forward? It would be nice to have multiple phones working but I wont be
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Maybe an example will be usefull:
voip services -- * (5600) -- cme (601) -- * (5901) -- * (voicemail
u601)
voip services on sip.conf
- -
register = 530:[EMAIL PROTECTED]:5061/5600
[5600]
type=friend
I've bought bunches of these: http://www.tigernetcom.com/products_USB_100.html
they work great. Very handy.
Paul
[EMAIL PROTECTED]
I'm trying to find a voip-suitable USB headset (I.E. headphones +
microphone) which I can use with my laptop while I'm traveling and using
Firefly or another
I want to establish voice overip between two locations in Turkey. But I don't know where should I start actually. Can any one tell me what can I do first.__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around
On Tue, June 14, 2005 13:04, Paul Mahler said:
I've bought bunches of these:
http://www.tigernetcom.com/products_USB_100.html
they work great. Very handy.
Paul
[EMAIL PROTECTED]
I'm trying to find a voip-suitable USB headset (I.E. headphones +
microphone) which I can use with my
What do you want to knwo exactly ?
The way I use to transfer the call is the same if it was an internal
extension, I use the Dial command using the trunk to go outside ...
Ask me the info you need ...
Thanks
David
-Message d'origine-
De : Bob Goddard [mailto:[EMAIL PROTECTED]
Envoy :
There is an USB softphone or MP3 softphone that you may find useful.
The USB softphone in size of USB flash disk comes with built-in sound
drive. It can embed a softphone such that it is portable anywhere even
an computer does not equip with a sound card. It is also a USB flash
disk that can be
Now I have this errors
When I lunch this command I got modprobe wcte11xp
This error:
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
1 error(s) detected
FATAL: Error running install command for wcte11xp
And when I lunch this command ztcfg
Hi,
I am trying to use the attended transfer. So I put this in my feature.conf:
[general]
[featuremap]
atxfer = *0
blindxfer = #0
I completly restart asterik, and not just make a RELOAD. But during a
call, when I press # it runs a blind transfer and if I press * I am
disconnected.
I am using
Google is very useful tool for research, if
you are interested in installation just write installation+asterisk+fedora
... etc . It works : )
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nevruz Mesut Sahin
Sent: Tuesday, June 14, 2005 2:11
PM
To:
Hi,
-Original Message-
SIP Phone (xten) - Linksys - Internet - PIX - Asterisk
I can get 5060 working with no prob (PIX has a helper built
in) but I need to forward RTP 8000 from my linksys to my SIP
phone. Is there anyway around the forward? It would be nice
to have multiple
On Sat, Jun 11, 2005 at 08:19:58AM -0400, Mike M wrote:
On Sat, Jun 11, 2005 at 02:03:59PM +0200, Michiel van Baak wrote:
On 13:22, Sat 11 Jun 05, Serge Schumacher wrote:
What platform should you suggest to use asterisk ?
I love the way the Debian updates work.
Me too, but has the
On 13:57, Tue 14 Jun 05, Alexis FECOURT wrote:
Hi,
I am trying to use the attended transfer. So I put this in my feature.conf:
[general]
[featuremap]
atxfer = *0
blindxfer = #0
snip/snip
exten = _10X,2,Dial(SIP/${EXTEN},20,htT)
exten = _10X,3,Hangup
Hi,
The problem is in that h
Hi list,
For months everything worked super here in our setup.
This week I implemented some new idea in our webbased
calendar system. I thought it would be nice to have an
option that tells asterisk you are not available for calls
during an appointment.
For this to work I could no longer use the
You guys have me second guessing my training and experience in this
area, so;
1. If I am wrong I apologize to the group.
2. I have been trying for a few minutes to find confirmation either way.
From what I know about the modulation techniques used by DSL (DMT, CAP,
QAM) it is impossible
On 14:26, Tue 14 Jun 05, Michiel van Baak wrote:
Hi list,
Forgot to tell I am using:
Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k, Copyright (C)
1999-2004 Digium.
build from source on a Debian Sid machine
Michiel van Baak
___
Asterisk-Users mailing list
Title: Message
Hi,
I use
Asterisk 1.0.5 and TDM04B.
When
an incoming call over ZAP channel hangs-up, it takes 10 seconds until Asterisk
realize that.
How
can I shorten the time of hang-up detection?
Regards,
Stojan
Sljivic
___
Asterisk-Users
Title: Message
Hi,
I'm
using TDM04B and Asterisk 1.0.5.
How
can I setup the Asterisk so that I get caller ID?
I do
not get caller ID currently.
Regards,
Stojan
Sljivic
___
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Asterisk-Users@lists.digium.com
On Monday 13 June 2005 22:54, Eric Rees wrote:
Correct me if I am wrong. I can remember installing a T1's with a HDSL
unit at the last CO, in which the T1 was delivered to the customer's
prem in two wires. I think they called this fast half-duplex.
Just because it's two-wire doesn't mean
Hello
We have around 50 phones in our company, and I am playing with the
thought to gradually go over to using sip services and ip-phones
internally. However at first I would liked the Asterisk just to sit
between the phone line and the Panaosnic, so I can take out one
lin/number at a time to use
Hi Olle,
Thanks for the info.
I have put sip.conf [general] section's context=default.
In default context I hang-up all calls.
This way only peers registered in sip.conf with context=sip will go to the
sip context that will allow them to call some extensions.
However now I have a problem with
On Monday 13 June 2005 21:14, Nir Simionovich wrote:
do a little math (23+1)*64 = 1536kbps = 1.536Mbps, hence the speed for a
single T1 circuit.
Your math's a little off.
T1 = 24 8-bit channels + 1 frame bit sent 8000 times a second.
24*8 = 192+1 = 193 bits sent 8000 times a second =
On Tuesday 14 June 2005 02:04, Michiel van Baak wrote:
On 20:00, Mon 13 Jun 05, Frank Cases wrote:
current setup
SIP phone 192.168.1.30 -- linksys wrt54g sveasoft -- INTERNET --
(xl0) Firewall (xl2:172.16.0.50)-- (em1:172.16.0.101) Asterisk
problem is RTP stream not oging trouhg
I would like to setup a test number, that speaks back my phone number.
How can I set this up?
bye
Ronald
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Hello,
Just want to tap the collective wisdom of this list as to experiences
pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters...
Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be
the top of the pick..Any comments and experiences esp. with Asterisk
I used to use the following but Festival is such a load hog I just
NoOp the same info and read off the console.
exten = 789,1,Festival('You are currently calling into context: $
{CONTEXT} as name: ${CALLERIDNAME}. number: ${CALLERIDNUM}.
channel: ${CHANNEL}. This is extension: ${EXTEN}.')
Well,
None of us is nitpicking here, are we? ;-)
Nir S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, June 14, 2005 2:56 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5
On Tuesday 14 Jun 2005 14:30, Bryce Chidester wrote:
I used to use the following but Festival is such a load hog I just
NoOp the same info and read off the console.
exten = 789,1,Festival('You are currently calling into context: $
{CONTEXT} as name: ${CALLERIDNAME}. number: ${CALLERIDNUM}.
LOL - Well, I think we all know a little more about DSL and T1 now at
least
Cheers all,
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nir
Simionovich
Sent: Tuesday, June 14, 2005 6:42 AM
To: 'Asterisk Users Mailing List - Non-Commercial
Ronald Wiplinger [EMAIL PROTECTED] uttered the following thing:
I would like to setup a test number, that speaks back my phone number.
How can I set this up?
A simple way to do this, extension 222:
exten = 222,1,Playback(privacy-your-callerid-is)
exten = 222,2,SayDigits(${CALLERIDNUM})
exten
Hi all.
Could someone point me an example to use SIP_HEADER function!? I want
to read the To: and send this INVITE to an internal extension.
Is there anybody using this function!?
Tks.
Denis Galvão
___
Asterisk-Users mailing list
On Tue, Jun 14, 2005 at 03:06:52PM +0300, Tzafrir Cohen wrote:
On Sat, Jun 11, 2005 at 08:19:58AM -0400, Mike M wrote:
On Sat, Jun 11, 2005 at 02:03:59PM +0200, Michiel van Baak wrote:
On 13:22, Sat 11 Jun 05, Serge Schumacher wrote:
What platform should you suggest to use asterisk ?
Yeah, but ION was a great service! I have it for year in my home office, the four line & two fixed IP version. What a deal. Way ahead of its time.
When they cancelled the service the buyout of the contract basically covered everything that I had paid for the prior year.
Michael
Wai-Sun Chia [EMAIL PROTECTED] wrote:
Just want to tap the collective wisdom of this list as to experiences
pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters...
Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be
the top of the pick..Any comments and
Hi all,
Is VOIP-info down?
Marcel van Kaam
Fonetica Teleservices
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On 13:57, Tue 14 Jun 05, Alexis FECOURT wrote:
Hi,
I am trying to use the attended transfer. So I put this in my feature.conf:
[general]
[featuremap]
atxfer = *0
blindxfer = #0
snip/snip
exten = _10X,2,Dial(SIP/${EXTEN},20,htT)
exten = _10X,3,Hangup
Hi,
The problem is in that h
Sure. I thought it was going to be a great service
and it was for about 10 days. The point is this.
You are telling me that the whole time they were installing
my circuit, no one knew that thedemise of ION was imminent? How
unscrupulous is that? To knowingly sell aservice that is
Seems to be all morning. I have not been able to access for several
hours now.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcel van
Kaam, Fonetica
Sent: Tuesday, June 14, 2005 7:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial
I had the same question... A portion of code is on there that I need to hack
openh323
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcel van
Kaam, Fonetica
Sent: Tuesday, June 14, 2005 10:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial
I have a whole bunch of remote devices connected to my Asterisk box,
including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only
rolled out recently and I am having a problem that is intermittent and
inconsistent.
It happens to some users but not other users on the same ISP. It
That interests me. Can you send me the informations about products and
suppliers?
Best regards,
--Hong
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quick and dirty example...
in zapata.conf
context=inbound-context
channel=(channels range here)
in extensions.conf
[inbound-context]
exten = _.,Dial(SIP/${desired_extension_here})
check the documentation about zapata.conf
best regards
On 6/14/05, Walid Azab [EMAIL PROTECTED] wrote:
Second day in a row...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Marcel van Kaam, Fonetica
Sent: Tuesday, June 14, 2005 8:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] VOIP-INFO
Sounds like to much use of the general context, remove etensions from
general that you require authentication for or use includes.
Post you extensions.conf for better help.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stojan Sljivic
What is the deal with voip-info.org, is it a commercial agreement or a
donation that has worn out its welcome? Needs more bandwidth or a faster
(load balanced) server!
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent:
To whoever owns this site. To help keep this up and running
I am willing to host it for free.
I run a regional ISP in the northeast.
Please contact me off list.
John Bittner
Simlab.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of
Damon Estep
Title: Info on ACD in Asterisk
Hello Sir,
I have few clarifications as we are planning to work with asterisk.
If you dont mind, please clarify the following:-
Q1. Do Asterisk support ACD functionality?
If Yes, can you give information on how to configure or work with ACD (and its
Hi,
Were using our legacy PBX as a channel bank with
asterisk sitting between the pbx and our telco provider spliced by a TE410P.
If it were a straight analog FXS card then wed use a
hook flash to break into asterisk for transfers etc, does anybody know what the
equivalent is for the
Hi,
Maybe a stun server somewhere on the internet can help you ? You could even
run your own on your remote asterisk server.
stun is useless if you have a simmetric nat
(like any iptables firewall, cisco etc etc)
matteo.
--
Matteo Brancaleoni
System Administrator
Tel +39.02.70633354
Sip
have you tryied to use 'socat' command?
best regards
On 6/14/05, Juan J. Sierralta P. [EMAIL PROTECTED] wrote:
Hi,
I just noted that :(
Anyway maybe is not so hard to modifiy the code to handle
multiple ports, I believe this would give a cleaner solution, only if
Prepaid can
Hi,
In [EMAIL PROTECTED] it's done like this:
exten = *65,1,Answer
exten = *65,2,AGI(festival-script.pl|Your phone number is ${CALLERIDNUM}.)
exten = *65,3,Hangup
(extensions.conf or an include file)
You need to have Festival installed
/HZ
- Original Message -
From: Ronald
On Tue, 14 Jun 2005, Amund Nygaard wrote:
We have around 50 phones in our company, and I am playing with the
thought to gradually go over to using sip services and ip-phones
internally. However at first I would liked the Asterisk just to sit
between the phone line and the Panaosnic, so I can
Where is the function? On source codes or any config file?
On 6/14/05, Denis Galvão - iSolve [EMAIL PROTECTED] wrote:
Hi all.
Could someone point me an example to use SIP_HEADER function!? I want
to read the To: and send this INVITE to an internal extension.
Is there anybody using this
Found the problem. I removed externip and localnet from sip.conf and everything is working now.Florian Overkamp [EMAIL PROTECTED] wrote:
Hi, -Original Message- SIP Phone (xten) - Linksys - Internet - PIX - Asterisk I can get 5060 working with no prob (PIX has a helper built in) but I
Hi,
What do I need to do to get asterisk to NOT pickup a Zap channel when
it rings? The channel in question is used for outbound calls only,
and all incoming calls are answered by an analog phone elsewhere in
the building that does not run through asterisk... so.. either make it
not answer.. or
I'm trying to clarify contexts and their uses. I do have a good
general understanding of them. My question is about undeclared
and non-existant contexts.
If I have a block somewhere (in sip.conf, for example), and it
has no context=thiscontext field, does it just automatically
use the default
Do a google search this has been asked and answered many times before (I
even think on Voip-info that there is an example of the one line of code
to change to make this happen).
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
On Jun 14, 2005, at 09:04, Matt wrote:
Hi,
What do I need to do to get asterisk to NOT pickup a Zap channel when
it rings? The channel in question is used for outbound calls only,
and all incoming calls are answered by an analog phone elsewhere in
the building that does not run through
Well voip-info seems to be down. However, a search on google does
bring up info about using Wait(x);... but as I stated.. using that
(even without an answer line) seems to make Asterisk take control of
the line!
On 6/14/05, Dean Collins [EMAIL PROTECTED] wrote:
Do a google search this has been
Hi,
I try one year ago to do the same trick :(
I was not able to use multiple ports on *.
I finish using ser on 5070 and asterisk on 5060 on combo box.
Some clients are register on ser using 5070 (with peers on asterisk
sip.conf) and few clients directly on asterisk using 5060.
Best regards,
Caller ID is still not working to certain areas. This problem was
confirmed by voipjet tech support in their last e-mail to me.
On 6/13/05, Matt [EMAIL PROTECTED] wrote:
I never noticed any problems.. so I can't comment :) hehe
On 6/11/05, Pedro [EMAIL PROTECTED] wrote:
Finally got a
In article [EMAIL PROTECTED],
Bryce Chidester [EMAIL PROTECTED] wrote:
On Jun 14, 2005, at 09:04, Matt wrote:
Hi,
What do I need to do to get asterisk to NOT pickup a Zap channel when
it rings? The channel in question is used for outbound calls only,
and all incoming calls are answered
On Jun 14, 2005, at 08:08, Damon Estep wrote:
What is the deal with voip-info.org, is it a commercial agreement or a
donation that has worn out its welcome? Needs more bandwidth or a
faster
(load balanced) server!
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Ok yes that should work.. but if it's not defined.. how do you
propose I make outgoing calls with it?
On 6/14/05, Bryce Chidester [EMAIL PROTECTED] wrote:
On Jun 14, 2005, at 09:04, Matt wrote:
Hi,
What do I need to do to get asterisk to NOT pickup a Zap channel when
it rings? The
In my situation, my zap line I wanted to skip answering was channel 1.
It was aimed at context [inpstn].
I simply made sure there were no Answer()s in my [inpstn] context... or
anything else that would answer the line. you could NoOp to put things
in the log, run any scripts you desire, but
I am having a problem using the Adtran 750 FXO quad card with a Groundstart
trunk line. I am able to receive calls on the trunk line, however dialing
out is not working. The Adtran does not seem to be doing the signaling. Has
anyone used the 750 FXO card in Groundstart mode? Any special
Did they say when it would be corrected?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedro
Sent: Tuesday, June 14, 2005 9:22 AM
To: Matt
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Anyone noticed
* Barton Fisher ([EMAIL PROTECTED]) ha scritto:
I found someone offering T1's for $290 a month + Loops or 3 Meg for
$561 a month + Loops. Is this a good deal?
when i read so high prices for bandwidth i wonder why i get 10Mbps
over optical fiber for 70Euros/month. and i'm not a business
There are ways of doing these types of changes without affecting the
users nearly as much. We do those kinds of things for our clients all
the time.
If its an issue of cost (monitarily or time) and there are others
willing to accept that cost, then I don't understand the reluctance to
let
On Tuesday 14 June 2005 12:04, Matt wrote:
What do I need to do to get asterisk to NOT pickup a Zap channel when
it rings? The channel in question is used for outbound calls only,
Just don't Answer(). Seems to work for me. I'm not sure why it doesn't for
you. Perhaps show us the relevant
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