Hi,
We are testing a SIP solution * + ser solution for a large implementation.
All the clients are nated.
When a client is dialing outside the domain (to a FWD sip account for
example) all is perfect ! ;-)
But ,when a call is done to a sip account, the client is ringing, then the
caller can hear
Le mardi 10 Mai 2005 08:01, Laurent Foulonneau a écrit :
Hi,
Hello,
We are testing a SIP solution * + ser solution for a large implementation.
All the clients are nated.
When a client is dialing outside the domain (to a FWD sip account for
example) all is perfect ! ;-)
But ,when a call is
Hi,
I want to use an Alacatel voip phones 4038 to connect to an Asterisk box,
does anyone know how I must configure this. If it is posible?
The phone is booting from an TFTP server and is looking for a file called
lanpbx.cfg.
Wiebe
___
Asterisk-Users
hello,
I am running asterisk on one linux PC and want to
talk through this server using Kphone installed on 2 different
PC's.These are the extra lines added to sip.conf and extensions.conf
respectively.
sip.conf
[jitha]type=friendhost=dynamicsecret=jithacontext=sipdtmfmode=inband
Hi
Understand your situation, had the same thing, bought 2 of 7690 phones
through e-bay, before i even had the * up and running.
They had CallManager installed and I wanted SIP, thanks to national
dealer the gave me a zip file for use with my two testphones.
Did struggle though to update from
Same..8a
On Mon, 2005-05-09 at 17:12, Eugenio De Vena wrote:
Which version of * and bristuff did you install, I had bristuff-0.2.0-RC8a
and now I am trying bristuff-0.2.0-RC8c
- Original Message -
From: Altus Snyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
On 5/9/05, Charles Wang [EMAIL PROTECTED] wrote:
Hi, ALL:
I use asterisk -r and sip debug to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.
I make a call from 1011 to on sip proxy,
sip proxy forwards this call to 0939749001.
And
Hi,
I just wanted to know if anyone managed to do call forwarding with
AT-320 phone from Atcom (not by reconfiguring Asterisk)?
For example, when I take my lunch break, I would like to forward all
calls to my mobile number.
Is it possible with Atcom AT320?
--
Tomek
Hi,
I plan to setup an * Server with a TE10P PRI card as E1 and a BN4S0 4 Port
BRI card in one box.
I tried it first with the bristuff drivers from Junghanns. The BRI card
worked fine alone, but as soon as I load the zaptel driver for the
TE110P the BRI card says, that the port is down.
So I
I have the same dtmf problem tried both inband and info but could not get
voicemail to work.
I have also been using voiptalk for the last year. The only isssue I have
with them is that during the day the service is fine but in the evening you
can get jitter problems and dropped calls.
Chris
Hi,
I use cvs stable version. I'd like to add database support on
extensions.conf. Other config files into db would be a big bonus. Looking
through the wiki, there are several ways... but not sure which one is the
best or widely used and supported...
ast_data and res_config seem to be the best
I have a simple dial plan to cascade calls when the first phone does not
answer:
exten = 100,1,Dial(SIP/1000,10,tr)
exten = 100,2,Dial(SIP/1000SIP/1001,10,tr)
exten = 100,3,Dial(SIP/1000SIP/1001SIP/1002,10,tr)
exten = 100,4,Voicemail(u100)
Problem is that the once the call goes onto the second
Either disable G729 or splash out $20 and get a couple of licenses, it
is hardly a King's ransom after all.
On Tue, 2005-05-10 at 01:04, Ren Mayorga wrote:
Hi, I have an asterisk server without any G729 licenses, and a couple of
BT-100 phones that actually works already with G729 passtrought
Kerry,
The only problem with the config you have outlined is that if you stick an
analogue phone onto the 3000 and get asterisk to route the trunked call back
to that phone then about 50% of calls are dropped with an error when you
answer the call. The only solution I have found is to configure
Hi,
-Original Message-
I tried it first with the bristuff drivers from Junghanns.
The BRI card
worked fine alone, but as soon as I load the zaptel driver for the
TE110P the BRI card says, that the port is down.
Is there any stable way, or as someone experience with this
two
Hello,
This is a little off-topic. I have an Ericsson FCT f251m, according to
the specs it supports call signalling through polarity reversals and
loop break, but it's currently disabled.
On my PSTN line, my TelCo does send polarity switchs to signal answer
and hangup (answeronpolarityswitch=yes
On Tue, 10 May 2005, Florian Overkamp wrote:
Hi,
-Original Message-
I tried it first with the bristuff drivers from Junghanns.
The BRI card
worked fine alone, but as soon as I load the zaptel driver for the
TE110P the BRI card says, that the port is down.
Is there any
Hi, a small question..
I'm using NTP to synch our phones with an ntp server, but it seems the
Cisco 7912G (with SIP image) does not handle daylight savings time very
well? Am I overlooking something or is this a known feature?
I'm using GMT+1 and minutes are correct but it doesn't respect DST.
Hi all,
i need some advices. In my office we have 7 PSTN lines from central
phone-office (one line - one number) and we plain to install an Asterisk
server as PBX. We need to have 15 PSTN devices (phones, fax, etc) in opur
office. I've seen FXS and FXO but i'm not sure: we need 7 FXO and 15
Hi
With asterisk is it possible to evenly distribute calls amongst agents,
even if a agent is free, but has previously received a call, is it
possible to send it to one who hasnt.
As for call stats, what is the best software for this, in terms of
allowing call snooping (like altigen), and also
Hi,
-Original Message-
Yes it can be done (at least with 'real' Junghanns QuadBRI
cards, I don't
know about the BN card, but I suppose it should work).
It is also possible with the Eicon DIVA Server cards (BRI,
4BRI and PRI).
The DIVA Server cards don't use Zaptel, they have
Have U tried to use DUNDI for that purpose ?
It's the best solution U could find.
http://www.voip-info.org/wiki-Asterisk+DUNDi+Call+Routing
On Mon, 2005-05-09 at 20:48, Vikram Rangnekar wrote:
+++ Kanuri, Seshu (Company IT) [09/05/05 11:25 -0400]:
Vikram,
Instead of trying to be
Le lundi 09 Mai 2005 23:56, Michele O-Zone Pinassi a écrit :
Hi all,
i need some advices. In my office we have 7 PSTN lines from central
phone-office (one line - one number) and we plain to install an Asterisk
server as PBX. We need to have 15 PSTN devices (phones, fax, etc) in opur
office.
Hullo :)
I'm using Debian's Asterisk 1.0.7 bristuffed (though I'm only using CAPI for
ISDN, and not HFC-S cards) and trying to transfer an incoming SIP call from
sipgate.co.uk to any other extension.
My phones are AT-320s (PA168S 1.43 firmware) whose documentation says to blind
transfer,
I'm using a cisco 1760 with a VIC2-4FXO card for my calls to PSTN.
If a user on a softphone hangs up first the PSTN port on the cisco is
released and new calls can be made on the same voice port. But when the
user on the PSTN side hangs up first the voice port on the cisco stays
open until the
Hi
Tks, and this can be done irrespective of the type of trunk used, i.e if
its over IP, since this will be a remotely hosted setup, with IP phones
only.
Iqbal
Tony Mountifield wrote:
In article [EMAIL PROTECTED], Iqbal [EMAIL PROTECTED] wrote:
With asterisk is it possible to evenly
It was also my problem...
Beware of generating ringtone (r, or rt string at the end of the call
command).
-b
- Original Message -
From: Torbjørn Lium [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 10, 2005 12:58 PM
Subject: [Asterisk-Users] BYE from Cisco
Hi!
I have an Asterisk Box with one E1. This is connected
with PSTN. My problem is that periodically the
Asterisk console shows the following message.
-- B-channel 0/1 succesfully restarted on span 1
-- B-channel 0/2 succesfully restarted on span 1
-- B-channel 0/3 succesfully restarted on span 1
Snippet from my extensions.conf
[pstn-ut]
exten = _0.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _0.,2,Congestion
exten = _0.,3,Hangup
Probably something is
a) horribly wrong
b) easy to fix
but how?
barney wrote:
It was also my problem...
Beware of generating ringtone (r, or rt string at the
no tira
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On Tue, 10 May 2005, Florian Overkamp wrote:
Hi,
-Original Message-
Yes it can be done (at least with 'real' Junghanns QuadBRI
cards, I don't
know about the BN card, but I suppose it should work).
It is also possible with the Eicon DIVA Server cards (BRI,
4BRI and
have you tried disabling DST on the phones, and just pointing them directly to the asterisk server? the asterisk box will make up for DST, the phones will just follow.
Chris
[EMAIL PROTECTED] 5/10/2005 5:40 AM
Hi, a small question..I'm using NTP to synch our phones with an ntp server, but it
In article [EMAIL PROTECTED],
Jairo Buendia [EMAIL PROTECTED] wrote:
Hi!
I have an Asterisk Box with one E1. This is connected
with PSTN. My problem is that periodically the
Asterisk console shows the following message.
-- B-channel 0/1 succesfully restarted on span 1
-- B-channel 0/2
Please!
Did someone make this happen?
I managed to install LDAPget application but I've been trying to make it work
for a few days.
I can connect to my database server, but every query returns -- LDAPget: Value
not found in directory. on the Asterisk console.
But it works fine when I run the
hello jairo,
Jairo Buendia wrote:
-- B-channel 0/1 succesfully restarted on span 1
unused b-channels are reset by asterisk every hour (default).
you can set the interval to another value in your
/etc/asterisk/zapata.conf
resetinterval=86400 ; e.g. reset every 24hours
or even longer.
i think
Hi,
I have a Asterisk with two Cisco ATA connected with
each other .
Cisco ATA1 is connected via satellite link
Cisco ATA2 is connected via leased line
The problem is
CISCO ATA2 can call CISCO ATA2 with no problem
But
CISCO ATA1 cannot call CISCO ATA2 .
Any
Hello,
I am a newbie in Asterisk IP PBX but I am very impressed by its
functionalities.
I have read that It can work over IP network and across the PSTN.
I am not very sure how it works over the PSTN..
In case if people have not yet the Internet or SDL access, I would like just to
know if it is
On Tuesday 10 May 2005 8:50 am, Brice Muangkhot wrote:
Hello,
I am a newbie in Asterisk IP PBX but I am very impressed by its
functionalities.
I have read that It can work over IP network and across the PSTN.
I am not very sure how it works over the PSTN..
In case if people have not yet the
Why don't you try a cisco user list?
On 5/10/05, Torbjørn Lium [EMAIL PROTECTED] wrote:
I'm using a cisco 1760 with a VIC2-4FXO card for my calls to PSTN.
If a user on a softphone hangs up first the PSTN port on the cisco is
released and new calls can be made on the same voice port. But when
Hello,
Im a newbie in connection several asterisk
servers with each others.
Ive got the following situation.
Ive got 9 asterisk servers (asterisk00 till
asterisk08).
When I call to asterisk08 then I want to redirect an
application which runs on asterisk00.
But how can I redirect
If one have an Internet/SDL access is it possible to have a dual access, in
such if one network fails, I can switch automatically to another?
Here below the network architecture I imagine.
(IP LAN1)-ASTERISK-(Digium FXO card)--(PSTN)--(Digium FXO
Card)-ASTERISK-(IP LAN 2)
I don't understand
For incoming calls on your POTS lines you need to have your telco
provide the busy rollover (sometimes called hunt). This is a common
feature available from the telco when you have several phone lines.
The idea is that if someone calls a phone number associated with a
line that is in use, rather
Franck Porcher wrote:
Le lundi 09 Mai 2005 23:56, Michele O-Zone Pinassi a écrit :
Hi all,
i need some advices. In my office we have 7 PSTN lines from central
phone-office (one line - one number) and we plain to install an Asterisk
server as PBX. We need to have 15 PSTN devices (phones, fax, etc)
Gavin Hamill wrote:
Hullo :)
I'm using Debian's Asterisk 1.0.7 bristuffed (though I'm only using CAPI for
ISDN, and not HFC-S cards) and trying to transfer an incoming SIP call from
sipgate.co.uk to any other extension.
My phones are AT-320s (PA168S 1.43 firmware) whose documentation says to
Rabii,
You should not double post if others have already answered
the original.
The previous answer pointed out that your description below
does not make sense.
Why would ATA2 call ATA2? It can call
itself? Do you mean "ATA2 can call
ATA1"??
Check your configs for the ATAs in sip.conf
No, asterisk is a software pbx, not an IP router.
-Original Message-
From: Brice Muangkhot [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 10, 2005 7:51 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Interconnecting two lans using
Asterisk over a PSTN
Hello,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I just noticed that the Skype API for linux seems to be available.
I've read before a number of posts where people were talking about
implementing a chan_skype with the skype API.
I wonder if there is any progress in that direction, and if anyone is
-Original Message-
From: Arjan Kroon [mailto:[EMAIL PROTECTED]
Sent: 10 May 2005 14:16
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Redirect to an application on other asterisk
server
Hello,
I'm a newbie in connection several asterisk servers with each others.
I've got
Set the SIP_CODEC var to ulaw on the first priority of VM calls.
Either disable G729 or splash out $20 and get a couple of licenses, it
is hardly a King's ransom after all.
On Tue, 2005-05-10 at 01:04, Ren Mayorga wrote:
Hi, I have an asterisk server without any G729 licenses, and a couple
Hi folks
!
I bought two sipura
841 phones. I used to have GN Netcom headset which I connect instead of
the handset. The problem is that I don't have any sound coming out the
headset and I can't speak neither !
I'am located in
France and I was wondering if the cabling in the sipura and
No, asterisk can't do that your phone line provider would have to
provide that service.
chawki hammoud wrote:
--- Tim Litwiller [EMAIL PROTECTED] wrote:
Your pstn land line can only handle 1 call at a time
To handle more at the same number you need a
rollover or busy redirect.
Then you could
Hi
We have been experimenting with different versions of Asterisk, and
found that the path to switch from one version to onother (say 1.0.7
to CVS for example) is somewhat unelegant, at least what we did:
make
make install
and replace everything in your system.
When you have problems with one
Hi all.
I've used LookupCIDName on incoming calls to tag them properly from the
internal DB -- that works well.
However, is it possible to use LookupCIDName or something else to tag
Outgoing calls as well? Specifically, in the logs? Be nice to have the
logs show the CID of the number I'm
Hi!
I have an Asterisk Box with one E1. This is
connected
with PSTN. My problem is that periodically the
Asterisk console shows the following message.
-- B-channel 0/1 succesfully restarted on span 1
-- B-channel 0/2 succesfully restarted on span 1
[..etc...]
I don't know if this
On Tuesday 10 May 2005 9:45 am, David Masure wrote:
Hi folks !
I bought two sipura 841 phones. I used to have GN Netcom headset which
I connect instead of the handset. The problem is that I don't have any
sound coming out the headset and I can't speak neither !
...
Orcan anyone
I'm trying to determine if this is a zaptel issue or a sangoma issue. When I
start our server, it gets to the point where it says starting wanrouter
(thats the sangoma drivers) and it just hangs there. Nothing happens. Can't
even use console.
I unplugged the 4 PRI lines and rebooted. This time
On May 10, 2005 09:37 am, Jay Milk wrote:
No, asterisk is a software pbx, not an IP router.
Don't be so closed-minded; someone can create a IPoV channel. :-)
-A.
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Thanks It's work with the SIP_CODEC var, anyway I've already buy a two
G729 Licenses, for testing purposes
On Tue, 2005-05-10 at 08:47 -0500, Don Dawson wrote:
Set the SIP_CODEC var to ulaw on the first priority of VM calls.
Either disable G729 or splash out $20 and get a couple of
Hi,
I have some Cisco IP Phone 7912 registered to
Asterisk. I wanted to know if it is possible to enter the IP address of the IP
phone situated in the same LAN you want to call instead of its phone number.
Is it possible with other IP Phones?
Thank you for your help,
Marlene
Interesting...I will test this myself tonight. Did you make sure you are
using the latest firmware?
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton
Sent: Tuesday, May 10, 2005 1:54 AM
To: Asterisk Users Mailing List -
On May 10, 2005 09:56 am, Jairo Buendia wrote:
why are the idle channels restarted?, the PSTN could
think that my box has some problem. Perhaps, other
equipments also make the same but I don't know
anything about it. I used Access Server of Cisco and I
think that the channels weren't
I am new to Asterisk
and Linux. I need help configuring my system. I have 5 Cisco 7940g IP phones, a
Cisco ATA186, and a Zyxel 2000W Wi-Fi IP Phone. I have 3 SIP VOIP Lines,
currently running on the ATA and 2000W. I have the server installed, but don't
know where to even begin to setup
I have two phones, one does not need stun, the other one needs.
All settings are identically, except the number/password and said above
stun - not stun
I use codec in the order:
g729
g711u
g711a
Any ideas, why the user can hear me, but I cannot hear him (stun) while
the other user without stun
I uses to have this when I enabled stun and did not need it
On Tue, 2005-05-10 at 16:55, Ronald Wiplinger wrote:
I have two phones, one does not need stun, the other one needs.
All settings are identically, except the number/password and said above
stun - not stun
I use codec in the
Hello * Users
Did somebody get managed to get AreskiCC work under mysql.
If so is there anywhere to find the database structure for mysql.
Thanks
Sjaak
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I use ASTCC (agi) and it jumps out into a local contest, where I choose
the real trunk I will dial.
I would like to use hear LCR than, but I am not sure if I can have two
agis at once
bye
Ronald
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Adam,
You should really look at [EMAIL PROTECTED].
http://asteriskathome.sourceforge.net
It has AMP and a ton of other features that will be useful
for a new user.
Cheers,
Wiley
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
CollardSent: Tuesday, May 10, 2005
It's on the way... I will provide it soon!
Areski
On Tue, 2005-05-10 at 17:08, Sjaak Nabuurs wrote:
Hello * Users
Did somebody get managed to get AreskiCC work under mysql.
If so is there anywhere to find the database structure for mysql.
Thanks
Sjaak
You can use Ethereal to see what your phone (stun) is
sending. Of this way you can see the RTP ports and IP
public that your phones are going to use. You can see
that information in INVITE and OK packets.
For other hand, If you use one router with symmetrical
NAT then Stun won't work
Hi guys,
I setup an IPPBX and some IP Phones to register
to it. IPPBX has 6 register= [EMAIL PROTECTED] sip
accounts. When these sip accounts register timeout due
to the WAN network problem all the extensions got
logon failed.
It seems IPPBX register to its own sip proxy server
first.
If it
Versiom 0.115 - 10. may 2005
* This version will encrypt your configuration file, so logins and passwords
are not easily readable.
* Lots of bug fixes.
Download: http://ipswitchboard.thorben.dk
Regards
Thorben
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Nevermind, I have solved the problem.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
Behalf Of [EMAIL PROTECTED]
Sent: 10 mai 2005 10:33
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] MGCP : chan_mgcp.c:1509 find_subchannel
Importance: High
When I
If your doing that many mins a month you could probably go to one of the
bigger cariers yourself
Level3, global crossing, broadwing, ATT, williams to just name a few
- Original Message -
From: VOIP Consultant [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Touché
I'm still waiting for chan_bt_gsm :)
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 10, 2005 9:16 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Interconnecting two lans using
Asterisk over aPSTN
On May 10,
Anybody knows what is and how to use the ISDNguard daemon included in
new bristuff packages ?
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hello
i want to get extension from ivr
its not working
exten =6000,1,ResponseTimeout(5)
exten =6000,2,Background(enterexten)
exten =6000,3,SetVar(myexten=${digitstack})
exten =6000,4,Wait(5)
exten =6000,5,Goto(default,myexten,1)
Kamran
__
Hi,
I'm using a SPA 3000 as FXO and FXS termination connected to my * box.
I'm using the caller ID prefix trick explained here:
http://www.voip-info.org/wiki-Sipura+3000.
All seems to work really fine, there is only a problem when I hangup my
IP phone after a conversation and the other party
Hello,
I have been trying to setup an Asterisk server behind a
Cisco 837 router (IOS ver 12.3(2)), but with no joy. I then tried
[EMAIL PROTECTED] as this is a far simplified method, but I am still unable to get
anything working, so I am almost sure that it is my router config.
Has
I will be out of the office starting 05/10/2005 and will not return until
05/28/2005.
I will respond to your message when I return.
--
The information contained in this communication is intended solely for the
use of the individual or
There are settings in the advanced config to detect hangups or dropped lines
as well as timeout and sensitivity adjustments for them.
---
Hi,
I'm using a SPA 3000 as FXO and FXS termination connected to my * box.
I'm using the caller ID prefix trick
all you have to do is nat port 5060 udp to the outbound interface.
Chris
[EMAIL PROTECTED] 5/10/2005 12:38 PM
Hello,
I have been trying to setup an Asterisk server behind a Cisco 837 router (IOS ver 12.3(2)), but with no joy. I then tried [EMAIL PROTECTED] as this is a far simplified
Whew... What a relief.
I know the list was worried about why we could not get a hold of Manoj
Shetty
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Manoj
Shetty
Sent: Monday, May 09, 2005 12:24 PM
To: asterisk-users
Subject: [Asterisk-Users]
We are using DISA with local SIP users. The user enters in a 2 digit
code then they get a dialtone and the phone dials out. The problem is
that the calls waits 10 seconds after the outgoing number is dialed, no
matter what I put for the timeout values. Anyone else using DISA that
has run into
Hello,
I have been trying to setup
an Asterisk server behind a Cisco 837 router (IOS ver 12.3(2)), but with no
joy. I then tried [EMAIL PROTECTED] as this is a far simplified method, but I am
still unable to get anything working, so I am almost sure that it is my router
config. Has anyone
We're going to be switching our 800 service soon and I just want to make
sure I have the right idea about what we'll be doing. Currently it runs
through an Adtran and the lines are just divided into groups in
zapata.conf, but our new service will be pure VoIP. I'm almost positive
it will be
We are using DISA with local SIP users. The user enters in a 2 digit
code then they get a dialtone and the phone dials out. The problem is
that the calls waits 10 seconds after the outgoing number is dialed, no
matter what I put for the timeout values. Anyone else using DISA that
has run into
On Tue, 10 May 2005, Chris Stinson wrote:
We are using DISA with local SIP users. The user enters in a 2 digit code then
they get a dialtone and the phone dials out. The problem is that the calls
waits 10 seconds after the outgoing number is dialed, no matter what I put for
the timeout values.
I bought two sipura 841 phones. I used to have GN Netcom headset
which I connect instead of the handset. The problem is that I don't
have any sound coming out the headset and I can't speak neither !
I'am located in France and I was wondering if the cabling in the
sipura and in the headset
George Pajari wrote:
I bought two sipura 841 phones. I used to have GN Netcom headset
which I connect instead of the handset. The problem is that I don't
have any sound coming out the headset and I can't speak neither !
I'am located in France and I was wondering if the cabling in the
hehe
[EMAIL PROTECTED] 5/10/2005 12:50 PM
Whew... What a relief.I know the list was worried about why we could not get a hold of ManojShettyW-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of ManojShettySent: Monday, May 09, 2005 12:24 PMTo:
What makes you think I'm not trying a cisco user list?
At least it's worth a try to post the question here also.
C F wrote:
Why don't you try a cisco user list?
On 5/10/05, Torbjørn Lium [EMAIL PROTECTED] wrote:
I'm using a cisco 1760 with a VIC2-4FXO card for my calls to PSTN.
If a user on a
And I find that my cisco will send BYE after 30 seconds after PSTN hangup.
On 5/11/05, Charles Wang [EMAIL PROTECTED] wrote:
yes, my cisco trunking gateway has also this problem.
On 5/11/05, Torbjørn Lium [EMAIL PROTECTED] wrote:
What makes you think I'm not trying a cisco user list?
At
Hi,
I am currently trying out the [EMAIL PROTECTED] (version 1)
release of Asterisk, and I want to configure it as follows:
Calls from regular telephony network (PSTN) come in through
my VoIP provider over SIP and outgoing calls to the PSTN should be routed
through the ViOP
yes, my cisco trunking gateway has also this problem.
On 5/11/05, Torbjørn Lium [EMAIL PROTECTED] wrote:
What makes you think I'm not trying a cisco user list?
At least it's worth a try to post the question here also.
C F wrote:
Why don't you try a cisco user list?
On 5/10/05, Torbjørn
We have been running into problems here, we have 2 PRI's when they
fillup, All channels in use, and we dial more calls asterisk becomes
unstable and crashes alot.
We are currently on: Asterisk CVS-v1-0-10/26/04-09:35:31 built by
[EMAIL PROTECTED] on a i686 running Linux
I know I need to
Handset wiring IS non-standard...
I suspect that only applies to headsets using RJ21 or whatever the
little RJ connector is called and not to 2.5mm headsets.
Agreed. Which is why I was referring to HANDset wiring (the original
author stated he was connecting his headset to the phone in place
On Tuesday 10 May 2005 9:45 am, David Masure wrote:
Hi folks !
I bought two sipura 841 phones. I used to have GN Netcom headset
which
I connect instead of the handset. The problem is that I don't have
any
sound coming out the headset and I can't speak neither !
...
Orcan anyone
I just installed a TE410P on a Debian Sarge system running kernel
2.6.11-1-686-smp. Zaptel and Asterisk seem to be working fine.
However, I have a couple of problems with the TE410P and Zaptel.
First, the TE410P is showing me red alarms on 2 of the 4 T1s. This
board (the TE410P) was just
[EMAIL PROTECTED] schrieb:
Nevermind, I have solved the problem.
It would be nice to know how! I have the same message, and i want to get
rid of it, too.
Best regards
Thomas
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Kyle Hagan wrote:
We have been running into problems here, we have 2 PRI's when they
fillup, All channels in use, and we dial more calls asterisk becomes
unstable and crashes alot.
We are currently on: Asterisk CVS-v1-0-10/26/04-09:35:31 built by
[EMAIL PROTECTED] on a i686 running Linux
I
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