On Mon, 21 Jun 2004, Greg Blakely wrote:
Can anyone help? Here is my extensions.conf, and the error message I
get.
Once you execute Congestion, everything stops.
You need something like
exten = _NXX,1,Dial(SIP/)
exten = _NXX,2,Dial(ZAP/26/...)
exten = _NXX,3,Congestion
No Caller ID comes from the FXO line ( The caller id is on and is
working with a standard phone)
in zapata.conf everything looks fine
usecallerid=yes
hidecallerid=no
When the call comes in there are some warnings in Asterisk Console
-- Starting simple switch on 'Zap/4-1'
Jun
Damian Minkov [EMAIL PROTECTED] wrote:
No Caller ID comes from the FXO line ( The caller id is on and is
working with a standard phone)
in zapata.conf everything looks fine
usecallerid=yes
hidecallerid=no
When the call comes in there are some warnings in Asterisk Console
Hi,
I've read your question also in the OpenSS7 mailinglist,
and I hoped, Brian F.G. Bidulock would answer. Brian is the
expert concerning SS7 support of those cards.
If I have understood right, Brian developed for his company
(OpenSS7.com) a special firmware for the digium T400P and
sells it now
Thanks for your reply Miklos.
I´m afraid I´m confronted with the same problem.
Now my optipoint is registering to asterisk.
(I had to configure the system type to SERVER , the Registrar Address Server Address
to my asterisks-ip-address.)
Now the optipoints are telling me:
No Server... , that´s
Hi Angel,
it's looking not so bad, similar messages i have on my * server and, in
spite of it, * is working almost ideally.
Send us more info about problems.
Konrad
Angel Diaz wrote:
*CLI Jun 22 20:37:55 NOTICE[1133718080]: chan_zap.c:4881
handle_init_event: Alarm cleared on channel 1
Jun 22
Hi,
-Original Message-
VoiceXML support would be great, but I know of any active work on it.
openVXI seems to have spri=ung to life again recently, after years of
languishing. Perhaps it would form a sound base to get
VoiceXML up and
running in a reasonable time.
Do any of
Florian Overkamp wrote:
Hi,
-Original Message-
VoiceXML support would be great, but I know of any active work on it.
openVXI seems to have spri=ung to life again recently, after years of
languishing. Perhaps it would form a sound base to get
VoiceXML up and
running in a reasonable
Hi!
Yes we have many kinds of phones hwere in the show room, snom, polycom,
cisco, grandstream, ipdialog, intracom, d-link, symbol all of them works
with asterisk with some testing and with some issues ...but works.
The optipoint is the only one that i´m really can´t make work till now.
In
I
dispose of a asterisk server with a quad pri card in it and a asterisk server
with a single pri card.
Could
I add a second single pri card to the second server ? It is for multiplexing
purposes.
Regards,
Jan
I have problem compiling it
chan_zap.c: In function `zt_get_history':
chan_zap.c:768: storage size of `hist' isn't known
chan_zap.c:771: `ZT_GET_HISTORY' undeclared (first use in this function)
chan_zap.c:771: (Each undeclared identifier is reported
Hi!
The Problem with the password is strange. I have the same password problems but my
phone is registering to port 5060.
But I think in our case the problem isn´t caused by the ports. it must be something
different.
I nearly tried everything to get the optipoint working.
I will keep trying,
Hi,
I'm thinking of purchasing a TDM400P card and was wondering if anyone has experienced any echo issues with phones off these cards connecting to the PSTN via the X100P cards?
I have had my fingers burnt with a Voip phone X100P.
Cheers,
Taff.
ALL-NEW
Yahoo! Messenger - so many
Ok so here's one
Has anyone managed to get the 'i' extension to work.
I have included within each context the following
exten = i,1,Goto(wrong-number,s,1)
then in
[wrong-number]
exten = s,1,GotoIf($[${EXTEN:0:2} = 43}]?10:2)
exten = s,2,GotoIf($[${EXTEN:0:2} = 62}]?11:99)
exten =
At 13:34 22/06/2004 +0300, you wrote:
I have problem compiling it
chan_zap.c: In function `zt_get_history':
chan_zap.c:768: storage size of `hist' isn't known
chan_zap.c:771: `ZT_GET_HISTORY' undeclared (first use in this function)
chan_zap.c:771: (Each
That message is created by the Voicemail application. Check your
extensions.conf and see what your action is for when the call can not
be connected.
For example, a correct dialplan for a SIP extension would read:
exten = _200Z,1,Dial(SIP/${EXTEN},20)
exten = _200Z,2,Voicemail(u${EXTEN})
exten =
Damian Minkov [EMAIL PROTECTED] wrote:
I have problem compiling it
chan_zap.c: In function `zt_get_history':
chan_zap.c:768: storage size of `hist' isn't known
chan_zap.c:771: `ZT_GET_HISTORY' undeclared (first use in this function)
Current Config:
Asterisk CVS-04/01/03-05:52:52
3 SNOM 200's
1 SNOM 100
The Extension State Stuff exists in chan_sip !
Sorry i have some Changes in the Asterisk Extension
State Source because this i have never compiled a
new Version.
In the current Asterisk CVS Version the SNOM 200
taf taffey [EMAIL PROTECTED] wrote:
(Article auto-converted from unnecessary HTML to nice plain text.)
I'm thinking of purchasing a TDM400P card and was wondering if anyone has
experienced any echo issues with phones off these cards connecting to the
PSTN via the X100P cards?
I have had
I've compiled and run it but no effect.
Then i noticed that there is warning when i run asterisk
Jun 22 14:37:47 WARNING[16384]: chan_zap.c:8762 setup_zap: Ignoring
ukcallerid
Kevin Walsh wrote:
Damian Minkov [EMAIL PROTECTED] wrote:
I have problem compiling it
For example, a correct dialplan for a SIP extension would read:
exten = _200Z,1,Dial(SIP/${EXTEN},20)
exten = _200Z,2,Voicemail(u${EXTEN})
exten = _200Z,102,Voicemail(b${EXTEN})
exten = _200Z,103,Hangup
Hi All... I'm a newbie, just busy getting to grips with asterisk.
I've set up the
Hi,
Just started to get this error after updating to the latest CVS. Asterisk dies if it
can't create a channel - not so good.
-- Executing SetCallerID(SIP/750-2550, 39660426) in new stack
-- Executing Dial(SIP/750-2550, CAPI/39660426:22179808) in new stack
Jun 22 13:52:05
At 14:39 22/06/2004 +0300, you wrote:
I've compiled and run it but no effect.
Then i noticed that there is warning when i run asterisk
Jun 22 14:37:47 WARNING[16384]: chan_zap.c:8762 setup_zap: Ignoring ukcallerid
Make sure you have the correct switch in zapata.conf
callerid=uk
Regards
Jason
Hi Guys,
Same problem here with latest CVS.
-cf
Hi,
Just started to get this error after updating to the latest CVS. Asterisk
dies if it can't create a channel - not so good.
-- Executing SetCallerID(SIP/750-2550, 39660426) in new stack
-- Executing Dial(SIP/750-2550,
We have an Asterisk server that speaks IAX2 to Magrathea to get to the
PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s
all configured for SIP with silence-suppression disabled. Everything
is configured to use a-law encoding. The version is:
sip*CLI show version
Asterisk
Not working :((
again this warning
Jun 22 15:21:58 NOTICE[213006]: callerid.c:281 callerid_feed: Unknown IE 17
Jun 22 15:21:58 NOTICE[213006]: callerid.c:281 callerid_feed: Unknown IE 48
here is my zapata.conf
[channels]
language=en
callerid=uk
usecallerid=yes
hidecallerid=no
callwaiting=yes
Hi,
I can't find anywhere on the Asterisk web the license terms for
commercial use of Asterisk software. Do I have to pay something
(and how much) if I want to use the Asterisk in our IP PBX solutions?
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:
Miroslav Nachev wrote:
Hi,
I can't find anywhere on the Asterisk web the license terms for
commercial use of Asterisk software. Do I have to pay something
(and how much) if I want to use the Asterisk in our IP PBX solutions?
From a README:
* LICENSING
Asterisk is distributed under GNU
Hi,
I'm thinking of purchasing a TDM400P card and was wondering if anyone has
experienced any echo issues with phones off these cards connecting to the
PSTN via the X100P cards?
I have had my fingers burnt with a Voip phone X100P.
I replaced two x100p's with a tdm04b 4-port fxo card,
Brian K. West wrote:
http://www.theregister.co.uk/2004/06/22/sip_versus_skype/
Thought you might like this link Brian :)
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com -
[EMAIL PROTECTED] wrote:
Hi,
I can't find anywhere on the Asterisk web the license terms for
commercial use of Asterisk software. Do I have to pay
something (and how much) if I want to use the Asterisk in our
IP PBX solutions?
GNU General Public License. Check out the LICENSE
Hi Keith,
Hi All... I'm a newbie, just busy getting to grips with asterisk.
I've set up the following, but it causes a segfault when I call somebody who
is offline:
exten = _[123456789],1,NoOp(.call for .${EXTEN})
exten = _[123456789],2,Dial(SIP/${EXTEN},60,tr)
exten =
Are you running Redhat or Fedora? If so, read this thread for a solution:
http://lists.digium.com/pipermail/asterisk-users/2004-January/031953.html
Nope, SUSE SLES 8
regards,
Keith
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi,
Does anybody know the place to download the firmware for swissvoice ip10s
I have several phones with application IP10 H3 v1.0.0 (Build 1)
I'm looking for newer H.323 and also MGCP firmwares
Are the SIP firmware available, according to web its targeted to Q1 2004,
but we have week left in
I'm trying to get two * boxes to talk no matter what variation I try
I get No Authority Found and connection refused from 192.168.1.5
I've googled, I've site searched to no avail.
Here is the server a configs (192.168.1.5):
iax.conf
[general]
port=5036
bandwidth=low
disallow=all
Hi!
callerid=br exists?
miklos
- Original Message -
From: Jason Williams [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 22, 2004 9:06 AM
Subject: Re: [Asterisk-Users] No Caller ID from FXO Problem
At 14:39 22/06/2004 +0300, you wrote:
I've compiled and run it but no
Miroslav Nachev [EMAIL PROTECTED] wrote:
I can't find anywhere on the Asterisk web the license terms for
commercial use of Asterisk software. Do I have to pay something
(and how much) if I want to use the Asterisk in our IP PBX solutions?
You don't need a commercial license for that. The
Nicholas Bachmann [EMAIL PROTECTED] wrote:
Miroslav Nachev wrote:
I can't find anywhere on the Asterisk web the license terms for
commercial use of Asterisk software. Do I have to pay something
(and how much) if I want to use the Asterisk in our IP PBX solutions?
From a README:
*
I'm not sure how I can handle timeouts and invalid extensions for my
Zaptel channels...
Their default context is [internal], and in internal I have defined
extensions i and t to handle timeouts and invalid extensions.
However, the default for the Zaptel channels is immediate=no, so the
Damian Minkov [EMAIL PROTECTED] wrote:
I've compiled and run it but no effect.
Also make sure that you installed the Zaptel drivers (make install)
before compiling Asterisk.
Then i noticed that there is warning when i run asterisk
Jun 22 14:37:47 WARNING[16384]: chan_zap.c:8762 setup_zap:
I was just trying to solve this one myself. I found this method worked
for me. I'm still calling this Method 1 in my document because I don't
fully understand the switch and the register versions and pros/cons
to implementation of each. But this one does work.
Method 1
Receiving Server
Iax.conf
I temporarily have a second T100p card in my asterisk box that will have a
PRI to a different telco than the first, with a different switch type to
make things even more interesting. I'm not seeing any config examples for
this situtation.. anybody else doing this or have recommendations?
-Mike
Keith Waters wrote:
Are you running Redhat or Fedora? If so, read this thread for a solution:
http://lists.digium.com/pipermail/asterisk-users/2004-January/031953.html
Nope, SUSE SLES 8
There are other users running the latest CVS-HEAD reporting that problem
(asterisk segfaults when unable to
I am configuring call forwarding in our * setup, but I am having
trouble triggering the correct voicemail call.
When I have an extension, e.g. 201, forwarded to another, e.g. 202, my
macro will call:
Dial(Local/[EMAIL PROTECTED]/n,30)
and once into the internal context, the macro is called
Tony Nichols [EMAIL PROTECTED] wrote:
I'm trying to get two * boxes to talk no matter what variation I try
I get No Authority Found and connection refused from 192.168.1.5
[snip]
Server b config (192.168.2.2):
[pbx]
type=peer
host=dynamic
trunk=yes
secret=test
qualify=yes
Use
At 10:30 22/06/2004 -0300, you wrote:
Hi!
callerid=br exists?
miklos
Not unless you write the code
Jason
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
David Cook [EMAIL PROTECTED] wrote:
[mycontext]
exten =
_5XXX,1,Dial(IAX2/REC_SERVER:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = _5XXX,2,Hangup exten = _5XXX,102,Hangup
You really don't want your username and password to appear (in plain
text) in your logs.
Put the sensitive details in
Well it is working good here in my office.
Only I have problem when calling to SIP phone and this SIP phone is not
located in our LAN network.
But when I am calling to the same sip ip hard phone from SIP phone
connection is good.
Only have problem calling from Adtran, very strange.
B.
-
So you're saying that the following would be the same?
iax.conf
[YOUR_REC_SERVER]
secret=mysecret
host=my.receiving.server.ca
context=local
extensions.conf
exten = _5XXX,1,Dial(IAX2/YOUR_REC_SERVER/${EXTEN})
If so, what about the type=peer/user/friend thing? I did read the docs
but maybe I'm
On Tuesday 22 June 2004 10:08, Kevin Walsh wrote:
You really don't want your username and password to appear (in plain
text) in your logs.
Put the sensitive details in iax.conf instead of extensions.conf.
As well as being more secure, it'll make your Dial() string shorter,
and will mean that
I'm not really being too lucky in the last days. After trying to compile cdr_mysql
with no success, I am switching to cdr_odbc. I have installed unixODBC, iODBC and
MyODBC correctly, I am even able to make queries with isql. But when trying to make
in the cdr directory of the latest CVS, that's
Hi,
I'm attempting to setup a server with 2 TDM0xB cards. The first has 4 FXO
modules and the second has 2 FXO modules. When I run ztcfg the output is as
expected showing 6 cards configured. I then setup 6 channels in
zapata.conf. When I startup asterisk I get an error trying to start the
Can someone plese clarify the correct way to use patlooptest?
I am trying to test my circuit and I have a loopback at the other end. When I run
./patlooptest /dev/zap/1 30 I get this:
(Error 1): Unexpected result, 255 != 0, 1 bytes since last error.
(Error 2): Unexpected result, 255 != 0, 1
I was finally able to compile asterisk with cdr_odbc.so. But now for some reason I get
that error:
*CLI load cdr_odbc.so
Jun 22 16:38:53 WARNING[-1084309376]: loader.c:240 ast_load_resource: libiodbc.so.2:
cannot open shared object file: No such file or directory
Unable to load module
I've got a number (10) Cisco 7960's connected to my network. All the phones
work perfectly except one.
The asterisk console keeps throwing up:
Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer
'4001' is now UNREACHABLE!
Jun 22 15:39:27 NOTICE[-1147470928]:
On Mon, 2004-06-21 at 23:26, Simon Brown wrote:
When I dial a SIP phone which is specified in the sip.conf, but the phone is
not connected, Asterisk gives the message The user at Extension XXX is on
the phone
Shouldn't the message be the unavailable message?
Is there something wrong with
On Tue, 22 Jun 2004, Tony Nichols wrote:
I'm trying to get two * boxes to talk no matter what variation I try
I get No Authority Found and connection refused from 192.168.1.5
I've googled, I've site searched to no avail.
I think you need to match a peer at one end to a user at the
David Cook [EMAIL PROTECTED] wrote:
So you're saying that the following would be the same?
iax.conf
[YOUR_REC_SERVER]
secret=mysecret
host=my.receiving.server.ca
context=local
extensions.conf
exten = _5XXX,1,Dial(IAX2/YOUR_REC_SERVER/${EXTEN})
If so, what about the
Manuel Wenger wrote:
I was finally able to compile asterisk with cdr_odbc.so. But now for some reason I get
that error:
*CLI load cdr_odbc.so
Jun 22 16:38:53 WARNING[-1084309376]: loader.c:240 ast_load_resource: libiodbc.so.2:
cannot open shared object file: No such file or directory
Unable to
Perfect! Thanks for the clarification. That's what my brain needed - on
both points.
dbc.
Quoting Kevin Walsh [EMAIL PROTECTED]:
If that's on your outgoing side then you'll also need type = peer
in there. The incoming side would have type = user.
Outgoing = peer, incoming = user. Friend
Thank you. That did the trick.
Once you execute Congestion, everything stops.
You need something like
exten = _NXX,1,Dial(SIP/)
exten = _NXX,2,Dial(ZAP/26/...)
exten = _NXX,3,Congestion
Steve
___
Asterisk-Users mailing
hello,
i 've edit all files as described in
http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO ,
but here is the problem...
capiinit give me no error at all,
but capiinfo - pc freeze!
here is proc status:
less /proc/capi/driver fcpci1 0.4
f2pci
Manuel Wenger [EMAIL PROTECTED] wrote:
*CLI load cdr_odbc.so
Jun 22 16:38:53 WARNING[-1084309376]: loader.c:240 ast_load_resource:
libiodbc.so.2: cannot open shared object file: No such file or directory
Unable to load module cdr_odbc.so
But the file is there...
# ls -lag
On Tue, 2004-06-22 at 10:20, David Cook wrote:
So you're saying that the following would be the same?
iax.conf
[YOUR_REC_SERVER]
secret=mysecret
host=my.receiving.server.ca
context=local
extensions.conf
exten = _5XXX,1,Dial(IAX2/YOUR_REC_SERVER/${EXTEN})
If so, what about the
Sorry for the stupid question:
What's the purpose of defining a peer as trunk in iax.conf ?
The question is also valid generally speaking (for other channel
types), for instance: why define a Zap group as trunk in
extension.conf ?
Tnx for any help !
--
Best regards,
Alessio
Hi all
I have installed dhe modifyed prepaid application and pouplated the
database to. the authentication works fine also the read credit but I cant
send calls, I think there is some missconfiguration with providers table.
I get the following logs from postgresql:
LOG: statement: SELECT * FROM
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rich Adamson wrote:
|Hi,
|I'm thinking of purchasing a TDM400P card and was wondering if anyone has
|experienced any echo issues with phones off these cards connecting to the
|PSTN via the X100P cards?
|
|I have had my fingers burnt with a Voip phone
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Kevin Walsh wrote:
~ You should experiment with your configuration settings for the FXO
| card. Check your gain values etc. If you don't have any luck with
| that then try changing your zconfig.h to switch on CONFIG_ZAPTEL_MMX.
| You can also play
On Jun 22, 2004, at 7:22 AM, Manuel Wenger wrote:
I'm not really being too lucky in the last days. After trying to
compile cdr_mysql with no success, I am switching to cdr_odbc. I have
installed unixODBC, iODBC and MyODBC correctly, I am even able to make
queries with isql. But when trying to
How about starting your macro with SetVar(DialedExten=${MACRO_EXTEN})
and then going into voicemail with VoiceMail(u${DialedExten})
-Original Message-
From: Michael George [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 22, 2004 9:00 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
Hello,
I've managed to build in the last number repeat outlined at
http://www.voip-info.org/wiki-Asterisk+last+number+repeat to call back
the last person _I_ called from a particular phone, and now I'd like to
try to do something similar for the common *69 -- call back the last
number that called
On Jun 22, 2004, at 12:30 PM, Jay Milk wrote:
How about starting your macro with SetVar(DialedExten=${MACRO_EXTEN})
and then going into voicemail with VoiceMail(u${DialedExten})
Well, I tried setting variables and saving them through the context
change, but the variable (DialedExten in your
How about just before you dial an extension you do a:
DBput(LAST/${EXTEN}=${CALLERID_NUMBER})
and then *69 does a:
DBget(dialNum=LAST/${CALLERID_NUMBER})
(from your extension) and you can dial it? I've never tried it, but
that might be a simple starting point...
On Jun 22, 2004,
Nice workaround. I was under the impression that SetVar works
(according to the wiki) within the scope of a channel, but apparantly it
gets lost once you move context. Bummer.
-Original Message-
From: Michael George [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 22, 2004 12:03 PM
Got it!
How about just before you dial an extension you do a:
DBput(LAST/${EXTEN}=${CALLERID_NUMBER})
and then *69 does a:
DBget(dialNum=LAST/${CALLERID_NUMBER})
(from your extension) and you can dial it?
Here's what I've done--
In my extension macro:
Hello
I recently setup PBX at one facility with One 4 Digium FXS card and one port
Digium T1 card connected to CA Channel bank for Analog lines. After running for
couple of hours, caller stop getting Main Menu from some 1 or 2 lines and Phone
keep on ringing for caller. Looks like Asterisk stop
On Jun 22, 2004, at 1:34 PM, Jay Milk wrote:
Nice workaround. I was under the impression that SetVar works
(according to the wiki) within the scope of a channel, but apparantly
it
gets lost once you move context. Bummer.
No, they will survive changes of context, but doing a Dial(Local/...)
is
This:
; Return last call
exten = *69,1,DBget(temp=LastCIDNum/${CALLERIDNUM}) ; read db value
LastCIDNum for this CALLERIDNUM (i.e. the extension making the call)
exten = *69,2,GotoIf($[${temp:0:3} = 208]?3:4)
; if it's area
code 208 (my local area), go to priority 3,
I am getting the following error as of today after updating both
asterisk and asterisk-addons. These are both under /usr/src.
...
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o
cdr_addon_mysql.o cdr_addon_mysql.c
cdr_addon_mysql.c:50: warning: parameter names (without types)
Hi,
I have setup agent 1000 in agents.conf, and the queue cytelcs in
queues.conf, and my extensions.conf file looks like
exten = 28,1,AgentCallbackLogin
exten = 9665,1,AddQueueMember(cytelcs|SIP/2815691212)
exten = 9665,2,Playback(agent-loginok)
exten = 9665,3,Hangup
I am able to log into the
I have fired off Asterisk Extensions conf with 2 extensions i.e 2000 and 2001
and made some test calls.
I forgot to set a time out. The calls between these two were partially
successful.
I have modified the original sip.conf and extension.conf file instead of
writing mine.
After writing my
Jeremy Jones wrote:
exten=s,1,DBput(LastCIDNum/${DNID}=${CALLERIDNUM}); grab
CALLERIDNUM store it for the dialed number as LastCIDNum exten=etc...
exten = *69,1,DBget(temp=LastCIDNum/${CALLERIDNUM})
exten = *69,2,GotoIf($[${temp:0:3} = 208]?3:4)
exten = *69,3,Macro(dialout,${temp:3})
have you install mysql-devel?
I am getting the following error as of today after updating both
asterisk and asterisk-addons. These are both under /usr/src.
...
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o
cdr_addon_mysql.o cdr_addon_mysql.c
cdr_addon_mysql.c:50: warning:
I am in process of setting up an Asterix PBX. I
expect to have many outbound long distance calls made simultaneously. I just
contacted voicepulse regarding their Connect service. They support 4 outbound
calls at a time.
My question is - How does one configure Asterix to
use:
1.
I can't park a call that comes from a queue. The situation is:
1. A call gets into a queue.
2. It is answered by an agent.
3. An agent wants to transfer that call to another agent by parking the
call, but when I dial 700 (to park the call) it hungs up and the call
isn't parked.
What would be?.
Andrew Thompson wrote:
I see you are accounting for local versus ld calls, but what about when the
person in the next cube over calls you?
Also, what will happen if no Caller ID was supplied, or it was marked
Private or Out-Of-Area?
___
Asterisk-Users
On Jun 22, 2004, at 7:44 AM, Matt wrote:
I've got a number (10) Cisco 7960's connected to my network. All the
phones
work perfectly except one.
The asterisk console keeps throwing up:
Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887
sip_poke_noanswer: Peer
'4001' is now UNREACHABLE!
Jun 22
Andrew Thompson wrote:
Are your internal-to-internal calls handled seperately from
external-to-internal calls?
I see you are accounting for local versus ld calls, but what
about when the
person in the next cube over calls you?
Well, I have a dual-purpose asterisk setup -- hosted business
As I understand it, you'd enter the extension at which you wish to be called
back at, your 9665 has nothing to do with it.
Instead of dialling 28 you could dial 9665 and that would add that SIP phone
as an agent to the cytelcs queue.
Steve
-Original Message-
From: Harold Workman
Prepaid app can not connect to the database,
[app_prepaid.so] = (Prepaid Application)
== Parsing '/etc/asterisk/prepaid.conf': Found
Jun 22 14:38:43 ERROR[-1084964736]: app_prepaid.c:127
check_connected: app_prepaid: cannot connect to
database server localhost. Calls will not be logged
==
Does anyone have this card working with Asterisk?
Thanks
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Daniel Jimenez wrote:
Trevor Peirce wrote:
The callee then presses # to transfer, and dials 9 then the number to
transfer the call to. Immediately the call is disconnected from the
callee, and the caller hears hold music. A couple seconds later when
the caller is transferred, call quality
What's the importance of the impedance matching in a FXO interface ?
Kind regards,
Miguel
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[EMAIL PROTECTED] wrote:
I am trying to compile pwlib in order to get h323 support working. Most
of the compile is fine except it falls over at the point below. Does
anyone have a solution?
You are not using the proper version of PWLib (and i'll guess Open H.323
also)
Read the README
Jeremy
Hi, i have a call center which receives many calls at day. Those calls are stored in a directory in my asterisk server as WAV files. The problem is that each call is divided in 2 files: an IN.WAV file and OUT.WAV file. TheOUT.WAV file is what im speaking to other person, the IN.WAV file is what
On Tuesday 22 June 2004 16:12, [EMAIL PROTECTED] wrote:
What's the importance of the impedance matching in a FXO interface ?
How important is call quality in your system? The two are directly related.
Regards,
Andrew
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Echo echo ech ech ec ec e e . .
:)
quote who=[EMAIL PROTECTED]
What's the importance of the impedance matching in a FXO interface ?
--
END OF LINE
-MCP
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That works OK for my Zaptel Hardware, because Asterisk detects when I pick
Up the phone. For my Grandstream BudgeTone 100 not so much, because there
is no traffic between phone and asterisk until I call some number.
Thank you anyway.
On Fri, 18 Jun 2004, Jay Milk wrote:
You're basically
soxmix will do it...On CVS HEAD asterisk Monitor() has an option to
automate this after each call.Do a 'show application monitor' to see all
the options.
-jwb
On Tuesday 22 June 2004 04:47 pm, Carlos Medina wrote:
Hi, i have a call center which receives many calls at day. Those
I quote from the wiki on voip-info:
In Asterisk, DND is controlled by dialing:
*78 to turn *on* Do Not Disturb mode and
*79 to turn *off* Do Not Disturb mode
How would one implement this?? I've seen dialplans that created these
extensions and you would need to use the Asterisk DB to set the
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