Re: [asterisk-users] config from DB
On Sun, Dec 7, 2008 at 9:19 AM, [EMAIL PROTECTED] wrote:
Hi Everyone,
Sorry, if this has been already discussed, but maybe someone encountered
interesting issue.
I have an * Dialplan configured with MySQL db. Everything works excellent,
except, I can't specify the ex-girlfriend logic. For example
Context exten Priorityapp
appdata
Default 400 1 Wait20
Default 400/100 1 Wait10
So, it does not matter what is my callerid, it will always go in wait(20) If
user with callerID 100 will try to dial x400, it will go to wait(20) as
well, and never wait(10). In another words Asterisk will disregard this
logic. If I place this logic in the extensions.conf file it will work as a
charm - no problem.
Thank you for your help.
Hi,
I believe it's a technological limitation. If it's in extensions.conf,
Asterisk can easilly draw a map of all possible matches in memory,
however for db it has to do query for each possible match. Perhaps
matching specific CallerID's was never thought of in realtime. If it
would work the same way as extension matching, probably a separate
column would be better (but that's just thought of how it should be).
Anyway, you can try creating separate context with callerid in exten
and then GoSub(${CALLERID(num)}) to it. Remember that ${EXTEN} is just
any number in your dialplan, and you can set it to CallerID when
jumping to other context. Upon returning from gosub it would be back
the same.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] IAX trunk mixing
If you set IAX2 debug on the HUNGARIAN machine and send the console
output
(or a wireshark output) I'll take a look.
At a guess it is a problem with your iax.conf file.
I generally find it clearer to have separate user and peer definitions
for
each system rather than relying on 'friend' which can be confusing.
Tim.
On 6 Dec 2008, at 20:14, Tóth Csaba wrote:
Hi List,
Help me pls, or you think this can be an asterisk bug and should i
make
a bug report?
thanks,
Csaba
Tóth Csaba írta:
hi,
i have a problem, and i am completely stuck with it, i hope someone
can
point out where is my config wrong.
I have three server, connect together with IAX trunking. The server
are
at romania (10.0.4.23, V1.4.22), hungary (10.0.1.23, V1.4.20) and
serbia
(10.0.3.4, V1.4.22). I have a hardphone (6251) connected to the
romanian
server, i dial a hungarian telephone number, the call goes to the
hungarian server well, but that server recognise the call come from
serbia.. and everything is mixed inside..
the phone starts at context do-phoning on the romanian server.
i called 003620XXX from the phone, and as you see, the romanian
server starts the call in good IAX trunk, but the hungarian server
identifies it badly..
Here is the message on the HUNGARIAN asterisk console about it:
-- Accepting AUTHENTICATED call from 10.0.4.23:
requested format = speex,
requested prefs = (gsm),
actual format = gsm,
host prefs = (),
priority = caller
-- Executing [EMAIL PROTECTED]:1]
MixMonitor(IAX2/telsrv-husrb-1541,
om_1228466966.19588_6251.wav) in
new stack
== Begin MixMonitor Recording IAX2/telsrv-husrb-1541
-- Executing [EMAIL PROTECTED]:2]
Macro(IAX2/telsrv-husrb-1541, kitelco|0620XXX) in new stack
-- Executing [EMAIL PROTECTED]:1] Set(IAX2/telsrv-husrb-1541,
telszam=0620XXX) in new stack
-- Executing [EMAIL PROTECTED]:2] Dial(IAX2/telsrv-husrb-1541,
ZAP/g2/0620XXX) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/0620XXX
-- Zap/37-1 is proceeding passing it to IAX2/telsrv-husrb-1541
here is ROMANIAN console:
[Dec 5 08:51:34] -- Executing [EMAIL PROTECTED]:1]
Set(SIP/6251-00c888c0, telszam=0620XXX) in new stack
[Dec 5 08:51:34] -- Executing [EMAIL PROTECTED]:2]
Set(SIP/6251-00c888c0, ~~EXTEN~~=s) in new stack
[Dec 5 08:51:34] -- Executing [EMAIL PROTECTED]:3]
Dial(SIP/6251-00c888c0, IAX2/telsrv-huro/0620XXX) in new
stack
[Dec 5 08:51:34] -- Called telsrv-huro/0620XXX
[Dec 5 08:51:34] -- Call accepted by 10.0.1.23 (format gsm)
[Dec 5 08:51:34] -- Format for call is gsm
[Dec 5 08:51:35] -- IAX2/telsrv-huro-16384 is proceeding
passing it
to SIP/6251-00c888c0
[Dec 5 08:51:35] -- Hungup 'IAX2/telsrv-huro-16384'
[Dec 5 08:51:35] == Spawn extension (macro-kitelsrvhu, s, 3)
exited
non-zero on 'SIP/6251-00c888c0' in macro 'kitelsrvhu'
[Dec 5 08:51:35] == Spawn extension (macro-kitelsrvhu, s, 3)
exited
non-zero on 'SIP/6251-00c888c0'
here are the snippets of the config files:
ROMANIAN server
iax.conf:
[telsrv-huro]
type=friend
host = 10.0.1.23
user = telsrv-huro
secret = xxx
bandwidth=low
qualify=yes
trunk=yes
timezone=Europe/Budapest
context=incoming-hu
[telsrv-rosrb]
type=friend
host = 10.0.3.4
user = telsrv-rosrb
secret = xxx
bandwidth=low
qualify=yes
trunk=yes
timezone=Europe/Bucharest
context=incoming-srb
extensions.ael:
context do-phoning {
includes {
do-nationalcall;
}
}
abstract context do-nationalcall {
_0036. = kitelsrvhu(06${EXTEN:4});
_6[2-8]XX = kitelsrvhu(${EXTEN});
_7[2-8]XX = kitelsrvhu(${EXTEN});
_00381. = kitelsrvsrb(${EXTEN:4});
_51[567]X = kitelsrvsrb(${EXTEN});
}
context incoming-hu {
includes {
template-companynumbers;
template-spec;
template-helyi;
template-mobil;
template-orszagos;
}
}
context incoming-srb {
includes {
template-companynumbers;
template-spec;
template-helyi;
template-mobil;
template-orszagos;
}
}
macro kitelsrvhu(telszam) {
Dial(IAX2/telsrv-huro/${telszam});
switch(${DIALSTATUS}) {
case CHANUNAVAIL:
Playback(/var/lib/asterisk/sounds/beeperr);
case CONGESTION:
Playback(/var/lib/asterisk/sounds/beeperr);
case BUSY:
Busy();
Wait(5);
};
Hangup();
}
macro kitelsrvsrb(telszam) {
Dial(IAX2/telsrv-srbro/${telszam});
switch(${DIALSTATUS}) {
case CHANUNAVAIL:
Playback(/var/lib/asterisk/sounds/beeperr);
case CONGESTION:
Playback(/var/lib/asterisk/sounds/beeperr);
case BUSY:
Busy();
Wait(5);
};
Re: [asterisk-users] Question on queue terms
hi there is only one king of queue, just queue if you want callback you shoulndt use agentcallbacklogin because is deprecated. you should use queueaddmember() or something like that. David 2008/12/6 Mike [EMAIL PROTECTED] Hello, I'm trying to setup a very simple queue with 5 SIP phones. I do NOT want the agents to have to be on the phone to get calls, but I want those 5 SIP phones to ring (according to the strategy chosen in queue.conf) to dispatch calls. Is this a call back queue, or is a callback queue a queue that calls back the customer? There is conflicting info when searching for callback queue. Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] International Calls still failing - Confused!
My international calls are not connecting. [general] pridialplan=dynamic ;prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 localprefix= I have the above in my zapta.conf - yet when I dial an international number, I get a ring, then I get the message the person you are calling, is currently unavailable This is an ubuntu machine, with a sangoma card, with FreePBX setup, on asterisk 1.4. Incoming calls are working fine - outgoing national, mobile, and local calls are also working fine. I cannot understand why international calls are not working. Any pointers, no matter how outrageous are very, very welcome! Kind Regards: Gabriel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on queue terms
Thanks. I know agentcallbacklogin is deprecated, but I am not even sure if I need anything special, I can`t find a clear answer. All the queues example I find are ones where the agent have to login. I simply need to have a queue that rings 5 SIP phones according to the ring strategy defined in queue.conf. Where exactly do I configure those SIP phones to be part of the queue? Is something as simple as agent = 1001,SIP/reg_1001 what I need? (or similar?) Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire Sent: Sunday, December 07, 2008 11:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question on queue terms hi there is only one king of queue, just queue if you want callback you shoulndt use agentcallbacklogin because is deprecated. you should use queueaddmember() or something like that. David 2008/12/6 Mike [EMAIL PROTECTED] Hello, I'm trying to setup a very simple queue with 5 SIP phones. I do NOT want the agents to have to be on the phone to get calls, but I want those 5 SIP phones to ring (according to the strategy chosen in queue.conf) to dispatch calls. Is this a call back queue, or is a callback queue a queue that calls back the customer? There is conflicting info when searching for callback queue. Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX trunk mixing
I am not sure this is the best way but this is the way I have three servers
connected to each other.
On server 1 I have this:
iax.conf:
register = f2t1:[EMAIL PROTECTED]
register = f3t1:[EMAIL PROTECTED]
[f1t2]
type=friend
host=dynamic
trunk=yes
secret=password
context=from_outside
deny=0.0.0.0/0.0.0.0
permit=2.2.2.2/255.255.255.255
timezone=America/Los_Angeles
[f1t3]
type=friend
host=dynamic
trunk=yes
secret=password
context=from_outside
deny=0.0.0.0/0.0.0.0
permit=3.3.3.3/255.255.255.255
timezone=America/Los_Angeles
extensions.conf:
exten = _2XX,1,NoOp(Calling to server 2)
exten = _2XX,n,Dial(IAX2/f1t2/${EXTEN})
exten = _2XX,n,Hangup()
exten = _3XX,1,NoOp(Calling to server 3)
exten = _3XX,n,Dial(IAX2/f1t3/${EXTEN})
exten = _3XX,n,Hangup()
On server 2 I have this:
iax.conf:
register = f1t2:[EMAIL PROTECTED]
register = f3t2:[EMAIL PROTECTED]
[f2t1]
type=friend
host=dynamic
trunk=yes
secret=password
context=from_outside
deny=0.0.0.0/0.0.0.0
permit=1.1.1.1/255.255.255.255
timezone=America/Los_Angeles
[f2t3]
type=friend
host=dynamic
trunk=yes
secret=password
context=from_outside
deny=0.0.0.0/0.0.0.0
permit=3.3.3.3/255.255.255.255
timezone=America/Los_Angeles
extensions.conf:
exten = _1XX,1,NoOp(Calling to server 1)
exten = _1XX,n,Dial(IAX2/f2t1/${EXTEN})
exten = _1XX,n,Hangup()
exten = _3XX,1,NoOp(Calling to server 3)
exten = _3XX,n,Dial(IAX2/f2t3/${EXTEN})
exten = _3XX,n,Hangup()
On server 3 I have this:
iax.conf:
register = f1t3:[EMAIL PROTECTED]
register = f2t3:[EMAIL PROTECTED]
[f3t1]
type=friend
host=dynamic
trunk=yes
secret=password
context=from_outside
deny=0.0.0.0/0.0.0.0
permit=1.1.1.1/255.255.255.255
timezone=America/Los_Angeles
[f3t2]
type=friend
host=dynamic
trunk=yes
secret=password
context=from_outside
deny=0.0.0.0/0.0.0.0
permit=2.2.2.2/255.255.255.255
timezone=America/Los_Angeles
extensions.conf:
exten = _1XX,1,NoOp(Calling to server 1)
exten = _1XX,n,Dial(IAX2/f3t1/${EXTEN})
exten = _1XX,n,Hangup()
exten = _2XX,1,NoOp(Calling to server 2)
exten = _2XX,n,Dial(IAX2/f3t2/${EXTEN})
exten = _2XX,n,Hangup()
--
Jim Dickenson
mailto:[EMAIL PROTECTED]
CfMC
http://www.cfmc.com/
From: Tóth Csaba [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sat, 06 Dec 2008 22:14:00 +0200
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] IAX trunk mixing
Hi List,
Help me pls, or you think this can be an asterisk bug and should i
make
a bug report?
thanks,
Csaba
Tóth Csaba írta:
hi,
i have a
problem, and i am completely stuck with it, i hope someone can
point out
where is my config wrong.
I have three server, connect together with IAX
trunking. The server are
at romania (10.0.4.23, V1.4.22), hungary
(10.0.1.23, V1.4.20) and serbia
(10.0.3.4, V1.4.22). I have a hardphone
(6251) connected to the romanian
server, i dial a hungarian telephone
number, the call goes to the
hungarian server well, but that server
recognise the call come from
serbia.. and everything is mixed inside..
the phone starts at context do-phoning on the romanian server.
i called
003620XXX from the phone, and as you see, the romanian
server starts the
call in good IAX trunk, but the hungarian server
identifies it badly..
Here is the message on the HUNGARIAN asterisk console about it:
--
Accepting AUTHENTICATED call from 10.0.4.23:
requested format =
speex,
requested prefs = (gsm),
actual format = gsm,
host prefs = (),
priority = caller
-- Executing
[EMAIL PROTECTED]:1]
MixMonitor(IAX2/telsrv-husrb-1541,
om_1228466966.19588_6251.wav) in
new stack
== Begin MixMonitor
Recording IAX2/telsrv-husrb-1541
-- Executing
[EMAIL PROTECTED]:2]
Macro(IAX2/telsrv-husrb-1541,
kitelco|0620XXX) in new stack
-- Executing [EMAIL PROTECTED]:1]
Set(IAX2/telsrv-husrb-1541,
telszam=0620XXX) in new stack
--
Executing [EMAIL PROTECTED]:2] Dial(IAX2/telsrv-husrb-1541,
ZAP/g2/0620XXX) in new stack
-- Requested transfer capability:
0x00 - SPEECH
-- Called g2/0620XXX
-- Zap/37-1 is proceeding
passing it to IAX2/telsrv-husrb-1541
here is ROMANIAN console:
[Dec 5 08:51:34] -- Executing [EMAIL PROTECTED]:1]
Set(SIP/6251-00c888c0, telszam=0620XXX) in new stack
[Dec 5
08:51:34] -- Executing [EMAIL PROTECTED]:2]
Set(SIP/6251-00c888c0,
Re: [asterisk-users] Question on queue terms
Mike wrote:
Thanks. I know agentcallbacklogin is deprecated, but I am not even
sure if I need anything special, I can`t find a clear answer. All the
queues example I find are ones where the agent have to login. I
simply need to have a queue that rings 5 SIP phones according to the
ring strategy defined in queue.conf.
Where exactly do I configure those SIP phones to be part of the
queue? Is something as simple as agent = 1001,SIP/reg_1001 what I
need? (or
I have 4 phones in an operator queue, it's setup so, if the operator is
busy enough that she can't grab the call (On inbound), it drops the
caller into the queue and start ringing the backup operators.
It puts an * in front of the number, to let the backup(s) know it's an
operator call and they can answer accordingly.
If nobody answers, then the callee is shown to the directory:
cat queues.conf
member = SIP/4100
member = SIP/4138
member = SIP/4140
member = SIP/4159
Then I have an extension that does the following:
;*
;* Place caller into the front-desk
;* queue and play hold music
;* for 120 seconds. If nobody picks
;* up call within that time, send
;* caller to the Directory
;*
exten = s,7,Set(CALLERID(num)=*${CALLERID(num)})
exten = s,8,Queue(front-desk|t|||120)
exten = s,9,Playback(local/transfer-dial)
exten = s,10,Goto(directory,s,1)
exten = s,11,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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[asterisk-users] Echo Cancelation
Hi All I Have an ISDN 30 circuit passing through an asterisk box to a legacy pbx, all is working well but I have had a problem that modems do not work, I thought of turning off echo cancelation but I cann t seem to find the ial switch do do it, could someone point me in the right direction to enable /disbale ec on a zap channel per call? Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International Calls still failing - Confused!
On Sun, Dec 7, 2008 at 12:02 PM, Mr Gabriel [EMAIL PROTECTED] wrote: My international calls are not connecting. [general] pridialplan=dynamic ;prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 localprefix= I have the above in my zapta.conf - yet when I dial an international number, I get a ring, then I get the message the person you are calling, is currently unavailable This is an ubuntu machine, with a sangoma card, with FreePBX setup, on asterisk 1.4. Incoming calls are working fine - outgoing national, mobile, and local calls are also working fine. I cannot understand why international calls are not working. Any pointers, no matter how outrageous are very, very welcome! Kind Regards: Gabriel ___ How many digits is the telco expecting? Have you called your telo for help (Better to leave out the Asterisk stuff Are you setting caller ID and is it one of your numbers, if not the BTN. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on queue terms
Thanks, that`s pretty close to what I want. I got confused between members
and agents.
I have enough to go on with this!
Regards,
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Sunday, December 07, 2008 12:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on queue terms
Mike wrote:
Thanks. I know agentcallbacklogin is deprecated, but I am not even
sure if I need anything special, I can`t find a clear answer. All the
queues example I find are ones where the agent have to login. I
simply need to have a queue that rings 5 SIP phones according to the
ring strategy defined in queue.conf.
Where exactly do I configure those SIP phones to be part of the
queue? Is something as simple as agent = 1001,SIP/reg_1001 what I
need? (or
I have 4 phones in an operator queue, it's setup so, if the operator is
busy enough that she can't grab the call (On inbound), it drops the
caller into the queue and start ringing the backup operators.
It puts an * in front of the number, to let the backup(s) know it's an
operator call and they can answer accordingly.
If nobody answers, then the callee is shown to the directory:
cat queues.conf
member = SIP/4100
member = SIP/4138
member = SIP/4140
member = SIP/4159
Then I have an extension that does the following:
;*
;* Place caller into the front-desk
;* queue and play hold music
;* for 120 seconds. If nobody picks
;* up call within that time, send
;* caller to the Directory
;*
exten = s,7,Set(CALLERID(num)=*${CALLERID(num)})
exten = s,8,Queue(front-desk|t|||120)
exten = s,9,Playback(local/transfer-dial)
exten = s,10,Goto(directory,s,1)
exten = s,11,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
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Re: [asterisk-users] Check variables on a live system - Is it possible?
On Fri, Dec 05, 2008 at 10:52:46AM +, Mr Gabriel wrote: Is it possible to check certain varibles on the live system, for example, what the current setting for pridialplan is? I know what is set in the config files, but the behaviour does not reflect this. Can this be checked? Generally: no. Something you can check is what were the configuration files that asterisk has read (as those values are remembered). The specific setting you mention is basically per-channel (though is really per-span, IIRC). You could patch 'dahdi show channel' to show its value. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX trunk mixing
Just use SIP so you don't have to back later to change all your IAX2 entries to SIP. On Fri, Dec 5, 2008 at 4:23 AM, Tóth Csaba [EMAIL PROTECTED] wrote: hi, i have a problem, and i am completely stuck with it, i hope someone can point out where is my config wrong. I have three server, connect together with IAX trunking. The server are at romania (10.0.4.23, V1.4.22), hungary (10.0.1.23, V1.4.20) and serbia (10.0.3.4, V1.4.22). I have a hardphone (6251) connected to the romanian server, i dial a hungarian telephone number, the call goes to the hungarian server well, but that server recognise the call come from serbia.. and everything is mixed inside.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unexpected behaviour in ForkCDR
Dear members of the list;
I am writing in the hope to get some help with a very peculiar problem with
my new asterisk 1.6.0.1 installation. The same code runs on version 1.2
without problems, but it seems the behaviour has changed (also on 1.4.7,
which I tried).
Please consider the following extension:
exten = 1213,1,Answer
exten = 1213,n,Set(counter=X)
exten = 1213,n(again),Set(CDR(accountcode)=forkcdr-test)
exten = 1213,n,Set(CDR(userfield)= ${counter})
exten = 1213,n,ForkCDR()
exten = 1213,n,Playback(one-moment-please)
exten = 1213,n,Wait(3)
exten = 1213,n,Set(counter=X${counter})
exten = 1213,n,Goto(again)
exten = 1213,n,Hangup
If left running, the above code should produce multiple Call Data Records,
each with a longer sting of Xs in the userfield. This is indeed also the
case on my 1.2 installation, but on 1.4.7 and 1.6.0.1, only one record (the
first) is written.
After a lot of testing, I discovered that if I add the R option to the
ForkCDR command (to prevent it from resetting the new CDR), multiple records
are in fact written, but they all contain the same call-info:
fork-cdr-test,ForkCDR,R,2008-11-20 09:00:28,2008-11-20
09:00:28,2008-11-20 09:00:40,12,12,,,1227171628.5,I
fork-cdr-test,ForkCDR,R,2008-11-20 09:00:28,2008-11-20
09:00:28,2008-11-20 09:00:40,12,12,,,1227171628.5,II
fork-cdr-test,ForkCDR,R,2008-11-20 09:00:28,2008-11-20
09:00:28,2008-11-20 09:00:40,12,12,,,1227171628.5,III
fork-cdr-test,,Wait,3,2008-11-20 09:00:28,2008-11-20
09:00:28,2008-11-20 09:00:40,12,12,,,1227171628.5,III
This suggests to me, that the broken (?) behaviour may be in the reset
function rather than in the ForkCDR itself.
I have also noticed that the unique-id column now contains the UNIX
timestamp. Since the timestamp is the same for all the records, it's hardly
unique. I tried changing the loguniqueid parameter in cdr.conf, but with
no effect.
Any help or suggestions on the above issues would be greatly appreciated.
Thank you very much in advance.
Best Regards
Torben
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Re: [asterisk-users] International Calls still failing - Confused!
In article [EMAIL PROTECTED], Mr Gabriel [EMAIL PROTECTED] wrote: My international calls are not connecting. [general] pridialplan=dynamic ;prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 localprefix= Looks ok. However, you might also want to try pridialplan=unknown if you haven't done so already. I have the above in my zapta.conf - yet when I dial an international number, I get a ring, then I get the message the person you are calling, is currently unavailable Have you been able to determine whether the ring and message are coming from the local Asterisk, or over the line from BT? This is an ubuntu machine, with a sangoma card, with FreePBX setup, on asterisk 1.4. Incoming calls are working fine - outgoing national, mobile, and local calls are also working fine. I cannot understand why international calls are not working. Any pointers, no matter how outrageous are very, very welcome! I think we need to see a PRI trace of a failed call. Please could you do the following: host*CLI pri set debug file /tmp/pri.txt host*CLI pri debug span 1 Then make just one outgoing international call attempt. After it has failed, do this: host*CLI pri no debug span 1 host*CLI pri unset debug file Then post the contents of /tmp/pri.txt. If possible try to ensure that your email program does NOT do automatic line wrapping at a certain column, to preserve the format of the log file. That will show us (a) exactly what your Asterisk is sending to BT, and (b) the error code being returned by BT. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on queue terms
Hi
Members are every one who can answer a call.
you have users and agents
what you want is a user.
in extensions.conf
put something like
exten = *555,1,addqueuememeber(SIP/${CALLERID})
exten = *556,1,removequeuememeber(SIP/${CALLERID})
thats all if the name are nor correct chech the aplications names...
David
2008/12/7 Mike [EMAIL PROTECTED]
Thanks, that`s pretty close to what I want. I got confused between members
and agents.
I have enough to go on with this!
Regards,
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Sunday, December 07, 2008 12:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on queue terms
Mike wrote:
Thanks. I know agentcallbacklogin is deprecated, but I am not even
sure if I need anything special, I can`t find a clear answer. All the
queues example I find are ones where the agent have to login. I
simply need to have a queue that rings 5 SIP phones according to the
ring strategy defined in queue.conf.
Where exactly do I configure those SIP phones to be part of the
queue? Is something as simple as agent = 1001,SIP/reg_1001 what I
need? (or
I have 4 phones in an operator queue, it's setup so, if the operator is
busy enough that she can't grab the call (On inbound), it drops the
caller into the queue and start ringing the backup operators.
It puts an * in front of the number, to let the backup(s) know it's an
operator call and they can answer accordingly.
If nobody answers, then the callee is shown to the directory:
cat queues.conf
member = SIP/4100
member = SIP/4138
member = SIP/4140
member = SIP/4159
Then I have an extension that does the following:
;*
;* Place caller into the front-desk
;* queue and play hold music
;* for 120 seconds. If nobody picks
;* up call within that time, send
;* caller to the Directory
;*
exten = s,7,Set(CALLERID(num)=*${CALLERID(num)})
exten = s,8,Queue(front-desk|t|||120)
exten = s,9,Playback(local/transfer-dial)
exten = s,10,Goto(directory,s,1)
exten = s,11,Hangup()
Doug
--
Ben Franklin quote:
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Safety, deserve neither Liberty nor Safety.
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(='.'=)This is Bunny. Copy and paste bunny into your
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Re: [asterisk-users] Rate My Dialplan Contest Announced - Win a Phone or Copies of APSTel Visual Dialplan Std or Pro!
3rd place: An APSTel dial plan (standard license) donated by APSTel! So... if you can write the slickest dialplan, you get dialplan generator software? Hi Andrew, Well, the thought is that most people are using SmartDraw, Dia, Visio, Illustrator or Corel Draw to create these types of diagrams so we thought we would give away software that is fully intended for this type of work :) Thanks, Matt G ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hobart/Tasmanian humans
Is there anyone is Tasmania (esp Hobart) doing Asterisk work? PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
