Re: [asterisk-users] Audiocodes MP-11X configuration to work withAsterisk
On 28 Dec 2008, at 18:36, Razza wrote: Please see below Console Messages, Pertinent section of SIP.CONF and AudioCodes Config. Console Messages: Dec 28 18:14:45 NOTICE[19109]: chan_sip.c:9808 handle_request_register: Registration from 'sip:2...@192.168.10.4' failed for '192.168.10.4' Dec 28 18:14:45 NOTICE[19109]: chan_sip.c:9808 handle_request_register: Registration from 'sip:2...@192.168.10.4' failed for '192.168.10.4' -- Saved useragent Audiocodes-Sip-Gateway-MP-114 FXS_FXO/v. 5.00A.035.003 for peer 272 -- Saved useragent Audiocodes-Sip-Gateway-MP-114 FXS_FXO/v. 5.00A.035.003 for peer 271 SIP.conf (pertinent section): ;- MP114 Play --- [pstn1] ;MP-114 FXO Port 1 type=friend username=276 secret=mp276 context=home_phones allow=alaw dtmfmode=inband host=192.168.10.4 nat=never canreinvite=no [pstn2] ;MP-114 FXO Port 2 type=friend username=277 secret=mp277 regextn=277 context=home_phones allow=alaw dtmfmode=inband host=192.168.10.4 nat=never canreinvite=no [audiocodes] username=audiocodes type=peer secret=NotTellingYou qualify=no host=123.123.123.123 dtmfmode=rfc2833 disallow=all context=from-provider-someone allow=alaw allow=ulaw insecure=port,invite That is what I use for mine (granted its a PRI gateway but I bet the SIP code is similar). ___ -- 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] Join empty queue property
I want the callers don't join in a queue when the agents are busy. I suposse it is easy but i can't get the solution for this. Can you suggest me something? Thanks. ___ -- 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] Background stress test
Hello, We did small test with sipp to test Asterisk Background command capability. Our goal was 700 sim. calls on HP Proliant DL160 G5 E5405 1 x Quad Core Xeon 2Ghz 2 Gb RAM Asterisk 1.4.18.1 Centos 5.2 We reached more then 1000 when our network (100mbps) become a bottleneck. As we achieved our goal - no further testing was performed. As conclusion - we are very happy with Asterisk in this case. If somebody is interested - more details are here: http://wiki.kolmisoft.com/index.php/Asterisk_Background_performance_test Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] Problems with sip registrations through HP Procurve 7102dl
Is it causing an issue? There are lots of firewalls that do nat and change the source port of packets to some random udp port. In my experience, for outbound registrations, it generally doesn't cause an issue. Robert Augustyn wrote: Hi, I have a strange problem, when I try to connect to les.net from our local asterisk server through Procurve router I seems to be connecting on any port above 1024 and when I reload sip the port is changing too ... So I never get 5060? Any ideas on what is going on and how to resolve it? Sincerely, Robert Augustyn 519-997-3106 ext:802 www.linqone.com http://www.linqone.com/ http://www.linqone.com/ ___ -- 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 ___ -- 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] 1.6, CDR and h extension
I have two version 1.6 Asterisks running. One is a small hobbyist
thing just at home, and the other is handling calls for several
customers.
On both, I have added the line
exten = h,1,Set(CDR(hangupcause)=${HANGUPCAUSE})
to all relevant contexts.
On my little hobbyist box this works perfectly; all calls have their
hangupcauses recorded with cdr_adaptive_odbc and cdr_custom. On the
production server, it only works sporadically; for 4 out of 5 calls
the hangupcause field is empty. If I look at the Asterisk console, I
see a message like this for every call:
-- Executing [...@calltovpbx:1] Set(DAHDI/49-1, CDR(hangupcause)=16)
in new stack
Unfortunately hangupcause doesn't actually make it to the database or
the csv file, despite that message.
My cdr.conf on both servers:
[general]
enable=yes
batch=yes
size=10
time=60
scheduleronly=no
safeshutdown=yes
endbeforehexten=no
I am probably just being stupid somehow, but I cannot figure out what
is wrong. Both asterisks are 1.6.0.1-2.fc10.
/Benny
___
-- 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] Manager API
Hi I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial out from manager's console and with Asterisk 1.4.X this settings were OK. Action: Originate Channel: SIP/384 Context: main Exten: 102 Priority: 1 Callerid: 384 I could dial out, but with asterisk 1.6 I get this error. Response: Error Message: Channel not specified I have originate and system privilege in managers.conf. Does anyone know the solution for this kind of error. Cheers Andrew ___ -- 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] Manager API
I had the same problem, I think with that version. You need to use a more current rc version. I think I am using rc2 but rc3 has been released as I recall. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ From: Andrew Nowrot andrew.now...@gmail.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 29 Dec 2008 16:10:23 +0100 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Manager API Hi I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial out from manager's console and with Asterisk 1.4.X this settings were OK. Action: Originate Channel: SIP/384 Context: main Exten: 102 Priority: 1 Callerid: 384 I could dial out, but with asterisk 1.6 I get this error. Response: Error Message: Channel not specified I have originate and system privilege in managers.conf. Does anyone know the solution for this kind of error. Cheers Andrew ___ -- 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 ___ -- 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] Digium sites down for maintenance
Hi all, Sorry for the inconvenience but the following Digium-hosted sites are currently down for maintenance purposes: svn.digium.com svncommunity.digium.com bugs.digium.com packages.digium.com reviewboard.digium.com Apologies for the unannounced downtime; we will let you know when they are back up. Happy Holidays, Mark Michelson ___ -- 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] Manager API
hi
that is a bug in manager.c
where saysstatic int action_timeout(struct mansession *s, const struct
message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, Channel);
int timeout = atoi(astman_get_header(m, Timeout));
if (!ast_strlen_zero(name)) {
astman_send_error(s, m, No channel specified);
return 0;
}
should say
static int action_timeout(struct mansession *s, const struct message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, Channel);
int timeout = atoi(astman_get_header(m, Timeout));
if (ast_strlen_zero(name)) {
astman_send_error(s, m, No channel specified);
return 0;
}
2008/12/29 Andrew Nowrot andrew.now...@gmail.com
Hi
I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial
out from manager's console and with Asterisk 1.4.X this settings were OK.
Action: Originate
Channel: SIP/384
Context: main
Exten: 102
Priority: 1
Callerid: 384
I could dial out, but with asterisk 1.6 I get this error.
Response: Error
Message: Channel not specified
I have originate and system privilege in managers.conf.
Does anyone know the solution for this kind of error.
Cheers
Andrew
___
-- 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
hi
that is a bug in manager.c
where saysstatic int action_timeout(struct mansession *s, const struct
message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, Channel);
int timeout = atoi(astman_get_header(m, Timeout));
if (!ast_strlen_zero(name)) {
astman_send_error(s, m, No channel specified);
return 0;
}
should say
static int action_timeout(struct mansession *s, const struct message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, Channel);
int timeout = atoi(astman_get_header(m, Timeout));
if (ast_strlen_zero(name)) {
astman_send_error(s, m, No channel specified);
return 0;
}
PD;
Double posting strikes again.
--
(\__/)
(='.'=)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] Join empty queue property
Then why use a queue? The purpose of a queue is exactly to keep people waiting while agents are all busy. The only way I can see something like what you want is to put a very low timeout (maybe 10 seconds) so if all your agents are busy then the caller will get dropped from the queue and continue with the dialplan where you can redirect them. On Mon, 2008-12-29 at 10:26 -0200, equis software wrote: I want the callers don't join in a queue when the agents are busy. I suposse it is easy but i can't get the solution for this. Can you suggest me something? Thanks. ___ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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] Join empty queue property
equis software wrote: I want the callers don't join in a queue when the agents are busy. I suposse it is easy but i can't get the solution for this. Can you suggest me something? Thanks. Unless you are using the trunk version of Asterisk right now, this can't be done very easily. In trunk, the joinempty setting for queues has been modified to not only allow the current allowed values of no, yes, loose, and strict, but also to allow a comma-separated list of conditions under which you consider a queue member to be unavailable. There is a detailed explanation of all the allowed conditions in the queues.conf.sample file in trunk's configs/ directory in the source. Since this feature is already in trunk but not in any released version of Asterisk, it will be present in Asterisk version 1.6.2. Mark Michelson ___ -- 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] Join empty queue property
Because I want this behabior, if all agents are busy I redirect the call to an IVR to register the question. In fact there is a property in queue.conf ; This setting controls whether callers can join a queue with no members. There ; are three choices: ; ; yes- callers can join a queue with no members or only unavailable members ; no - callers cannot join a queue with no members ; strict - callers cannot join a queue with no members or only unavailable ; members ; ; joinempty = yes I try to use joinempty = strict But didn´t work... On Mon, Dec 29, 2008 at 2:31 PM, Carlos Chavez cur...@telecomabmex.comwrote: Then why use a queue? The purpose of a queue is exactly to keep people waiting while agents are all busy. The only way I can see something like what you want is to put a very low timeout (maybe 10 seconds) so if all your agents are busy then the caller will get dropped from the queue and continue with the dialplan where you can redirect them. On Mon, 2008-12-29 at 10:26 -0200, equis software wrote: I want the callers don't join in a queue when the agents are busy. I suposse it is easy but i can't get the solution for this. Can you suggest me something? Thanks. ___ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- 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 ___ -- 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] Asterisk as MGCP client
Has there been any work done on using Asterisk as an MGCP client? I see the 'Asterisk MGCP channels' page on voip-info hasn't been updated since 2006, and I was wondering if anyone has been able to accomplish this yet. I have a situation where it might be helpful to have an Asterisk system register to a local phone service using MGCP. Thanks for your help. Bob ___ -- 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] Dailplan code for holiday detection?
On Tuesday 23 December 2008 04:08:00 pm Daniel Hazelbaker wrote:
We chose to use a mySQL database to store the holiday information.
When a call is answered we query the database to see if there is a
holiday greeting recorded, if so we play the indicated greeting,
otherwise play the default menu greeting. (We do our dialplans in AEL)
context checkHoliday {
s =
{
begin:
MYSQL(Connect temp communicator username password
asterisk);
MYSQL(Query resultid ${temp} SELECT greeting FROM
menuGreetings WHERE startTime=FROM_UNIXTIME(${EPOCH}) AND
endTime=FROM_UNIXTIME(${EPOCH}) LIMIT 1);
MYSQL(Fetch foundRow ${resultid} sqlGreeting);
MYSQL(Clear ${resultid});
MYSQL(Disconnect ${temp});
if (${foundRow}==1)
{
Background(custom/mainMenu/${sqlGreeting});
goto mainMenu,s,begin;
}
else
{
goto checkTime,s,begin;
}
}
includes
{
mainMenu;
tempGreeting;
voicemail;
publicExt;
}
};
The 'checkTime' context simply checks if we are open or closed and
plays the appropriate greeting (if no holiday greeting is found).
This dialplan is illustrative of the particular problem of the MYSQL command
in that no cleanup is performed if the dialplan terminates abnormally. If a
device hangup occurs between the Connect and Disconnect, or worse, between
the Query and the Clear, then extra resources will be consumed until a restart
is performed. To avoid this problem, you should ensure that you always clear
your query resources and disconnect your handles in the h extension.
Or use func_odbc, which performs this sort of cleanup for you.
--
Tilghman
___
-- 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] SMS text messaging capabilities
On Friday 26 December 2008 04:26:24 am Elliot Murdock wrote: I noticed that Asterisk has an SMS function, but I am not farmiliar enough with that technology to make it useful. If you are located on the US side of the pond and not in a locality which has British Telecom-supplied lines, I believe you'll find the SMS command to be of limited (read: no) use. AFAIK, those are the only type of lines which support the particular type of signalling expected for the SMS application to work. -- Tilghman ___ -- 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] 1.6, CDR and h extension
On Monday 29 December 2008 08:57:19 am Benny Amorsen wrote:
I have two version 1.6 Asterisks running. One is a small hobbyist
thing just at home, and the other is handling calls for several
customers.
On both, I have added the line
exten = h,1,Set(CDR(hangupcause)=${HANGUPCAUSE})
to all relevant contexts.
On my little hobbyist box this works perfectly; all calls have their
hangupcauses recorded with cdr_adaptive_odbc and cdr_custom. On the
production server, it only works sporadically; for 4 out of 5 calls
the hangupcause field is empty. If I look at the Asterisk console, I
see a message like this for every call:
-- Executing [...@calltovpbx:1] Set(DAHDI/49-1, CDR(hangupcause)=16)
in new stack
Unfortunately hangupcause doesn't actually make it to the database or
the csv file, despite that message.
My cdr.conf on both servers:
[general]
enable=yes
batch=yes
size=10
time=60
scheduleronly=no
safeshutdown=yes
endbeforehexten=no
I am probably just being stupid somehow, but I cannot figure out what
is wrong. Both asterisks are 1.6.0.1-2.fc10.
The only thing that occurs to me is that you might be using ForkCDR on your
production machine, in which case, you might want:
Set(CDR(hangupcause,r)=${HANGUPCAUSE})
for recursive setting of the hangupcause.
--
Tilghman
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Most Digium services are back on-line
Hello again, The following Digium-hosted services are no longer down and should be functioning properly: svn.digium.com svncommunity.digium.com packages.digium.com reviewboard.digium.com At this time, bugs.digium.com remains down, but we will have it back up as soon as we can. Thank you for your patience. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API
Hi
Thanks for so fast reply, but I already have this part like this:
static int action_timeout(struct mansession *s, const struct message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, Channel);
int timeout = atoi(astman_get_header(m, Timeout));
if (ast_strlen_zero(name)) {
astman_send_error(s, m, No channel specified);
return 0;
}
no negation in 7th line. So I guest it is not the case.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL: how to check if variable is defined
Hi!
I use an if condition in extensions.ael to check if a channel variable
is defined and if defined I add a certain header:
context toNormaleRufe {
_X. = {
if (${NUMBER}) {
SIPAddHeader(X-NUMBER: ${NUMBER});
};
...
};
This works fine, except NUMBER starts with the + sign.
I tried using quotes but
if (${NUMBER})
evaluates always true.
What is the suggested way to solve this?
thanks
klaus
___
-- 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] Manager API
2008/12/29 Andrew Nowrot andrew.now...@gmail.com
Hi
Thanks for so fast reply, but I already have this part like this:
static int action_timeout(struct mansession *s, const struct message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, Channel);
int timeout = atoi(astman_get_header(m, Timeout));
if (ast_strlen_zero(name)) {
astman_send_error(s, m, No channel specified);
return 0;
}
no negation in 7th line. So I guest it is not the case.
___
-- 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
did you re compile and re installed?
make
make install
after the code change?
david
--
(\__/)
(='.'=)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] Manager API
I did not need to change the code. My manager.c already has all the lines you specified that are wrong. did you re compile and re installed? make make install after the code change? david Cheers ___ -- 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] Problems with sip registrations through HP Procurve 7102dl
It is not the source port being changed, it looks like the destination port is being changed. robert -Original message- From: pe...@networkoblivion.com Sent: Mon 29-12-2008 09:49 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Subject: Re: [asterisk-users] Problems with sip registrations through HP Procurve 7102dl Is it causing an issue? There are lots of firewalls that do nat and change the source port of packets to some random udp port. In my experience, for outbound registrations, it generally doesn't cause an issue. Robert Augustyn wrote: Hi, I have a strange problem, when I try to connect to les.net from our local asterisk server through Procurve router I seems to be connecting on any port above 1024 and when I reload sip the port is changing too ... So I never get 5060? Any ideas on what is going on and how to resolve it? Sincerely, Robert Augustyn 519-997-3106 ext:802 www.linqone.com http://www.linqone.com/ http://www.linqone.com/ ___ -- 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 ___ -- 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 ___ -- 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] Dailplan code for holiday detection?
On Tue, 2008-12-23 at 10:11 -0800, Dan Austin wrote:
This has been on my ToDo list far too long.
I have a small call-center setup, with basic
time of day/day of week validation before putting
callers in the queues.
With the holidays upon us, I need to add check to
see if 'today' is a holiday so I do not put callers
in unmanned queues. Due to how the agents work, I have
to allow joinwhenempty.
Does anyone have a snippet of dialplan code, perhaps using
Astdb, to check it 'today' is a listed holiday?
Thanks,
Dan
Here is a little script I use on my home system;
There are others on
http://voip-info.org/wiki/view/AEL+Example+Snippets
ifTime(*|*|20-25|dec)
{
Playback(greetings/christmas);
}
else ifTime(*|*|31|dec)
{
Playback(greetings/newyear);
}
else ifTime(*|*|1|jan)
{
Playback(greetings/newyear);
}
else ifTime(*|*|14|feb)
{
Playback(greetings/valentines);
}
else ifTime(*|*|17|mar)
{
Playback(greetings/stPat);
}
else ifTime(*|*|31|oct)
{
Playback(greetings/halloween);
}
else ifTime(*|mon|15-21|jan)
{
Playback(greetings/mlkDay);
}
else ifTime(*|thu|22-28|nov)
{
Playback(greetings/thanksgiving);
}
else ifTime(*|mon|25-31|may)
{
Playback(greetings/memorial);
}
else ifTime(*|mon|1-7|sep)
{
Playback(greetings/labor);
}
else ifTime(*|mon|15-21|feb)
{
Playback(greetings/president);
}
else ifTime(*|sun|8-14|may)
{
Playback(greetings/mothers);
}
else ifTime(*|sun|15-21|jun)
{
Playback(greetings/fathers);
}
else
{
Playback(greetings/hello); // None of the above? Just
a plain hello will do
}
murf
--
Steve Murphy
Digium, Inc. | Software Developer
57 Lane 17, Cody, WY 82414 USA
direct: +1 256-428-6002
mobile: +1 307-899-5535
fax/home: +1 307-754-5675
irc: codefreeze | jabber: m...@digium.com
Check us out at: www.digium.com www.asterisk.org
smime.p7s
Description: S/MIME cryptographic signature
___
-- 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] AEL Variable Warning Messages
On Tue, 2008-12-23 at 12:14 -0600, Brent Davidson wrote:
I have two offices sharing a phone system. They also share a common
internal context because all of the employees of the second office also
work for the first office. Each office has 4 outside lines and I have
defined 2 channel groups in my zapata.conf. The second office needs all
of their outgoing calls to go out over their lines so the people they
call will have the correct callerID. I created an asterisk database and
with entries in the database for all extensions in the second office and
defined the following macro:
globals {
CONSOLE=Console/dsp;
TRUNK=Zap/r1;
TCTC_Operator=15;
Law_Operator=12;
};
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
It's working and correctly routing outside calls, but I get the
following messages when I reload the extensions.ael file:
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 93-93: expression
Zap/r2 has operators, but no variables. Interesting...
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 95-95: expression
Zap/r1 has operators, but no variables. Interesting...
Any idea what is causing the warnings?
Yes, I do! I was concerned that users were falling into a common
error, where they forget to wrap variable references in $(); so,
if it looks like an expr has arithmetic operators, but no variable
refs, then you get this message.
Yes, I *could* have made it more intelligent. File a bug, and I'll
see if I can do so. At the worst, you can ignore this warning, or
I can simply remove this overly-simple warning.
murf
Thanks,
Brent
--
Steve Murphy
Digium, Inc. | Software Developer
57 Lane 17, Cody, WY 82414 USA
direct: +1 256-428-6002
mobile: +1 307-899-5535
fax/home: +1 307-754-5675
irc: codefreeze | jabber: m...@digium.com
Check us out at: www.digium.com www.asterisk.org
smime.p7s
Description: S/MIME cryptographic signature
___
-- 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] noise in Asterisk 1.4 and 1.6 versions
Hi Abel - I had installed Asterisk 1.4 and when I call to a exist extension, the voice have noise, but, when I call to a extension does no exist, asterisk played a voice that say me that extension does no exist, but without noise I want I some body can test with a softphone my server, ip: 75.74.115.209 user: ramses pass: ramses the extension 1000 exist, try what ever other extension does not exist to hear the difference.. I would be willing to bet that the clear voice that you hear is generated by your phone (probably x-lite?), and not by asterisk. I'd also be willing to bet that fuzzy voice is caused by a bug that is present in certain versions of gcc. What version do you have? You can fix by compiling with the DONT_OPTIMIZE option (which should give you clear sounds), or just upgrade to a more recent version of gcc. - Noah ___ -- 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] AEL: how to check if variable is defined
Klaus Darilion schrieb:
I use an if condition in extensions.ael to check if a channel variable
is defined and if defined I add a certain header:
context toNormaleRufe {
_X. = {
if (${NUMBER}) {
SIPAddHeader(X-NUMBER: ${NUMBER});
};
...
};
This works fine, except NUMBER starts with the + sign.
I tried using quotes but
if (${NUMBER})
evaluates always true.
What is the suggested way to solve this?
if (${NUMBER} != ) {
// ...
}
That doesn't tell you whether the variable is defined but in
most cases (if any) that doesn't matter anyway.
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
--
___
-- 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] Join empty queue property
equis software equissoftw...@gmail.com writes: I want the callers don't join in a queue when the agents are busy. I suposse it is easy but i can't get the solution for this. Can you suggest me something? If you don't need the full functionality of queues, they are reasonably easy to reinvent in the dial plan. If you just do something like exten = _X!,n,Dial(SIP/phone1SIP/phone2SIP/phone3) you will automatically get to the next priority if all agents are busy. If you need dynamic agents, that can be handled by storing the agents in a variable in the asterisk database -- or even better, in an external database. /Benny ___ -- 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] Dailplan code for holiday detection?
I'm making extensive use of the MYSQL command.do you know if this behavior is considered a bug or not? This dialplan is illustrative of the particular problem of the MYSQL command in that no cleanup is performed if the dialplan terminates abnormally. If a device hangup occurs between the Connect and Disconnect, or worse, between the Query and the Clear, then extra resources will be consumed until a restart is performed. To avoid this problem, you should ensure that you always clear your query resources and disconnect your handles in the h extension. Or use func_odbc, which performs this sort of cleanup for you. ___ -- 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] AEL: how to check if variable is defined
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
I use an if condition in extensions.ael to check if a channel variable
is defined and if defined I add a certain header:
context toNormaleRufe {
_X. = {
if (${NUMBER}) {
SIPAddHeader(X-NUMBER: ${NUMBER});
};
...
};
This works fine, except NUMBER starts with the + sign.
I tried using quotes but
if (${NUMBER})
evaluates always true.
What is the suggested way to solve this?
if (${NUMBER} != ) {
// ...
}
That doesn't tell you whether the variable is defined but in
most cases (if any) that doesn't matter anyway.
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
--
___
-- 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] Cut Through DTMF caller ID on SIP phone
Trevor Peirce schrieb: Sriram wrote: 2. Is there any way to block the caller id from appearing on the SIP Phone ? my trunk is E1 PRI while i used softphones internally - i dont want my agents to see the caller id - is their any way to block caller ids from appearing on softphones ? a) SetCallerPres(restircted) or b) Set(CALLERID(name)=Anonymous) Set(CALLERID(all)=Anonymous anonymous); Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Dailplan code for holiday detection?
Next question: When you say extra resources will be consumed until a restart is performed. Do you mean I have to restart asterisk to free up said resources? Will a reload do it also? This dialplan is illustrative of the particular problem of the MYSQL command in that no cleanup is performed if the dialplan terminates abnormally. If a device hangup occurs between the Connect and Disconnect, or worse, between the Query and the Clear, then extra resources will be consumed until a restart is performed. To avoid this problem, you should ensure that you always clear your query resources and disconnect your handles in the h extension. Or use func_odbc, which performs this sort of cleanup for you. ___ -- 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] Dailplan code for holiday detection?
Adam Moffett schrieb: I'm making extensive use of the MYSQL command.do you know if this behavior is considered a bug or not? This dialplan is illustrative of the particular problem of the MYSQL command in that no cleanup is performed if the dialplan terminates abnormally. If a device hangup occurs between the Connect and Disconnect, or worse, between the Query and the Clear, then extra resources will be consumed until a restart is performed. To avoid this problem, you should ensure that you always clear your query resources and disconnect your handles in the h extension. Or use func_odbc, which performs this sort of cleanup for you. http://lists.digium.com/pipermail/asterisk-users/2008-December/223847.html Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] AEL: how to check if variable is defined
Philipp Kempgen wrote:
*snipped
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what ISNULL is for?
___
-- 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] AEL: how to check if variable is defined
Philipp Kempgen wrote:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
I use an if condition in extensions.ael to check if a channel variable
is defined and if defined I add a certain header:
context toNormaleRufe {
_X. = {
if (${NUMBER}) {
SIPAddHeader(X-NUMBER: ${NUMBER});
};
...
};
This works fine, except NUMBER starts with the + sign.
I tried using quotes but
if (${NUMBER})
evaluates always true.
What is the suggested way to solve this?
if (${NUMBER} != ) {
// ...
}
That doesn't tell you whether the variable is defined but in
most cases (if any) that doesn't matter anyway.
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what EXISTS() is for?
-Dave
___
-- 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] DTMF does not work
I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both set to register with the same account right now. I shut Asterisk down on the one and now we don't drop any longer. doh!!! Last, We are having DTMF problems with our provider (via:talk). Does anyone have any experience with them and if so can you share it? via:talk does have a sample sip.conf and extensions.conf file to use but the dial plan they set up does not require any DTMF so they may never have tested it. We have tried inband, auto, rfc2833 for our DTMF and nothing works. I have submitted a ticket with them but the last time I did that they never responded so that is why I am posting here. I signed up with another SIP provider for a test account and the DTMF passes no problem from them so I must conclude there is some setting that via:talk has that is causing the problem. via:talk will not confirm this but they must be using Asterisk as all the menus and such they have feel very Asteriskish. Is there something I can tell via:talk to try on their end to make this work? As a side symptem every time our system registers with via:talk it seams to jump from server to server on their end. They must have some sort of load balancing going on that is causing that. In the past we could get the DTMF to pass when we were on the initial server we registered with but when we got pushed to another server the DTMF would fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = rfc2833 relaxdtmf = yes rfc2833compensate = yes port = 5060 canreinvite = no fromdomain = galvatron.vtnoc.net disallow = all allow = ulaw,gsm If you need to see more of the setup info I can provide. Thanks Brent ___ -- 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] AEL: how to check if variable is defined
Dave Fullerton schrieb:
Philipp Kempgen wrote:
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what EXISTS() is for?
Well, yes. :-)
I've never needed it before and it didn't come to my mind.
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
--
___
-- 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] AEL: how to check if variable is defined
Richard Lyman schrieb:
Philipp Kempgen wrote:
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what ISNULL is for?
No. ISNULL() works on values not on variables.
But Dave Fullerton found EXISTS():
http://lists.digium.com/pipermail/asterisk-users/2008-December/224059.html
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
--
___
-- 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] 1.6, CDR and h extension
Tilghman Lesher tilgh...@mail.jeffandtilghman.com writes:
The only thing that occurs to me is that you might be using ForkCDR on your
production machine, in which case, you might want:
Set(CDR(hangupcause,r)=${HANGUPCAUSE})
for recursive setting of the hangupcause.
No ForkCDR, unfortunately. There appears to be no rhyme or reason to
which calls get hangupcause recorded and which don't.
I am going to try 1.6.1 beta 4, but I was having a bit of fun with
configure. It's compiling now.
/Benny
___
-- 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] Audiocodes MP-11X configuration to work with Asterisk
AudioCodes blatantly violates the terms of the GPL by not distributing the source code even after requesting it. Please don't use their hardware. On Thu, Jul 24, 2008 at 07:34, Frank Tarczynski ft...@mindspring.com wrote: I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It registers fine and I can call between the MP-114 and other extensions, but I'm not having much luck with the FXO ports. syslog shows the problem to be in the MP-114 configuration. Can anyone help? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] DTMF does not work
On Mon, Dec 29, 2008 at 1:55 PM, Brent Vrieze bvri...@cimsoftware.comwrote: I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both set to register with the same account right now. I shut Asterisk down on the one and now we don't drop any longer. doh!!! Last, We are having DTMF problems with our provider (via:talk). Does anyone have any experience with them and if so can you share it? via:talk does have a sample sip.conf and extensions.conf file to use but the dial plan they set up does not require any DTMF so they may never have tested it. We have tried inband, auto, rfc2833 for our DTMF and nothing works. I have submitted a ticket with them but the last time I did that they never responded so that is why I am posting here. I signed up with another SIP provider for a test account and the DTMF passes no problem from them so I must conclude there is some setting that via:talk has that is causing the problem. via:talk will not confirm this but they must be using Asterisk as all the menus and such they have feel very Asteriskish. Is there something I can tell via:talk to try on their end to make this work? As a side symptem every time our system registers with via:talk it seams to jump from server to server on their end. They must have some sort of load balancing going on that is causing that. In the past we could get the DTMF to pass when we were on the initial server we registered with but when we got pushed to another server the DTMF would fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = rfc2833 relaxdtmf = yes rfc2833compensate = yes port = 5060 canreinvite = no fromdomain = galvatron.vtnoc.net disallow = all allow = ulaw,gsm If you need to see more of the setup info I can provide. Thanks Brent I have the same problems with Viatalk. The problem is with their new servers. You are pointed to galvatron.vtnoc.net which is one of those. I currently have mine working by using their old servers. Try calling support, changing your account to rfc2833 if you haven't already and then point to chicago-1e.vtnoc.net with your same settings . You will have DTMF working, but I am not sure when the old servers are going away. Good Luck, Sean ___ -- 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] SIP host=dynamic help needed for CCME
Hi, I'm trying to get a remote Cisco Call Manager Express (CME) system behind a dynamic IP address routing both inbound and outbound calls via SIP to my local asterisk server. I've got a local CME system working fine on the LAN, where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't figure out how to get it working with host=dynamic, even locally on a test setup (to avoid NAT complications, etc...) Here's the local static one, which works fine: sip.conf: -- [general] context=default allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes ; [ccme-inbound] type=friend host=10.5.7.130 qualify=yes context=from-ccme allow=all insecure=port,invite canreinvite=no ; [ccme-outbound] type=friend host=10.5.7.130 qualify=yes context=from-ccme trustrpid=yes sendrpid=yes allow=all canreinvite=no dtmfmode=rfc2833 And, in CME: - dial-peer voice 200 voip session protocol sipv2 incoming called-number 211212 dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 101 voip description softphones 4-N destination-pattern 4[0-9] monitor probe icmp-ping session protocol sipv2 session target dns:sylvester.home.misty.com dtmf-relay rtp-nte codec g711ulaw no vad ! sip-ua no remote-party-id registrar dns:sylvester.home.misty.com expires 3600 secondary sip-server dns:sylvester.home.misty.com I think if I want to use host=dynamic in sip.conf on asterisk, I need to do something like this in CME: --- dial-peer voice 101 voip destination-pattern [1-2][0-9] session protocol sipv2 session target dns:sylvester.home.misty.com dtmf-relay rtp-nte codec g711ulaw no vad ! sip-ua authentication username foobar password 7 060F06233B583F4B00 realm NOTSURE registrar dns:sylvester.home.misty.com expires 3600 sip-server dns:sylvester.home.misty.com And, maybe for sip.conf, something like this: --- [foobar] type=friend context=from-ccme host=dynamic secret=notthis username=foobar dtmfmode=rfc2833 But, I'm really not getting far with this. There are tons of examples online of asterisk configurations to initiate connections to static hosts such as SIP providers, and CCME examples using static hosts, but I can't find anything like what I'm doing, even though it seems to me like a common kind of thing to set up. Any help would be greatly appreciated. Mark -- Mark G. Thomas (m...@misty.com) http://mail-cleaner.com/ ___ -- 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] AEL: how to check if variable is defined
Philipp Kempgen wrote:
Richard Lyman schrieb:
Philipp Kempgen wrote:
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what ISNULL is for?
No. ISNULL() works on values not on variables.
But Dave Fullerton found EXISTS():
http://lists.digium.com/pipermail/asterisk-users/2008-December/224059.html
Philipp Kempgen
if (${ISNULL(${CAMPAIGN})}) {
Set(CAMPAIGN=INBOUND);
};
This is how i use it.
___
-- 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] Dailplan code for holiday detection?
On Monday 29 December 2008 02:06:09 pm Adam Moffett wrote: This dialplan is illustrative of the particular problem of the MYSQL command in that no cleanup is performed if the dialplan terminates abnormally. If a device hangup occurs between the Connect and Disconnect, or worse, between the Query and the Clear, then extra resources will be consumed until a restart is performed. To avoid this problem, you should ensure that you always clear your query resources and disconnect your handles in the h extension. Or use func_odbc, which performs this sort of cleanup for you. Next question: When you say extra resources will be consumed until a restart is performed. Do you mean I have to restart asterisk to free up said resources? Will a reload do it also? Yes. No. -- Tilghman ___ -- 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] 1.6, CDR and h extension
Tilghman Lesher tilgh...@mail.jeffandtilghman.com writes:
The only thing that occurs to me is that you might be using ForkCDR on your
production machine, in which case, you might want:
Set(CDR(hangupcause,r)=${HANGUPCAUSE})
for recursive setting of the hangupcause.
You are onto something, even if I don't actually use ForkCDR. With
1.6.1beta4 the behaviour is consistent at least, for a particular call
scenario CDR(hangupcause) is either recorded correctly or not
recorded. Still, some call scenarios work and some don't.
In particular, some of the call scenarios create weird extra empty
CDR's like:
,,s,from-ipvision,SIP/ipvision-0871eaa02008-12-29
23:04:14,,2008-12-29 23:04:16,2,0,NO
ANSWER,DOCUMENTATION,,lpbx02-1230588254.29,,
[..] time passes [..]
Hey, I think I have it! The problem appears when using the g option
for the Dial command! With asterisk 1.6.1 beta 4, that closes the CDR
once the Dial is done, but the dial plan continues and doesn't get to
the h extension until later -- and by then the setting of
CDR(hangupcause) does nothing at all.
I'll have to make a proper bug report, but it's time for bed now.
/Benny
___
-- 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] Audiocodes MP-11X configuration to work with Asterisk
What does Audiocodes release under GPL? j On Mon, 29 Dec 2008, Andrew Joakimsen wrote: AudioCodes blatantly violates the terms of the GPL by not distributing the source code even after requesting it. Please don't use their hardware. On Thu, Jul 24, 2008 at 07:34, Frank Tarczynski ft...@mindspring.com wrote: I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It registers fine and I can call between the MP-114 and other extensions, but I'm not having much luck with the FXO ports. syslog shows the problem to be in the MP-114 configuration. Can anyone help? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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 ___ -- 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] Originate and fax detection
Cheers! You won't believe it. It actually detect fax quite nicely. How does AMD() do in a production environment? BR, Dex -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Sunday, December 28, 2008 12:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Originate and fax detection Perhaps AMD() supports this. Not necessarily reliably. On Dec 27, 2008, at 6:10 PM, Asterisk aster...@abraxas.si wrote: Hi everybody, I have an application that uses Originate AMI command to initiate outbound calls. However, I cannot find any way of redirecting calls that were originated to a fax machine to some other extension (e.g. fax extension). Is this possible? Or is there any way to get info from the AMI that the originated call went to a fax machine? BR, Dex ___ -- 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 ___ -- 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 ___ -- 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] Manager API
2008/12/29 Andrew Nowrot andrew.now...@gmail.com
I did not need to change the code. My manager.c already has all the lines
you specified that are wrong.
did you re compile and re installed?
make
make install
after the code change?
david
Cheers
___
-- 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
hi
sorry
the error wasnt in the line i told you before
is here, same file diferent function
static int action_originate(struct mansession *s, const struct message *m)
{
const char *name = astman_get_header(m, Channel);
const char *exten = astman_get_header(m, Exten);
const char *context = astman_get_header(m, Context);
const char *priority = astman_get_header(m, Priority);
const char *timeout = astman_get_header(m, Timeout);
const char *callerid = astman_get_header(m, CallerID);
const char *account = astman_get_header(m, Account);
const char *app = astman_get_header(m, Application);
const char *appdata = astman_get_header(m, Data);
const char *async = astman_get_header(m, Async);
const char *id = astman_get_header(m, ActionID);
const char *codecs = astman_get_header(m, Codecs);
struct ast_variable *vars = astman_get_variables(m);
char *tech, *data;
char *l = NULL, *n = NULL;
int pi = 0;
int res;
int to = 3;
int reason = 0;
char tmp[256];
char tmp2[256];
int format = AST_FORMAT_SLINEAR;
pthread_t th;
if (ast_strlen_zero(name)) {
astman_send_error(s, m, Channel not specified);
return 0;
}
--
(\__/)
(='.'=)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] Bug in contact header from Asterisk 1.6.0.3-rc1 ?
Hi all, I'm not sure wether it is a bug or not, so I'm asking for your opinion before submitting it to the bugtracker. The problem: I use asterisk with in sip.conf a non standard bind port of 5070 set. Now when asterisk sends out an Invite message to my sip proxy, the contact header in de request is something like: Contact: sip:12329...@123.123.123.123 The call succeeds and gets answered. So far so good. By using the 'Via' headers the 200 OK repsonse gets properly routed to asterisk. But now the client wants to end the call, and sends 'BYE sip:12329...@123.123.123.123'. Now the proxy can't route the messages by means of the Via header (because this is a new transaction? and Asterisk didn't insert a record-route header). The proxy forwards the 'Bye' to the default sip port on '123.123.123.123', with no success. The other way round, when the client initiates the call, asterisk answers with a '200 OK'. This response includes a correct 'Contact' header, consisting of both username,domain/ip ánd port. Can someone acknowledge my observations and conclusion is right? thanks, Egbert Groot. ___ -- 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] Audiocodes MP-11X configuration to work with Asterisk
On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote: What does Audiocodes release under GPL? j The MP-202 is running Linux. At first they said no it's not and later they admitted it did, but refused to supply the source code. Oddly enough, the Linux distribution is OpenRG, which itself had GPL problems a while back. I don't know about any other products, but I have never used them either. Of course, if they use GPL software they probably have the same attitude towards it. They shipped me the devices from their offices in Israel, so I could not just go to small claims court to get the code from them, I just gave up and never used their products again. Too bad, because the product was very nice. ___ -- 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] Audiocodes MP-11X configuration to work with Asterisk
On Mon, 29 Dec 2008, Andrew Joakimsen wrote: On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote: What does Audiocodes release under GPL? j The MP-202 is running Linux. At first they said no it's not and later they admitted it did, but refused to supply the source code. Oddly enough, the Linux distribution is OpenRG, which itself had GPL problems a while back. I don't know about any other products, but I have never used them either. Of course, if they use GPL software they probably have the same attitude towards it. They shipped me the devices from their offices in Israel, so I could not just go to small claims court to get the code from them, I just gave up and never used their products again. Too bad, because the product was very nice. This is interesting: http://www.audiocodes.com/bsd-bsds No mention of OpenRG... j ___ -- 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] Audiocodes MP-11X configuration to work with Asterisk
On Mon, 29 Dec 2008, Andrew Joakimsen wrote: On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote: What does Audiocodes release under GPL? j The MP-202 is running Linux. At first they said no it's not and later they admitted it did, but refused to supply the source code. Oddly enough, the Linux distribution is OpenRG, which itself had GPL problems a while back. I don't know about any other products, but I have never used them either. Of course, if they use GPL software they probably have the same attitude towards it. They shipped me the devices from their offices in Israel, so I could not just go to small claims court to get the code from them, I just gave up and never used their products again. Too bad, because the product was very nice. Oops - I take it back: http://www.audiocodes.com/gpl-lgpl Looks like they are at least attempting to comply... did you follow these steps? j ___ -- 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] connect a LAN server to a WAN server as a sip client
hello
I have a asterisk server on WAN, and one server on LAN. And now there are
some users on the LAN server. Users on the LAN server need to make phone
calls to PSTN through the WAN server.
Now on WAN server, there is a user account for the LAN server to register,
and it's an account with multiple ports.(So more than one users on the LAN
server can make phone calls through the WAN server at the same time)
Here are my config files of the LAN server:
sip.conf:
[general]
register=121022:123...@122.102.5.42 121022%3a123...@122.102.5.42
[121022]
type=peer
username=121022
secret=123456
host=122.102.6.43
fromuser=121022
canrenvite=no
insecure=very
quality=yes
nat=yes
context=
extensions.conf:
[]
exten=_013X.,1,Dial(SIP/${ext...@121022)
exten=_010,1,Dial(SIP/${ext...@121022)
exten=_X.,1,Dial(SIP/${EXTEN},60)
The good news is that user on the LAN server can make call throught the WAN
server; the problem is there can be only one user make phone calls at one
time. Can anyone tell me how to fix it? thanks
___
-- 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] AEL: how to check if variable is defined
Thanks Dave, Philipp and Richard!
klaus
Richard Lyman wrote:
Philipp Kempgen wrote:
Richard Lyman schrieb:
Philipp Kempgen wrote:
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what ISNULL is for?
No. ISNULL() works on values not on variables.
But Dave Fullerton found EXISTS():
http://lists.digium.com/pipermail/asterisk-users/2008-December/224059.html
Philipp Kempgen
if (${ISNULL(${CAMPAIGN})}) {
Set(CAMPAIGN=INBOUND);
};
This is how i use it.
___
-- 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
___
-- 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] 1.6, CDR and h extension
On Mon, Dec 29, 2008 at 10:19 PM, Benny Amorsen benny+use...@amorsen.dk wrote: You are onto something, even if I don't actually use ForkCDR. With 1.6.1beta4 the behaviour is consistent at least, for a particular call scenario CDR(hangupcause) is either recorded correctly or not recorded. Still, some call scenarios work and some don't. In particular, some of the call scenarios create weird extra empty CDR's like: ,,s,from-ipvision,SIP/ipvision-0871eaa02008-12-29 23:04:14,,2008-12-29 23:04:16,2,0,NO ANSWER,DOCUMENTATION,,lpbx02-1230588254.29,, Hi Benny, There was some tweaking done to the CDRs in Asterisk recently by murf that I believe did influence the behaviour of the hangup extension so that may be the cause of your first issue. The CDR you have posted above is not that abnormal, it looks like an outgoing call that did not end up getting sent anywhere meaningful and therefore its end destination was the s extension in your dialplan. Or it could be something else entirely. There is a currently a year long conversation going on about Asterisk CDRs to try and design out numerous glitches you can search the list archives if interested. Regards, Greyman. ___ -- 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] Newbie Polycom: Cannot conference with 10 digit 3rd party
Calling all Polycom gurus: I am using Polycom IP601 phones with Asterisk 1.4.21.2 In all Polycom phones, I set the following in sip.cfg. dialplan dialplan.impossibleMatchHandling=2 /dialplan (I leave the digitmap unchanged because I thought setting impossibleMatchHandling will ignore the bitmap) ...so that I could dial any number by entering a variable-size telephone number and then hit the send or dial key. This works quite well except when I am doing conferencing. It goes like this: I dialled the 1st party and was answered. Then I press conf key and then enter the 3rd party. I can keep entering until it reaches the 10th digit and then the 10-digit number is automatically dialled. Any thoughts? ___ -- 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
