thanks all
i found the telco only send me the normal number 87654321
i just want to start a fax service and people can direct dial some extend
num like 87654321...but it never send to me ... so the only thing i can do
is to provide a ivr and let the people enter the extend num then
In my previous reply, I may be wrong, "877" is probably a valid toll free
NPA, add it in the mix.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Thursday, March 12, 2009 7:45 PM
To:
I am not sure of the complexity of your issue.
There are only 4 toll free NPAs. 800, 888, 877, 866.
Any thing starting with them is toll free. So, filtering them is not a major
task, it would take only 4 statements at most and could be done with 2.
I don't program on a daily basis... so this is
I've finally got it working on 1.6.0.6, but it seems to be a problem:
Situation:
Queue realtime musiconhold class = prueba
I have default class on musiconhold.conf
When a call is made to the Queue checks that is not on memory so goes to the
db, I had 2 situations:
1) If digit
The patch doesn't work for me. Here's what I did:
Changed to my asterisk-1.4.23.1 directory
Executed the wget / patch command from the link you provided
> make
>> saw that res_features.so was recompiled
Moved /usr/lib/asterisk/modules/res_features.so to res_features.so.old
> make install
Confirmed
I posted this before, but it didn't show up. So if it's a dup...
I'm setting up dialplans to deal with 800 dialing through a different
channel than regular long distance in the US.
The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten
David Ruggles wrote:
> Wow!
>
> Thanks! That's a very clear answer and completely understandable. Is this
> something I should open a bug report on?
>
> Thanks,
Nope, I've already got that taken care of.
http://bugs.digium.com/view.php?id=14657
There's a patch there that I have tested and "it
Wow!
Thanks! That's a very clear answer and completely understandable. Is this
something I should open a bug report on?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 da...@safedatausa.com
-Original Message-
From: asterisk-users-bou
What version of * are you running?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Sebastian
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Thu, 12 Mar 2009 18:41:33 -0300
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subj
David Ruggles wrote:
> I don't really think that's a problem, because I'm able to use the other
> built in options: *1 to record; ## transfer (I changed this from a single
> pound) and there have been a couple times that I wouldn't hit them quickly
> enough.
>
> Thanks,
Ah, sorry about that. The
There was already lots of discussion, e.g. google for
asterisk monitor nfs
or
asterisk monitor ramdisk
regards
klaus
Tarek Sawah wrote:
> Hello,
> a local prison contacted us regarding some calling card solution.
> they need 4 E1s to serve 120 rooms in that prison.
> we are planning on usin
Darrin Henshaw wrote:
> Hello,
>
>
>
> We had an incident recently where a call was in queue for an extended
> period of time. We use queuemetrics for reporting, and it reports that
> the call was waiting for 20 minutes. The different thing about it is
> that the disconnect reason is stated
Hello,
a local prison contacted us regarding some calling card solution.
they need 4 E1s to serve 120 rooms in that prison.
we are planning on using 4 servers to serve the calls and one for the database
servers' specifications are:
2.8 Dual Core Proccessors
2 GB Ram
160 Sata Drive
each server wi
Is there any AMI action that logs a realtime agent?
I mean, if you send it, queue_log and queue_member get the corresponding
inserts.
Regards
Sebastian
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asterisk-users
I don't really think that's a problem, because I'm able to use the other
built in options: *1 to record; ## transfer (I changed this from a single
pound) and there have been a couple times that I wouldn't hit them quickly
enough.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer
No, I'm not using ForokCDR, I just have a bridged call and try to set the
CDR(userfield) and others on the channels, I look at the channes and I can
see the new value. But when I hangup the cdr record does not contain these
value, if I set it from dialplan works ok, but from cli or from ami is not
Hello,
We had an incident recently where a call was in queue for an extended period of
time. We use queuemetrics for reporting, and it reports that the call was
waiting for 20 minutes. The different thing about it is that the disconnect
reason is stated as Timeout. Is there a set maximum time a
On Thursday 12 March 2009 15:34:20 Sebastian wrote:
> I'm using 1.6.0.5 I'm trying to set CDR(userfield) for example from the
> cli, I look at the channel variables and I can see the new status, but que
> it hang-ups the CDR doesn't have this value.
Are you using ForkCDR()? Only a single record w
In version 1.6.0.x if you enable dialplan events in AMI you then get a
packet for each dialplan step executed. You should be able to capture that
data and generate records in a database.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
> From: nik600
> Reply-To: Asterisk U
On Thu, Mar 12, 2009 at 9:22 PM, BJ Weschke wrote:
>
> We generated a patch for a client probably about a year ago against the
> 1.4 branch that logged apps for each call, params, and exit status codes
> into a separate file. Like others have said, it generates a tremendous
> amount of data and p
Hi,
I'm using 1.6.0.5 I'm trying to set CDR(userfield) for example from the cli,
I look at the channel variables and I can see the new status, but que it
hang-ups the CDR doesn't have this value.
I'm using mysql backend for cdr
Any idea?
Thnks
__
Okay, then why do I still have error "unable to create channel of type "DAHDI"
(cause 66 - Channel not implemented) after typing dahdi restart; after typing
dahdi show staus and I see the card?
> Date: Thu, 12 Mar 2009 19:59:47 +
> From: j...@jeff.net
> To: asterisk-users@lists.digium.c
nik600 wrote:
> Hi to all.
>
> What can i do if a customer needs to log in the CDR all the dialpan
> actions related to a call?
> I mean, not only the lastapp e the lastdata but all the dialpan actions!
>
> I know that the actual CDR system store one record for each call (and
> for billing purposes
On Thu, 12 Mar 2009, Aqua Man wrote:
>
> After typing ps aux | grep asterisk I noticed there are five instances of
> asterisk running
>
> asterisk 4493 0.0 0.6 23720 3540 ? S15:12 0:00
> /usr/sbin/apache2 -k start
> asterisk 4494 0.0 0.6 23720 3540 ? S15
On Thu, Mar 12, 2009 at 8:44 PM, Steve Murphy wrote:
> My current thinking
> is to specify exactly which app invocations you want to track; those
> involved
> with dialing would be automatically tracked. Or time groups of invocations
> via
> forcing a leg-split via a simple dialplan application
On Thu, 12 Mar 2009, nik600 wrote:
> On Thu, Mar 12, 2009 at 8:13 PM, Matt Riddell wrote:
>> On 13/03/2009 8:02 a.m., nik600 wrote:
>>> Hi to all.
>>>
>>> What can i do if a customer needs to log in the CDR all the dialpan
>>> actions related to a call? I mean, not only the lastapp e the lastdat
2009/3/12 Danny Nicholas
> Greetings Listers,
>
> I’m running 1.4.21.2 on SUSE 11.0 with and zaptel
> 1.4.12.1 on a TDM400P. Most of my calls work great, but occasionally we try
> to connect to a customer or vendor external conference call and the call
> will drop afte
After typing ps aux | grep asterisk I noticed there are five instances of
asterisk running
asterisk 4493 0.0 0.6 23720 3540 ? S15:12 0:00
/usr/sbin/apache2 -k start
asterisk 4494 0.0 0.6 23720 3540 ? S15:12 0:00
/usr/sbin/apache2 -k start
aster
On Thu, Mar 12, 2009 at 1:13 PM, Matt Riddell wrote:
> On 13/03/2009 8:02 a.m., nik600 wrote:
> > Hi to all.
> >
> > What can i do if a customer needs to log in the CDR all the dialpan
> > actions related to a call?
> > I mean, not only the lastapp e the lastdata but all the dialpan actions!
> >
2009/3/12 ssmax
> Hi all
>
>i have just set up a asterisk in china, using DE410P and one E1 line
>and get a phone number like: +86 020 87654321 from my sp
>when somebody dial +86 020 87654321 , the asterisk will get the call in
> number by ${EXTEN} variable, but it can only get 87654
On Thu, Mar 12, 2009 at 8:13 PM, Matt Riddell wrote:
> On 13/03/2009 8:02 a.m., nik600 wrote:
>> Hi to all.
>>
>> What can i do if a customer needs to log in the CDR all the dialpan
>> actions related to a call?
>> I mean, not only the lastapp e the lastdata but all the dialpan actions!
>>
>> I kn
On 13/03/2009 8:02 a.m., nik600 wrote:
> Hi to all.
>
> What can i do if a customer needs to log in the CDR all the dialpan
> actions related to a call?
> I mean, not only the lastapp e the lastdata but all the dialpan actions!
>
> I know that the actual CDR system store one record for each call (a
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of nik600
Sent: Thursday, March 12, 2009 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] log to cdr each dialpan action
Hi Geriant,
My apologies for the delay in reply. We won't be using php but Perl and
there is an AGI module for perl Asterisk::AGI. I may be using Manager API
for sending Hangup signal. Im planning to write a bash script which perl
invokes when hangup button is pressed in the web interface. Bash sc
Hi to all.
What can i do if a customer needs to log in the CDR all the dialpan
actions related to a call?
I mean, not only the lastapp e the lastdata but all the dialpan actions!
I know that the actual CDR system store one record for each call (and
for billing purposes this can be correct) but in
On 13/03/2009 7:01 a.m., Simon P. Ditner wrote:
> I'm looking for some software to do emulation of phones so that I can
> test out a whole range of phones and their firmware revisions against
> asterisk. Anyone know of something like that?
>
> I'm hoping that the hardware vendors have something lik
Jimmy Godbout wrote:
> ssmax,
>
> Use CALLERID(num) to get the number that was dialed.
CALLERID(num) is the calling number, not the called number
klaus
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asterisk-users mailing
ssmax wrote:
> Hi all
>
> i have just set up a asterisk in china, using DE410P and one E1 line
> and get a phone number like: +86 020 87654321 from my sp when
> somebody dial +86 020 87654321 , the asterisk will get the call in
> number by ${EXTEN} variable, but it can only get 87654321, no area
2009/3/12 Tzafrir Cohen
> On Thu, Mar 12, 2009 at 10:18:22AM -0500, Mark Michelson wrote:
>
> >
> > Apparently bristuff has added new required parameters to call files.
>
> Rather: a new optional parameter. When you initiate a call from a
> channel it can also have either a 'message' or a 'pdo' m
On Thu, Mar 12, 2009 at 2:07 PM, Steve Davies wrote:
> 2009/3/12 Julian Lyndon-Smith :
> > Has anyone in the UK got ANI to work on an inbound call ?
> >
> > Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30
> >
> > Julian
> >
>
> Have you asked the Telco to send the ANI data? AFA
2009/3/12 Julian Lyndon-Smith :
> Has anyone in the UK got ANI to work on an inbound call ?
>
> Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30
>
> Julian
>
Have you asked the Telco to send the ANI data? AFAIK, this is disabled
by default on all BT lines. I assume they are able
David Ruggles wrote:
> I'm trying to actually use the example application map in features.conf:
>
> testfeature => #9,peer,Playback,tt-monkeys ;Allow both the caller and
> callee to play
> ;tt-monkeys to the opposite
> channel
>
> I see the feature get
I'm looking for some software to do emulation of phones so that I can
test out a whole range of phones and their firmware revisions against
asterisk. Anyone know of something like that?
I'm hoping that the hardware vendors have something like that, and
_maybe_ I'm really lucky and they are usin
On Thu, Mar 12, 2009 at 10:18:22AM -0500, Mark Michelson wrote:
>
> Apparently bristuff has added new required parameters to call files.
Rather: a new optional parameter. When you initiate a call from a
channel it can also have either a 'message' or a 'pdo' message.
Those are then passed alon
Thanks for the responses.
I have solved the problem by using a different tiff generator. I used the gs
command:
# gs -q -sDEVICE=tiffg3 -dSAFER -dNOPAUSE -sOutputFile=test.tif test.pdf
Best regards,
Santi
On Thu, Mar 12, 2009 at 3:30 PM, David Backeberg wrote:
> On Wed, Mar 11, 2009 at 7:32
I'm trying to actually use the example application map in features.conf:
testfeature => #9,peer,Playback,tt-monkeys ;Allow both the caller and
callee to play
;tt-monkeys to the opposite
channel
I see the feature get registered at the CLI:
== Registe
Hi, I am in a predicament and any help/pointers would be appreciated.
We are using chanspy to listen in on conversations. We are doing this via a
web interface. The web interface lists all the ongoing calls. We click on a
call and then my local phone rings allowing me to spy on the session I
click
Steve wrote:
Speaking from T1 PRI and E1 PRI in West Africa, you tell the telco how many
digits to send. Often times, at least in my experience, if not specified, they
will only send the last four providing there are no conflicts.
They should be able to send however many digits you require, but
I'm setting up dialplans to deal with 800 dialing through a different
channel than regular long distance in the US.
The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten statement. But it's
a pain dealing with all the 8xx area codes th
All of this really depends on the signaling.
You did not specify whether E1 was ISDN, CAS, etc.
--
Sent from mobile device
On Mar 12, 2009, at 11:18 AM, Steve Totaro
wrote:
Speaking from T1 PRI and E1 PRI in West Africa, you tell the telco
how many digits to send. Often times, at least
Speaking from T1 PRI and E1 PRI in West Africa, you tell the telco how many
digits to send. Often times, at least in my experience, if not specified,
they will only send the last four providing there are no conflicts.
They should be able to send however many digits you require, but maybe they
won
Greetings Listers,
I'm running 1.4.21.2 on SUSE 11.0 with and zaptel
1.4.12.1 on a TDM400P. Most of my calls work great, but occasionally we try
to connect to a customer or vendor external conference call and the call
will drop after 60-65 seconds unless I have an Answer
Peer Oliver Schmidt wrote:
> Olivier wrote:
>>> Do you by chance use bristuff?
>> Yes, I do.
>
> bristuff patches pbx/pbx_spool.c
>
> I have no knowledge of C, but there seems to be a problem around line
> 266.
>
> The original line (pre-bristuff) looks like this:
>
> if (ast_strlen_zero(o
Gavin Henry wrote:
> Hi All,
>
> We've got msidn configured:
>
> Port 1: TE-mode BRI S/T interface line (for phone lines)
> -> Protocol: DSS1 (Euro ISDN)
> -> childcnt: 2
>
I don't know if it depends on the card, but in my case I need to set the
termination jumper on TE mode for lin
Sorry, I indicated the wrong variable.
You can always ask your provider what is sent.
> -Original Message-
> From: ss...@126.com
> Sent: Thu, 12 Mar 2009 22:11:32 +0800
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Is it possible to get full callin number
> fromE1?
On Wed, Mar 11, 2009 at 7:32 AM, Santiago Gimeno
wrote:
> I finally solved the issue by changing the resolution and the width of the
> TIFF file to one that is accepted by the fax standard. In my case I changed
> to a resolution of 96x96 and a width of 1728.
>
> Now I am able to send faxes, but so
hi,Jimmy Godbout
when +86 136 make a call to ->
+86 020 87654321 -> asterisk
the CALLERID(num) will show the caller number +86136
the ${EXTEN} is the dialed number 87654321
i will try the CALLERID(dnid) tomorrow, will this get the whole dialed number
87654321 ?
i don
Hi Gavin,
if you can make and receive calls it works...do not worry if your line
is shown as DOWN, some telco turns it off but it works without problem.
Remember to ask your telco for the right signalling and set it the right
way (PTP or PMP).
Giorgio Incantalupo
Gavin Henry wrote:
> Hi All,
>
answering myself ...
Klaus Darilion schrieb:
> Hi!
>
> AFAIS the incoming SUBSCRIBE is handled in the same context as INVITE -
This is a bug which is fixed in Asterisk 1.4 branch, thus probably will
be fixed in 1.4.24
regards
klaus
> but how should I handle the SUBSCRIBE in the context?
>
Hi All,
We've got msidn configured:
Port 1: TE-mode BRI S/T interface line (for phone lines)
-> Protocol: DSS1 (Euro ISDN)
-> childcnt: 2
mISDN_close: fid(3) isize(131072) inbuf(0x8fd5060) irp(0x8fd5060)
iend(0x8fd5060)
and running on Asterisk 1.4.21.2:
pbx*CLI> misdn show stacks
ssmax,
Use CALLERID(num) to get the number that was dialed.
Jimmy
> -Original Message-
> From: ss...@126.com
> Sent: Thu, 12 Mar 2009 20:40:59 +0800
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Is it possible to get full callin number from
> E1?
>
> Hi all
>
>
Hi all
i have just set up a asterisk in china, using DE410P and one E1 line
and get a phone number like: +86 020 87654321 from my sp
when somebody dial +86 020 87654321 , the asterisk will get the call in
number by ${EXTEN} variable, but it can only get 87654321, no area code .
Olivier wrote:
>> Do you by chance use bristuff?
>
> Yes, I do.
bristuff patches pbx/pbx_spool.c
I have no knowledge of C, but there seems to be a problem around line
266.
The original line (pre-bristuff) looks like this:
if (ast_strlen_zero(o->tech) || ast_strlen_zero(o->dest) ||
(ast_s
Hi Steve,
I am not too much worried about the quality, but soon we will have 7000 users
with voicemail. We don't have accurate statistics on the voicemail usage but it
might well be that it will take a lot of space on the servers.
Ogg has been implemented in asterisk but it seems that nobody us
voip crazy wrote:
> Hello list,
>
> is working nicely. Now I need the fax to be print when arriving.
>
> ¿Anybody have this feature implementing in their systems?
>
We have several network printers mapped on the HylaFAX+ server and do
the following from the FaxDispatch script:
FILETYPE=tif;
2009/3/12 Peer Oliver Schmidt
> Hello Olivier,
>
> > With an extensions.ael enabled system, I keep getting whatever I change
> > into my "astup.call" file :
> >
> > [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At
> > least one of app or extension (or keyword message/pdu) must
Hi,
I thought of this but I don't know where to intercept the file to convert it.
The email is automatically sent, this is configured in voicemail.conf. I tried
to change the mailcmd with a custom script thinking I would get the parameters
passed to sendmail but it doesn't work. Has anybody an
Hello Olivier,
> With an extensions.ael enabled system, I keep getting whatever I change
> into my "astup.call" file :
>
> [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At
> least one of app or extension (or keyword message/pdu) must be
> specified, along with tech and dest
2009/3/12 voip crazy
> Hello list,
>
> I have an asterisk / hylafax / iaxmodem configured in one machine. All
> is working nicely. Now I need the fax to be print when arriving.
>
> ¿Anybody have this feature implementing in their systems?
>
> ¿How is the best way to get that?
>
> Any clue will be
2009/3/12 Tristan
> Hi,
>
> Send it to cups via the FaxDispatch script ;)
>
> Regards,
>
> Tristan
>
> voip crazy a écrit :
> > Hello list,
> >
> > I have an asterisk / hylafax / iaxmodem configured in one machine. All
> > is working nicely. Now I need the fax to be print when arriving.
> >
> > ¿
Gordon Henderson a écrit :
> Anyone here used these phones?
>
We stopped to use them, not stable IAX or SIP. A customer ask us to
exchange 26 pieces after 6 month battling to make the phones working. We
did it.
[...]
--
Daniel
___
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Hi,
Send it to cups via the FaxDispatch script ;)
Regards,
Tristan
voip crazy a écrit :
> Hello list,
>
> I have an asterisk / hylafax / iaxmodem configured in one machine. All
> is working nicely. Now I need the fax to be print when arriving.
>
> ¿Anybody have this feature implementing in thei
Gordon Henderson wrote:
> Anyone here used these phones?
>
> I'm getting more and more frustrated by todays modern crop of routers with
> their so-called SIP ALGs which are invariably broken, or routers with
> built-in ATAs which block internal SIP phones from working, so looking to
> use IAX f
Hello list,
I have an asterisk / hylafax / iaxmodem configured in one machine. All
is working nicely. Now I need the fax to be print when arriving.
¿Anybody have this feature implementing in their systems?
¿How is the best way to get that?
Any clue will be welcomed.
Thanks.
VoipCrazy
___
Has anyone in the UK got ANI to work on an inbound call ?
Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30
Julian
__
This email has been scanned by the MessageLabs Email Security System.
For more information pl
Hi
It's the same, *31# for Hungary. It may be common then.
Cheers
András
2009/3/12
>
> On 3/11/2009, "Hĺkan Källberg" wrote:
>
> >On Wed, Mar 11, 2009 at 04:16:43PM +0100, Christian Victor wrote:
> >> 2009/3/11 Hĺkan Källberg
> >> > Does anyone of you have Caller Presentation working in the
For a long time now we've used the astmanproxy process to handle manager
connections (75+ clients) so that these clients can tap into the
dialplan / send commands etc.
We use astmanproxy because at that time the manager connection routines
of asterisk did not cope well with numerous connections
hi
I had use more than 200 pcs of atcom model at530 phones with IAX and SIP
for more than 3 years whit no problems until now.
Gordon Henderson wrote:
> Anyone here used these phones?
>
> I'm getting more and more frustrated by todays modern crop of routers with
> their so-called SIP ALGs which
Anyone here used these phones?
I'm getting more and more frustrated by todays modern crop of routers with
their so-called SIP ALGs which are invariably broken, or routers with
built-in ATAs which block internal SIP phones from working, so looking to
use IAX for some end-users.
I already suppo
On Wed, 11 Mar 2009, Lutgring, Sam wrote:
> I have been using a number of the Grandstream GXP-2000 (74 in
> production), GXP-2010 (1 in production), and BT-200 (15 in production)
> with great success. The only issue that we have had is killing power
> supplies, not sure if this is related to o
On 3/11/2009, "Hĺkan Källberg" wrote:
>On Wed, Mar 11, 2009 at 04:16:43PM +0100, Christian Victor wrote:
>> 2009/3/11 Hĺkan Källberg
>> > Does anyone of you have Caller Presentation working in the other
>> > direction?? My mv370 is working well, execpt the Caller ID on outgoing
>> > GSM calls.
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