Yes, Please see the following example.
In version 1.4 of asterisk, we used to get atleast 2 records in the CDR table
for one incoming call. One is the main record and second one is the record with
the status of that particular extension number which answered the call.
Additionally if any more ex
Thanks for the reply, I think I sorted things out.
> Is there some reason you need them to be different?
I have a remote sip system which sends traffic load balanced via two
redundant gateways. The remote system can't send a different auth
name based on which gateway its going out over. This me
On Fri, 5 Oct 2012, Jerry Geis wrote:
I place a call to a polycom phone, it answers, my AGI calls "Exec
SendDTMF 11 " but I do not hear the DTMF tones on the phone.
I'm just a 1.2 Luddite, but this works on my Polycom 501:
// senddtmf()
exec_agi("exec sipdtmfmode inband");
Perfect! Thank you.
Mitch
On 10/05/2012 01:07 PM, Ioan Indreias wrote:
Hi Mitch,
Glad that it works for you.
Regarding the CallerID I suggest to set some the variables before the
actual Dial.
Something like:
Action: Originate
Channel: Local/s@callmenow/n
Context: to-customer
Exten: s
Prior
Anybody know a good sip trunking provider for use in Hawaii? Big Island
Specifically. Need to move a client off a dozen pots lines.
Jared
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New to Asterisk?
So here's what I used:
$['x${CALLERID(num)}'='x2024324321']
And that worked!
On 10/05/2012 08:28 AM, Richard Kenner wrote:
I'm getting a parsing error with the folllowing:
same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1($
{thisexten}):)
WARNING[11356]: ast_expr2
On Fri, Oct 5, 2012 at 4:51 AM, Shanavaz E A wrote:
> Hi,
>
> No replies until now. Some one please help... There must be some people
> who are using it...
>
> Thanks
>
>
>
Can you provide an example of what you expect it to be doing (from the old
version) and what it is doing now (from the new v
On 10/06/2012 02:53 AM, Gabriel Ortiz Lour wrote:
Hi,
Does anyone had the problem of asterisk SendFax + spandsp sending
only the first page of a multi-page TIFF file?
Seams to be related to spandsp ECM config.
Any thoughts about it?
Thanks,
Gabriel
Check the file with tiffinfo. Perhap
It could be a coding issue in your TIFF file. I have successfully sent
multiple page TIFF's using plain POTS and DAHDI, but in 1.4 the sendfax
module was finicky.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gabriel Ortiz
Lour
Sent
Hi,
Does anyone had the problem of asterisk SendFax + spandsp sending only
the first page of a multi-page TIFF file?
Seams to be related to spandsp ECM config.
Any thoughts about it?
Thanks,
Gabriel
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-- Bandwidth and
On Fri, Oct 05, 2012 at 12:01:58PM -0500, Vladimir Mikhelson wrote:
> I do not seem to be able to compile DAHDI 2.6.1 against the new kernel
> as well.
>
> See https://issues.asterisk.org/jira/browse/ANOW-168 and
> https://issues.asterisk.org/jira/browse/DAHLIN-303
According to your build output,
Hi,
Did anybody upgrade the kernel to 2.6.18-308.16.1.el5 on CentOS 5.7?
If the answer is "Yes" did you run into issues with DAHDI 2.6.1?
I am observing the missing "kmod-dahdi-linux.i686
2.6.1-1_centos5.2.6.18_308.16.1.el5" in Digium depository.
I do not seem to be able to compile DAHDI 2.6.1
On Fri, 2012-10-05 at 05:21 -0700, Vieri wrote:
>
> --- On Fri, 10/5/12, Vieri wrote:
>
> > An Asterisk queue uses field names / config variables such
> > as:
> >
> > announce-holdtime
> >
> > However, documentation regarding realtime is very unclear.
> >
> > voip-info.org suggests to use ann
Jerry Geis wrote:
I place a call to a polycom phone, it answers, my AGI
calls "Exec SendDTMF 11 " but I do not hear the DTMF tones on the
phone.
Why is that?
Hola,
Some phones do not play the DTMF tones as it is generally not useful for
a human to hear them and they can be considered off
I place a call to a polycom phone, it answers, my AGI
calls "Exec SendDTMF 11 " but I do not hear the DTMF tones on the
phone.
Why is that?
Jerry
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This is mostly working. See below. My only problem is being able to
set the caller ID on the outbound call to the customer. I've tried both
a queue connected macro and gosub (see below), and those both execute,
but the caller ID is not showing up correctly for the customer. I
assume this is
Benoit Panizzon wrote:
Hi Joshua and all others who replied.
Hola,
exten => _X.,1,Set(CDR(userfield)=${CHANNEL(recvip)})
Thank you, that did it.
Glad to hear it!
It's an asterisk 1.6.2.9 actualy. Are additional CDR fields like CDR(recvip)
only possible from some newer release or do the
Hi Joshua and all others who replied.
> exten => _X.,1,Set(CDR(userfield)=${CHANNEL(recvip)})
Thank you, that did it.
It's an asterisk 1.6.2.9 actualy. Are additional CDR fields like CDR(recvip)
only possible from some newer release or do they have to be defined somewhere?
Well sure I now have
- Original Message -
From: "Patrick Lists"
To: asterisk-users@lists.digium.com
Sent: Friday, 5 October, 2012 11:46:48 AM
Subject: Re: [asterisk-users] LDAP Driver and VoiceMail
On 10/04/2012 10:00 PM, Phil Daws wrote:
> Hello:
>
> I am investigating the possibility of using LDAP f
Jonas Kellens wrote:
On 05-10-12 15:27, Joshua Colp wrote:
Jonas Kellens wrote:
Using this will make Asterisk hang. Done that in the past and result was
that Asterisk hung after a certain amount of asterisk -rx "command". So
my experience is that this is not the correct solution.
If only Cha
On 05-10-12 15:27, Joshua Colp wrote:
Jonas Kellens wrote:
Using this will make Asterisk hang. Done that in the past and result was
that Asterisk hung after a certain amount of asterisk -rx "command". So
my experience is that this is not the correct solution.
If only ChanIsAvail could return
Ishfaq is right, that's the way to go.
Here's a dialplan line to help you achieve that:
exten => YOUREXTEN_CHANGE_ME,PRIORITY_CHANGE_ME,Set(CDR(UserField)=SIP
HEADER CONTACT: ${SIP_HEADER(CONTACT)}, SIPURI: ${SIPURI}, SIP PEER IP:
${SIPCHANINFO(peerip)}, SIP RECEIVED IP: ${SIPCHANINFO(recvip)}, SI
On Fri, 2012-10-05 at 14:10 +0200, Benoit Panizzon wrote:
> Hello
>
> We had this situation:
>
> Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk
> Server was abused to call a large number of expensive destinations.
>
> It is clear that the sip logins have been passe
I'll give this a try today and post the results here.
Mitch
On 10/04/2012 02:30 PM, Ioan Indreias wrote:
Hello Mitch,
Hoping that the Queue application is not automatically Answering the
line (till an agent will do this) my suggestion is to switch between
"who have to answer" in order to pro
Jonas Kellens wrote:
Using this will make Asterisk hang. Done that in the past and result was
that Asterisk hung after a certain amount of asterisk -rx "command". So
my experience is that this is not the correct solution.
If only ChanIsAvail could return the correct value...
You may have miss
Hi
Has anyone experienced any issues with calls through asterisk server
with a netted phone connected to a Cisco 18XX series router.
I'm experiencing one way audio when the caller calls from the phone
connected to the asterisk server to the outside world (via a SIP
provider). It's the audio in to
Sorry for my last post,
> > Here is my IP-PBX setupmy setup is : sips softphone <-> asterisk <-> xorcom
> > PSTN gateway <-> pstn line to telcoi'm using xlite for windows
>
> > when I make a phone call (sip - outgoing channel),I can hear my own voice
> > so clear. it's very annoying mewhen
On 05-10-12 15:19, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Friday, October 05, 2012 7:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: R
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Friday, October 05, 2012 7:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AVAILSTATUS always 0
J
> > Here is my IP-PBX setupmy setup is : sips softphone <-> asterisk <-> xorcom
> > PSTN gateway <-> pstn line to telcoi'm using xlite for windows
>
> > when I make a phone call (sip - outgoing channel),I can hear my own voice
> > so clear. it's very annoying mewhen talking a little loud... any
Jonas Kellens wrote:
Hello,
I do not want to know if the remote side may or may not decline the
call, I just want to know if the SIP peer is registered or not. That is
information that Asterisk has without placing a call. Placing a call to
an unregistered peer would fail.
Indeed,
I just want
On 05-10-12 14:45, Joshua Colp wrote:
Jonas Kellens wrote:
Hello,
I notice that the function ChanIsAvail always returns result : 0
It does not matter if the realtime SIP peer is registered or not.
How come ??
My dialplan :
exten => s,n,ChanIsAvail(SIP/${SIPPEERNAME})
exten => s,n,NoOp(avail
Jonas Kellens wrote:
Hello,
I notice that the function ChanIsAvail always returns result : 0
It does not matter if the realtime SIP peer is registered or not.
How come ??
My dialplan :
exten => s,n,ChanIsAvail(SIP/${SIPPEERNAME})
exten => s,n,NoOp(availstatus = ${AVAILSTATUS})
${SIPPEERNAME
Jonas Kellens wrote:
Hello,
Hola,
I notice that the function ChanIsAvail always returns result : 0
It does not matter if the realtime SIP peer is registered or not.
How come ??
My dialplan :
exten => s,n,ChanIsAvail(SIP/${SIPPEERNAME})
exten => s,n,NoOp(availstatus = ${AVAILSTATUS})
If
On 10/05/2012 02:10 PM, Benoit Panizzon wrote:
Hello
We had this situation:
Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk
Server was abused to call a large number of expensive destinations.
I'm sorry to hear that. In the Asterisk source there is a doc that
focu
> I'm getting a parsing error with the folllowing:
>
> same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1($
> {thisexten}):)
>
> WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
> error: syntax error, unexpected '=', expecting $end; Input:
>
Benoit Panizzon wrote:
Hello
Hola,
Well for this case it is too late now. But is there a way to get the IP
Address of the SIP Client being logged in each CDR?
You can access the IP address of the received signaling traffic
(provided it has not been spoofed) using ${CHANNEL(recvip)} in th
--- On Fri, 10/5/12, Vieri wrote:
> An Asterisk queue uses field names / config variables such
> as:
>
> announce-holdtime
>
> However, documentation regarding realtime is very unclear.
>
> voip-info.org suggests to use announce_holdtime.
> Is this correct?
>
> What about monitor-type? Shou
John Wolthuis wrote:
Hello All,
Hola,
I am trying to debug an odd issue. I have two UACs that are sending
INVITEs to my asterisk 1.8 server. I want to start authenticating
these incoming invite requests with digest auth. The UACs are not
registered and I am using host ip to match them wit
On Fri, Oct 5, 2012 at 7:32 AM, sean darcy wrote:
> I'm getting a parsing error with the folllowing:
>
> same=n,GoSubIf($[${CALLERID(**num)} = 2024324321]?other,1(${**
> thisexten}):)
>
> WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
> error: syntax error, unexpected '=', e
Try defaultuser=test instead of username=test
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Wolthuis
Sent: Thursday, October 04, 2012 11:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Hello
We had this situation:
Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk
Server was abused to call a large number of expensive destinations.
It is clear that the sip logins have been passed to various persons (probably
posted on a forum somewhere inviting to do
Hi
An Asterisk queue uses field names / config variables such as:
announce-holdtime
However, documentation regarding realtime is very unclear.
voip-info.org suggests to use announce_holdtime.
Is this correct?
What about monitor-type? Should it be underscored too (monitor_type)?
Thanks,
Vieri
On 10/05/2012 11:51 AM, Shanavaz E A wrote:
Hi,
No replies until now. Some one please help... There must be some people
who are using it...
Thanks
No idea but since Asterisk is making you money why don't you hire an
experienced Asterisk consultant to get it resolved.
Regards,
Patrick
--
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On 10/04/2012 10:00 PM, Phil Daws wrote:
Hello:
I am investigating the possibility of using LDAP for storing certain Asterisk
configuration parameters.
I have examined res_ldap.conf and see where mailbox can be defined from
AstAccountMailbox but I do not see where the password can be stored ?
If the function ChanIsAvail does not work to check if a SIP peer is
registered or not, what function should I use then ??
Jonas.
On 04-10-12 17:05, Jonas Kellens wrote:
On 04-10-12 16:59, Danny Nicholas wrote:
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@l
Hi,
No replies until now. Some one please help... There must be some people who are
using it...
Thanks
--- On Mon, 9/24/12, Shanavaz E A wrote:
From: Shanavaz E A
Subject: [asterisk-users] CDR Unanswered calls
To: asterisk-users@lists.digium.com
Date: Monday, September 24, 2012, 11:40 AM
Hello,
what exactly does hangupcause 111 mean ?
I read on the wiki : 111 protocol error 500 Server internal error
Is the the SIP response that was received form the other end ? Or is
this an internal server (Asterisk) error ?
Kind regards,
Jonas.
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