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.
--
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 shanava...@yahoo.com wrote:
From: Shanavaz E A shanava...@yahoo.com
Subject: [asterisk-users] CDR Unanswered calls
To: asterisk-users@lists.digium.com
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
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
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
--
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,
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
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:
On Fri, Oct 5, 2012 at 7:32 AM, sean darcy seandar...@gmail.com wrote:
I'm getting a parsing error with the folllowing:
same=n,GoSubIf($[${CALLERID(**num)} = 2024324321]?other,1(${**
thisexten}):)
WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
error: syntax error,
John Wolthuis wrote:
Hello All,
Hola,
I am trying to debug an odd issue. I have two UACs that are sending
INVITEs to my asterisk 1.8 server. I want to start authenticating
these incoming invite requests with digest auth. The UACs are not
registered and I am using host ip to match them
--- On Fri, 10/5/12, Vieri rentor...@yahoo.com wrote:
An Asterisk queue uses field names / config variables such
as:
announce-holdtime
However, documentation regarding realtime is very unclear.
voip-info.org suggests to use announce_holdtime.
Is this correct?
What about
Benoit Panizzon wrote:
Hello
Hola,
snipped out parts, check archives for those who are curious
Well for this case it is too late now. But is there a way to get the IP
Address of the SIP Client being logged in each CDR?
You can access the IP address of the received signaling traffic
I'm getting a parsing error with the folllowing:
same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1($
{thisexten}):)
WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
=
On 10/05/2012 02:10 PM, Benoit Panizzon wrote:
Hello
We had this situation:
Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk
Server was abused to call a large number of expensive destinations.
I'm sorry to hear that. In the Asterisk source there is a doc that
Jonas Kellens wrote:
Hello,
Hola,
I notice that the function ChanIsAvail always returns result : 0
It does not matter if the realtime SIP peer is registered or not.
How come ??
My dialplan :
exten = s,n,ChanIsAvail(SIP/${SIPPEERNAME})
exten = s,n,NoOp(availstatus = ${AVAILSTATUS})
If
Jonas Kellens wrote:
Hello,
I notice that the function ChanIsAvail always returns result : 0
It does not matter if the realtime SIP peer is registered or not.
How come ??
My dialplan :
exten = s,n,ChanIsAvail(SIP/${SIPPEERNAME})
exten = s,n,NoOp(availstatus = ${AVAILSTATUS})
${SIPPEERNAME}
On 05-10-12 14:45, Joshua Colp wrote:
Jonas Kellens wrote:
Hello,
I notice that the function ChanIsAvail always returns result : 0
It does not matter if the realtime SIP peer is registered or not.
How come ??
My dialplan :
exten = s,n,ChanIsAvail(SIP/${SIPPEERNAME})
exten =
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
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 solution?
-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
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:
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 talking a
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
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
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
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 passed to
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: ${SIPCHANINFO(recvip)},
On 05-10-12 15:27, Joshua Colp wrote:
Jonas Kellens wrote:
Using this will make Asterisk hang. Done that in the past and result was
that Asterisk hung after a certain amount of asterisk -rx command. So
my experience is that this is not the correct solution.
If only ChanIsAvail could return
Jonas Kellens wrote:
On 05-10-12 15:27, Joshua Colp wrote:
Jonas Kellens wrote:
Using this will make Asterisk hang. Done that in the past and result was
that Asterisk hung after a certain amount of asterisk -rx command. So
my experience is that this is not the correct solution.
If only
- Original Message -
From: Patrick Lists asterisk-l...@puzzled.xs4all.nl
To: asterisk-users@lists.digium.com
Sent: Friday, 5 October, 2012 11:46:48 AM
Subject: Re: [asterisk-users] LDAP Driver and VoiceMail
On 10/04/2012 10:00 PM, Phil Daws wrote:
Hello:
I am investigating the
Hi Joshua and all others who replied.
exten = _X.,1,Set(CDR(userfield)=${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
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
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
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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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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
On Fri, 2012-10-05 at 05:21 -0700, Vieri wrote:
--- On Fri, 10/5/12, Vieri rentor...@yahoo.com 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
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, 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, you
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
--
_
-- Bandwidth and
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
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.
On Fri, Oct 5, 2012 at 4:51 AM, Shanavaz E A shanava...@yahoo.com 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
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]:
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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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
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
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);
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
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
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