On Fri, Sep 11, 2009 at 4:18 AM, SIP s...@arcdiv.com wrote:
See... I would say the 'trapezoid' is one of the great strengths of SIP.
Which is why we need you all to come and discuss this, bringing up
other aspects, thoughts (even I have a few occasionally) and ideas.
@Dean - you will always
Hi,
documentation shows me:
Dial(Tech/User:passw...@host/Extension,Timeout,Optionen)
This is working for IAX2.
If Iam using
DIAL(SIP/u...@secret@sip.domian.tls/123456)
Asterisk shoes no host with name sip.domian.tls/123456
How to put in extension if using the DIAL command with userid and
My system is Debian
I have successfully installed Asterisk 1.6.1.6 to a non standart
folder /opt/asterisk
Then for asterisk-addons-1.6.1.1 I run:
./configure --prefix=/opt/asterisk/
make menuselect
And then command make fails with following errors:
Generating embedded module rules ...
make[1]:
As an ultra cheap way of doing it, you could simply output the caller-id to
a log file and display a tail 20 of it on a web page.
Something like this:
exten = s,1,System( echo${EPOCH}|${CALLERID(num)}
/var/log/asterisk/incoming )
It should be trivial to display the last n lines of it on a web
Thanks a lot for your help.
Since we decreased tx on the patton gateway (-3db iirc) , no more echo
report... so far. But this is a bit too early to conclude everything
is fine now.
Now I guess we would have had similar results with the pci card we
replaced by the gateway (Digium B410p with
Thank you Shaun for your answer!
Indeed, I have made some basic tests to convert a file to g729 using the
software codec and it works!
Have a nice day!
--
-- --
Marc LEURENT
lf...@leurent.eu
Le mercredi, 9 septembre 2009 19.32:30, Shaun Ruffell a écrit :
On 09/09/2009 09:33 AM, Marc Leurent
Is there any program Asterisk users use to calculate ASR and ACD ??
i calculate them from CDR, using php in a fancy webpage. But now as i think
of it, probably an sql query would do just fine.
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ABBAS SHAKEEL wrote:
But I cant generate calls using the loop back connector and get the
following error
*CLI [Sep 9 14:42:55] NOTICE[9981]: channel.c:3749
__ast_request_and_dial: Unable to request channel DAHDI/1/123
You cannot use a loopback connector for a PRI interface; PRI
HI all ,
I am using ,Dial(SIP/Gateway/${EXTEN},m)
how can i modify asterisk, if it detects two early media to stop OR MUTE
the first RTP early media AND let the user hear the second early media
any one developed something like that or know from where I can do this from
chan_sip.c?
Hello,
I may have found a serious issue with 1.6.1.6. I just compiled it
yesterday on our server. When anyone tries to access the name
directory through the Directory app, the asterisk process completely
dies.
Our extensions are in a realtime MySQL table, and the directory has
worked
Thanks you very much Kevin.I will try it by connecting one end of Ethernet
cable to one slot and other to second slot . Configuring one
as pri_net and the other as pri_cpe.
I will provide you feed on monday either i succed or not
On Fri, Sep 11, 2009 at 4:23 PM, Kevin P. Fleming
and how are those POTS lines connected to Asterisk?
In any event doing something like:
Set(CALLERID(num)=${CALLERID(num):0:10}) should do the trick.
On Tue, Sep 8, 2009 at 12:27 PM, Jeremy Taylor jer...@getwiredright.com wrote:
Hi,
I'm using asterisk 1.4.22-4 in Trixbox with snom 360 phones.
On Fri, 11 Sep 2009, Thomas Winter wrote:
documentation shows me:
Dial(Tech/User:passw...@host/Extension,Timeout,Optionen)
This is working for IAX2.
If Iam using
DIAL(SIP/u...@secret@sip.domian.tls/123456)
Asterisk shoes no host with name sip.domian.tls/123456
You need better
Change the client in voicemail.conf from sendmail to a MIME-enabled client
like pup.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gabriel Ortiz
Lour
Sent: Friday, September 11, 2009 9:36 AM
To: Asterisk Users Mailing
Hi all,
I'm trying to send an email with the voicemail details and I want to send
a HTML link on it to make a click2call to the voicemail main, but the email
is send with 'text/plain' encoding and thus it will not show the link, but
the HTML in plain text on the body of the email,
How can I
Hello Danny,
I've also the same question :-)
I've tried to find more information on the pup mime enabled program
but I haven't find something on the internet (every search refers to
puppy linux :-( )
Can you give more info ? Where can I find it ?
Thanks in advance
Patrick
On Fri, Sep 11, 2009
Hello,
I've migrated from CSV CDR to MySql CDR and the customization of my
CDR's is not working anymore.
Do you know if the cdr_mysql is supporting custom cdr's ? If not, is
there any alternative/workaround ?
Best regards,
Patrick
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Hello Team,
Can you help me on this? I have attached here the
queue logs of my asterisk. I've searching a parser for this. I do not
know what are the meaning of that logs. Thank you so much.
Your response is highly appreciated.
Regards,
Cristina
Maria Cristina Bayno escribió:
Hello Team,
Can you help me on this? I have attached here the queue logs of my
asterisk. I've searching a parser for this. I do not know what are the
meaning of that logs. Thank you so much.
Your response is highly appreciated.
Regards,
Cristina
Hi,
This
Hi Cristina,
You can find meanings in queuelog.txt (or queuelog.tex in * 1.6), it's
attached.
Daniel
On Fri, Sep 11, 2009 at 11:14 AM, Maria Cristina Bayno falls_m...@yahoo.com
wrote:
Hello Team,
Can you help me on this? I have attached here the queue logs of my
asterisk. I've searching a
I have internal mailboxes that I don't want visible to callers going through
the directory. Is it possible (in * 1.4) to hide mailboxes fom the
directory, without creating a new context?
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Use the 'hidefromdir' option in voicemail.conf for that particular
entry. This should be clearly documented in the example voicemail.conf
file.
Michelle Dupuis wrote:
I have internal mailboxes that I don't want visible to callers going
through the directory. Is it possible (in * 1.4) to
OK this is the RTFM question of the day but I need a sanity check.
I have 2 Astra 51i phone and a linksys PAP2 on a single DSL connection.
2 Aastra 51i-|
|-NAT on dsl moden--(Internet)--Asterisk
PAP2t|
The DSL modem/router which has QOS set for the
thanks for the reply..I will see it...
--- On Fri, 9/11/09, Daniel - Asterisk earohua...@gmail.com wrote:
From: Daniel - Asterisk earohua...@gmail.com
Subject: Re: [asterisk-users] Parser for Asterisk Queue Logs
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Friday 11 September 2009 11:05:36 am Patrick wrote:
I've migrated from CSV CDR to MySql CDR and the customization of my
CDR's is not working anymore.
Do you know if the cdr_mysql is supporting custom cdr's ? If not, is
there any alternative/workaround ?
This module doesn't support custom
Thank you Miguel.
Cristina
--- On Fri, 9/11/09, Miguel Molina mmol...@millenium.com.co wrote:
From: Miguel Molina mmol...@millenium.com.co
Subject: Re: [asterisk-users] Parser for Asterisk Queue Logs
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
On Friday 11 September 2009 11:05:36 am Patrick wrote:
I've migrated from CSV CDR to MySql CDR and the customization of my
CDR's is not working anymore.
Do you know if the cdr_mysql is supporting custom cdr's ? If not, is
there any alternative/workaround ?
On Fri, 11 Sep 2009, Tilghman
Greetings!
I have Asterisk 1.4.25.1 running as a Xen virtual machine using 64-bit
domU and Dom0 with faxing and was curious about other users' experiences
with it. So far receiving fax works just fine, but I'm curious if
anyone else is doing international faxing with asterisk. If yes, could
Any tip on where to find install doc on chan_mobile ?
http://www.chan_mobile.org
Thnak you!
HB
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now:
I don't do international faxing, but I'll share these tidbits:
Asterisk Faxing is a hit-or-miss proposition; either you're a genius or a
fool (sometimes both in the same day).
Faxing using POTS is a pretty simple proposition; SIP and T.38 add
complexity to the equation. Simplification of the
On Friday 11 September 2009 02:04:05 pm Steve Edwards wrote:
On Friday 11 September 2009 11:05:36 am Patrick wrote:
I've migrated from CSV CDR to MySql CDR and the customization of my
CDR's is not working anymore.
Do you know if the cdr_mysql is supporting custom cdr's ? If not, is
I am GETTING tired of CALLS getting stucked !!!
Any help ?
On Thu, Sep 10, 2009 at 2:52 AM, David @ULC ucoms2...@gmail.com wrote:
8186223080 : No entry in the log file !!!
On Thu, Sep 10, 2009 at 2:06 AM, David @ULC ucoms2...@gmail.com wrote:
Local/718186223...@d 718186223...@default
On Fri, 11 Sep 2009, Steve Edwards wrote:
I'm a 1.2 Luddite, but I found cdr_addon_mysql.c pretty easy to hack
on. For example, I added a channel variable named PRODUCT. Here's the
code I used:
channel_pointer = ast_get_channel_by_name_locked(channel);
product_pointer =
2009/9/11 ABBAS SHAKEEL shakeel.abbas@gmail.com
Thanks you very much Kevin.I will try it by connecting one end of
Ethernet cable to one slot and other to second slot . Configuring one
as pri_net and the other as pri_cpe.
I will provide you feed on monday either i succed or not
Hi,
Last night I was working on trying to get Fax for Asterisk installed.
I followed all the instructions, but noted that the download selector
only provider an X86 32 Bit binary.
The machine I'm using is:
2.6.26-1-amd64 #1 SMP Fri Mar 13 17:46:45 UTC 2009 x86_64 GNU/Linux
Consequently, on
Matt Riddell wrote:
Consequently, on load it presents an error:
[Sep 12 10:25:35] WARNING[31306]: loader.c:368 load_dynamic_module:
Error loading module 'res_fax.so': /usr/lib/asterisk/modules/res_fax.so:
wrong ELF class: ELFCLASS32
Is there any plan to provide a 64 bit binary for my
On Friday 11 September 2009 16:45:30 Steve Edwards wrote:
On Fri, 11 Sep 2009, Steve Edwards wrote:
I'm a 1.2 Luddite, but I found cdr_addon_mysql.c pretty easy to hack
on. For example, I added a channel variable named PRODUCT. Here's the
code I used:
channel_pointer =
On Thu, 2009-09-10 at 15:26 -0400, Kristian Kielhofner wrote:
On Wed, Sep 9, 2009 at 10:22 PM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
Hello, all. I've come across a nasty problem just as we are ready to
put our first system into production. During our final testing, we
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