use Asterisk CDR (Call Detail Record)
ref: http://areski.net/asterisk-stat-v2/about.php
Greetings!
Prompt how to make that the asterisk wrote down all calls automatically
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Hello, All!
It is a Piece of my log At the moment of a call:
Jan 9 09:50:21 NOTICE[10902]: chan_zap.c:8194 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 3
-- Accepting call from '' to '0033444' on channel 1/1, span 1
-- Executing Dial(Zap/1-1,
On Mon, 8 Jan 2007 20:03:50 -0500
Andrew Joakimsen [EMAIL PROTECTED] wrote:
Good luck dealing with Linksys on that
http://voxilla.com/tools/device-configuration-wizard/certificate-authority-service-for-linksys-analog-voip-adaptors-808.html
Hi Andrew!
Thanks for the response,
I have success register polycom in to asterisk and it can called by other
extension. But why it can't calling other extension ? and i have warning
from asterisk
chan.sip.c:3602 process_sdp: Unknown SDP media type in offer: application
49200 RTP/AVP 100
anyone undertand this warning ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mitcheloc
Sent: Sunday, December 31, 2006 8:52 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] WIFI SIP- The Best phone
Those wifi phones
Hi all,
I have an Asterisk server running, and some hardware phones, and I want to
do 3PCC : third party call control.
The third party is a software running on the asterisk box, which can for
example ask a hard SIP phone to put a call on hold. To do that, this
software has to send a SIP
Gregory,
I know there is something called SIP CTI TR87.
It's used by Nortel to integrate with Microsoft's Live Communication Server.
Don't know if something similar exists for Asterisk.
This links could be helpfull:
http://www.ecma-international.org/publications/techreports/E-TR-087.htm
Regards,
Seems that this has to be implemented by the phones, or by a B2BUA
I think that a B2BUA could be used for 3PCC, but dont know if an
open-source B2BUA exists and works with Asterisk
Greg
_
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Koen Van Impe
uhm...
On Tue, 2007-01-09 at 12:28 +0100, Gregory Duchatelet wrote:
Seems that this has to be implemented by the phones, or by a B2BUA…
I think that a B2BUA could be used for 3PCC, but don’t know if an
open-source B2BUA exists and works with Asterisk …
asterisk IS a B2BUA
just my 2cents.
On 09/01/07, Nigel Kendrick [EMAIL PROTECTED] wrote:
I've had a play with a Nokia E70 - the 'bar' version of the E61 and gave up!
Menu navigation is dire - I went through hoops trying to get SIP working - I
know from others it can be done, but I bailed out when I realised that to
put these
I have a bunch of 7910's that I managed to get registered with
Asterisk 1.2.14:
managed5*CLI skinny show devices
Name DeviceId IP TypeId R Model NL
--- -- - -- --
test7 SEP0004C1878F8E 192.168.0.226 6 Y 7910 1
The problem is that the
True :)
Here is an example of what i want to do :
- a phone call extension 100
- asterisk enter the context, and execute Dial() to call another phone
- ringing...
- now I want that asterisk ask the called phone to answer : how to do that
??
Greg
uhm...
On Tue, 2007-01-09 at 12:28 +0100,
is anyone using this to initiate a call back. . . I am epically interested
in AIM, as it can can serve as a free GSM gateway.any ideas?
On 1/8/07, Hall, Eric M. [EMAIL PROTECTED] wrote:
Has anyone got Asterisk IM to work
Using this link
http://www.sipalive.com/dev/asterisk/
And a clean
C F wrote:
I knew I was doing the right thing, here is the proof, enjoy when you
read it, and have a good laugh.
-- Forwarded message --
From: Al Bochter [EMAIL PROTECTED]
Date: Jan 8, 2007 8:22 PM
Subject: Re: [asterisk-users] Some queries on g729 license.
To: [EMAIL
Has anyone got this annoying sidecar to illuminate when users are on the
phone?
In my function key settings I have:
Context: Active
Type: Extension
Number: sip:[EMAIL PROTECTED];user=phone (4000 is the extension I
want to see/dial on the key).
I can press the key and it will dial the
Hi
On Tue, Jan 09, 2007 at 06:20:04AM -0800, Derek Whitten wrote:
[ unrelated message completely quoted snipped]
[ signatures snipped ]
[ offline message posted on-list snipped ]
[ foul language snipped ]
At least you didn't top-post.
and have a nice day
Thank you. Now could we please get
Hi,
I have made some headway with this. Let me explain a abit of the setup. I have an Orion GSM
Gateway, that was connected to a Cisco AS5300 via E1. When I looked at the AS5300 config, it was
talking R2 to the Orion. So I have tried to connect the Orion direclty to Asterisk (leaving out
Yusuf, there are several things can be wrong. Make sure you have
configured the correct protocol variant, DNIS and CID. Also check you
really need CRC4 checking.
I wrote a document to help debugging this stuff, you can find it here:
Hello,
I have just installed Asterisk 1.4 and I am playing with it. I've created some
sip accounts and some queues. When I start asterisk I see many queues like
this:
all-phones-r has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime),
W:0, C:0, A:0, SL:0.0% within 0s
Members:
There has been talk about it before and I think people have done it.
Paging Sam Tam
- Original Message -
From: Joao Pereira [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Sent: Tuesday, January 02,
Hello,
in our company we are trying to do this:
Fax -- Traditional PBX -- Asterisk -- PSTN
In practice, we have put an Asterisk equipped with a Sangoma A102 (2 PRI
ports) between our PBX (Siemens HiCom) and the PSTN in order to have a VoIP
network along the traditional telephony network.
The
Does somebody know a similar device that does the same for GSM networks ?
Zoa
Dovid B wrote:
There has been talk about it before and I think people have done it.
Paging Sam Tam
- Original Message - From: Joao Pereira [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Do you have the videosupport=yes in your sip.conf for that device? You might
try adding:
videosupport=yes
allow=h263 ; H.263 is our video codec
allow=h263p ; H.263p is the enhanced video codec
On 1/9/07, [EMAIL PROTECTED] [EMAIL PROTECTED]
wrote:
I have success register polycom in to
Has anyone heard of a build or instructions for installing Asterisk on a
Suse 10.1 system?
Bob Rawlinson
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I would like to connect an Asterik server to an Avaya IP Office IP406 and use
the * as an VoIP Gateway.
The IP Office has two Analog extensions available. I thought connecting this
analog extensions to 2 FXO ports in the * to interconnect the PBXs.
Is this possible? Does any one
This is by far the most volotile list I have ever been on. I'm not
sure that's exactly the reputation Digium/Asterisk is shooting for,
but even so it does provide some much needed comedy relief.
After seeing the G.729 pricing direct from SIPRO, I now take the
shut-up and be thankful position. I
David Thomas wrote:
This is by far the most volotile list I have ever been on. I'm not
sure that's exactly the reputation Digium/Asterisk is shooting for,
but even so it does provide some much needed comedy relief.
After seeing the G.729 pricing direct from SIPRO, I now take the
shut-up and
jeremij jerome wrote:
The problem is with the fax. We just want to send and receive faxes
from/to our fax machine connected to the Siemens (without needing any
interaction with our VoIP network, the faxes are sent to/received from
PSTN). Unfortunately we are experiencing a lot of problems:
My garage door is...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Crompton
Sent: Monday, December 04, 2006 11:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Is there any Asterisk controllable
you are asking about Shared line apperance or hints. Look at this
http://www.voip-info.org/wiki/view/snom+360
On 1/9/07, J. Oquendo [EMAIL PROTECTED] wrote:
Has anyone got this annoying sidecar to illuminate when users are on the
phone?
In my function key settings I have:
Context: Active
Hi,
I have two asterisk servers where softphone A is connected to asterisk A.
On those two asterisk servers, ooh323c is installed.
I tried to call a test context on asterisk B from softphone A. But I
always fall into context default of asterisk B.
( I don't know how to tell asterisk A
When you say build do you mean a plug and play binary? I use SUSE 7.3
here and it is easy to get the source files and compile it. It should
just work. The instructions would be in the README or INSTALL file in the
source.
1. Get the source at digium (the 1.2.x version might be better to start
Dear List,
My problem is that the incoming Caller Id is not displayed on the local analog
phones (connected to a TDM400 card).
I receive the CID correctly from my telco, but when I place the call to the
internal analog line, the CID is not propagated.
An interesting point: when I try to place
(C)harlie (F)oxtrot
Incredible! I didn't see it until just now. I use postini and it was in
the quarantine with a XXX icon as the reason it got filtered.
Let's set up a betting pool on him. :)
C F wrote:
I knew I was doing the right thing, here is the proof, enjoy when you
read it, and have
after the Dial has connected, I want the caller (on a SIP phone) to be
able to press keys in order to record call status. is this possible?
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
Incoming faxes, the Sangoma will detect the tones and disable echo cancel.
To send outbound, you will have to add another trunk group, of one or more
channels and disable echo cancellation and use that to dial out.
Example (/etc/asterisk/zapata.conf)
blah blah
echocancel=yes
blah blah
group
I've had the E70 for about a month.
The first few days were not fun.
But now that I've learned the gotchas and the workarounds, it is GREAT.
You -can- configure it, and asterisk, to work perfectly together, every time.
With automatic failover to conventional GSM phone behaviour if not in 802.11
housi mueller wrote:
I would like to connect an Asterik server to an Avaya IP Office IP406
and use the * as an VoIP Gateway.
The IP Office has two Analog extensions available. I thought connecting
this analog extensions to 2 FXO ports in the * to interconnect the PBX’s.
What sort of
Anyone out there using VOIP for business class inbound/outbound
services? I've found my VOIP provider to be less than reliable, SIP
registrations timeout, calls drop, they claim IAX2 is too buggy (I
find that hard to believe), and pretty much blame all problems on
other circumstances and don't
I am using a extension to dial the console which has autoanswer
enabled. I am getting a strange warning, has anyone seen this before?
Nothing on Google, or Voip-Info
[Jan 9 13:50:05] WARNING[5009]: chan_oss.c:1048 oss_request:
oss_request ty console data 0x0xb7851e00 dsp
Call to device 'dsp'
I did a search:
http://www.google.com/search?q=voip+quality+%28test+OR+testing%29+asterisk-users+site%3Amail-archive.com
and found this:
http://www.testyourvoip.com/
This seems to have quite a bit of detail.
Does anyone have a better solution for testing
VOIP quality?
Comments?
The main goal is that any extension from the Avaya PBX can make long distance
calls using the asterisk server as VoIP gateway (using a SIP Provider).
It would be also great if from a remote IP Phone (in an other location), a
user could use the Asterisk server to dial in and the * forwards
Hi O.Youssef,
if you asterisk version is 1.2.X
edit apps/Makefile
and discomment the line that contain 'app_sql_postgres.so':
#
# Obsolete things...
#
APPS+=app_sql_postgres.so
#APPS+=app_sql_odbc.so
save
if you use debian:
aptitude install libpq-dev
and compile again
I hope this be
Andrew Latham wrote:
you are asking about Shared line apperance or hints. Look at this
http://www.voip-info.org/wiki/view/snom+360
Been there done that page. Nothing worth noting in there.
--
J. Oquendo
J. Oquendo wrote:
Andrew Latham wrote:
you are asking about Shared line apperance or hints. Look at this
http://www.voip-info.org/wiki/view/snom+360
Been there done that page. Nothing worth noting in there.
Do the line appearances work on the 12 non-sidecar buttons?
- Mike
Just done this for a client using an E1 Pri card in the avaya box and a
sangoma a102, using qsig , works fine, I wouls recommend this to any
oneits been up and stable for two months now
Regards
Robb
housi mueller wrote:
The main goal is that any extension from the Avaya PBX can make long
Dr. Michael J. Chudobiak wrote:
J. Oquendo wrote:
Andrew Latham wrote:
you are asking about Shared line apperance or hints. Look at this
http://www.voip-info.org/wiki/view/snom+360
Been there done that page. Nothing worth noting in there.
Do the line appearances work on the 12 non-sidecar
always include a wait before a dial
give the callerid time to get into * before dialing, it arrives
between the first and second ring, if you have * dial after the first
ring it will not be there yet to pass along
On Jan 9, 2007, at 12:16 PM, Anton Frolov wrote:
Dear List,
My problem
I am siting in a building with 30 Snom 360s and 25 sidecars, I can
assure you that it can work. Check you Snom Firmware, settings on the
extra lines (you set them as shared).
I should update the wiki someday, been a while...
On 1/9/07, J. Oquendo [EMAIL PROTECTED] wrote:
Dr. Michael J.
All - this is probably a simple problem, but I've been pulling my hair
out trying to figure out what I'm doing wrong. I'm building a
*simple* IVR menu. Here it is:
[main-menu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(30)
Eric ManxPower Wieling wrote:
Chris Miller wrote:
I would tend to agree, but the context that holds these number is an
inbound context which includes additional logic that would fail
normal calls. Yes, I can add the DIDs to the outbound context, but
the point here is not to have a bloated
From: Jerry Jones [EMAIL PROTECTED]
always include a wait before a dial
give the callerid time to get into * before dialing, it arrives between
the first and second ring, if you have * dial after the first ring it will
not be there yet to pass along
Is there a way to count number of
I don't really know the name of what I want to look for but maybe
someone could tell me if it would be available.
I have a number of old analogue cell phones laying about here and I was
thinking it would be useful if I could set up a short range base station
for them that would cover maybe an
On Tue, Jan 09, 2007 at 05:11:55PM -0500, M.Hockings wrote:
I don't really know the name of what I want to look for but maybe
someone could tell me if it would be available.
I have a number of old analogue cell phones laying about here and I was
thinking it would be useful if I could set up
thanks, Jerry
but I don't thinks it's a problem, since I correctly get the CID from external
line (moreover, I do some lookup of the received number in my LDAP database and
making some decisions based on it).
So when I call the Dial function, the CID is present in asterisk for sure.
AF.
Jerry
This is a reminder that the Twin Cities Asterisk Users Group will be
meeting this Saturday, January 13 at 11:30am. - This month's meeting will
focused on IP Telephony (VoIP) and network security, threats, defenses and
countermeasures you can use to strengthen your Asterisk system.
Meetings
Wait for the iPhone...seriously.
On 1/9/07, Jerry Glomph Black [EMAIL PROTECTED] wrote:
I've had the E70 for about a month.
The first few days were not fun.
But now that I've learned the gotchas and the workarounds, it is GREAT.
You -can- configure it, and asterisk, to work perfectly together,
You need a 'waitexten()' after the background command.
On Tue, 9 Jan 2007, Erik Anderson wrote:
All - this is probably a simple problem, but I've been pulling my hair
out trying to figure out what I'm doing wrong. I'm building a
*simple* IVR menu. Here it is:
[main-menu]
exten =
Derek Whitten
Messages like this SHOULD NOT be posted to the list
I have been trying to block you from my servers do to your abuse
I will add this email address to the list also and contract your service
provider.
You are not doing the right thing you are acting like a child.
I think you are
On 1/9/07, Doug Crompton [EMAIL PROTECTED] wrote:
You need a 'waitexten()' after the background command.
Gah! That worked perfectly. Thanks Doug.
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David
So do you think Digum and Sipro is now one in the same code with G729 in
mind?
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
David Thomas wrote:
This is by far the most volotile list I have ever been on. I'm not
sure that's exactly the
On Tue, Jan 09, 2007 at 02:40:07PM -0800, mitcheloc wrote:
Wait for the iPhone...seriously.
I assume you mean Apple iPhone not Linksys iPhone ?
It looks lovely, shame it's not available in UK until Q4.
(also not FCC approved yet, but I assume that was deliberate as most
phone leaks tend to
I've looked over EVERY resource I can find, but have run short of a
solution. I'm running CentOS 4.4. Just installed Asterisk 1.4 and Zaptel
1.4 and libpri, but when I run ztcfg I get this error: line 0: Unable to
open master device '/dev/zap/ctl'
I realize this is a udev error (or from what
Hi List,
I am using asterisk 1.2.14 with real time and I am trying to send the email to
more than one email address. In that field I put in [EMAIL PROTECTED];[EMAIL
PROTECTED] When the call goes to VM I see in the CLI:
uniqueid = 17
customer_id = 0
context = techmast
mailbox = 14
password = 1234
On 1/9/07, Dovid B [EMAIL PROTECTED] wrote:
Hi List,
I am using asterisk 1.2.14 with real time and I am trying to send the
email to more than one email address. In that field I put in
Send the email to an alias on the system and then have the alias point to
the two email addresses.
This
On 1/9/07, Al Bochter [EMAIL PROTECTED] wrote:
So do you think Digum and Sipro is now one in the same code with G729 in
mind?
If saying this will make this go away, then yes. They both use the same
code. The patented code is the same.
___
I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4, and
Zaptel 1.4
The Digium cards installed are TDM2400 and TE110P.
Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9
Now when I run ztcfg I get the following error message:
(CAS signalling on span 2
dear miche,
pls place your number of softphone B into the context test dial plan.
with best regards,
osochebol
- Original Message
From: Michel [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, January 9, 2007 9:44:20 AM
Subject: [asterisk-users] ooh323c calls
Hi,
Is that procedure the way to completely switch Asterisk from dependency
on MySQL to dependency on Postgres instead? How about with Asterisk 1.4?
And anyone have any idea whether FreePBX can be switched from MySQL to
Postgres, too?
On Tue, 2007-01-09 at 16:01 -0700,
[EMAIL PROTECTED]
Al Bochter wrote:
Derek Whitten
Messages like this SHOULD NOT be posted to the list
I have been trying to block you from my servers do to your abuse
I will add this email address to the list also and contract your service
provider.
You are not doing the right thing you are acting like
M.Hockings wrote:
I don't really know the name of what I want to look for but maybe
someone could tell me if it would be available.
I have a number of old analogue cell phones laying about here and I was
thinking it would be useful if I could set up a short range base station
for them that
Chris Miller wrote:
Eric ManxPower Wieling wrote:
Chris Miller wrote:
I would tend to agree, but the context that holds these number is an
inbound context which includes additional logic that would fail
normal calls. Yes, I can add the DIDs to the outbound context, but
the point here is not
It is true what Eric and Steve have said, you do need a licensed GSM
frequency to operate and sell GSM services (even for rural areas).
however, this link might be of interest to you
http://rfdesign.com/mag/radio_field_trials_allsoftware/
On 1/10/07, Eric ManxPower Wieling [EMAIL PROTECTED]
On Tue, Jan 09, 2007 at 05:50:52PM -0600, Chris Bullock wrote:
I've looked over EVERY resource I can find, but have run short of a
solution. I'm running CentOS 4.4. Just installed Asterisk 1.4 and Zaptel
1.4 and libpri, but when I run ztcfg I get this error: line 0: Unable to
open master
On Tue, Jan 09, 2007 at 06:01:55PM -0700, Administrator wrote:
I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4, and
Zaptel 1.4
The Digium cards installed are TDM2400 and TE110P.
Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9
Now when I
Hi, any one test rtp packetization in 1.4?___
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Hi All,
I am trying to tune out some echo on a analogue line and have run
ztmonitor to get some info. When i run it, i get a RX reading when the
line is idle - is this normal? eg:
[EMAIL PROTECTED] zaptel-1.2.10]# ./ztmonitor 1 -vv
Visual Audio Levels.
Use zapata.conf
Hi
On Tue, Jan 09, 2007 at 05:59:55PM -0500, Al Bochter wrote:
Derek Whitten
Messages like this SHOULD NOT be posted to the list
I fully agree here.
However:
I have been trying to block you from my servers do to your abuse
If you want to blacklist someone on your own servers. However
On Tue, Jan 09, 2007 at 06:44:38PM -0600, Lacy Moore - Aspendora wrote:
On 1/9/07, Dovid B [EMAIL PROTECTED] wrote:
Hi List,
I am using asterisk 1.2.14 with real time and I am trying to send the
email to more than one email address. In that field I put in
Send the email to an alias on
On Wed, Jan 10, 2007 at 05:41:50PM +1100, Ben Dinnerville wrote:
Hi All,
I am trying to tune out some echo on a analogue line and have run
ztmonitor to get some info. When i run it, i get a RX reading when the
line is idle - is this normal? eg:
[EMAIL PROTECTED] zaptel-1.2.10]#
hello all.
i switched to * 1.4 and have now 2 problems.
1. i can't make a call out with the current branch i always have in the
logfile:
[Jan 9 14:45:09] NOTICE[15246] chan_sip.c: Unable to create/find SIP
channel for this INVITE
With the asterisk 1.4 Release it is working,
2. when i do core
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