Phibee Network Operation Center a écrit :
Hi
Now, my Cisco AS5300 sent call to my asterisk, but two problems:
When i call the phone number, i have:
[Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension '042600' rejected because extension
On Oct 27, 2009, at 10:50 PM, trebaum wrote:
Ok, so this might seem like a stupid question, but I don't quite
understand how to dial out to the pstn though my T1 from a specific
number. Maybe i'm missing something, but everything I'm reading has
you dial a number from the group but that's
hello,
is there any facility to get SIP client (ex. softphone,ipphone) MAC address
on asterisk.
based on that we authenticated client in anyway.
i tried with sip debug but i didn't got any MAC address related field in all
packets.
regards
Dhaval
From Linux you could use
arp | grep 192.168.0.1
substituting the IP address of the SIP device.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Wednesday, 28 October 2009 4:29 PM
To: Asterisk Users
hello david,
what in case of sip client is behind NAT, and i want SIP client IP
address. not from system from which client
registered. if it is a SIP phone then what? if you have any idea then tell
me.
regards
dhaval
On Wed, Oct 28, 2009 at 12:02 PM, Klaverstyn, David C
If there is more than one SIP devices operating from the same NAT device
then I'm not sure what you could do as it would always show the same IP
for all SIP devices behind the same NAT. If there is only one device
behind that NAT making a connection to your server then that is easy, if
not I
hi,
though , the SIP client is behinf the NAT cannot we get MAC address of that
client , from SIP headers.
or do you suggest any alternate method .
regards
dhaval
On Wed, Oct 28, 2009 at 12:20 PM, Klaverstyn, David C
david.klavers...@intergraph.com wrote:
If there is more than one SIP
Hi,
One more interesting fact, i see correlation with DTMF features, after i
disabled corresponding options on dial commands (like htw) the
timestamps on rtp are constantly growing and no more one way audio
problems after call transfer, hold, parking etc. So it seems there is a
bug related to
On Sun, 2009-10-25 at 17:10 -0400, Matt wrote:
Greetings,
Where can I get the chan_echolink channel driver from? I've seen
reference to it, but have yet to find a place to download/compile it.
It is part of the app_rpt.so module... I am told, but do not see the
source with app_rpt.
Hello
I have a scenerio to integrate an Existing Panasonic PBX with a new PBX that
will be Asterisk system.
I know that Asterisk can be integrated with existing Panasonic TDA 100 PBX
to recieve calls (ie PSTN lines to Panasonic PBX and out lines of Panasaonic
to in of Asterisk PBX).
--But i am
Thanks all
Robin Drop Box looks cool but I have developed my own code in JAVA that will
use Sockets to syncronize files across different servers.
Thanks Arjan for the link.
@ li...@torrenga.com yeah i do have considered but finally developed my own
code for sysncronization. thanks :)
if Any One
Double-check the IP and port associated with the AS5300 peer. The
messages below indicate that calls coming in from it are not being
matched to the right peer, and as a result, not routed to the correct
dial plan context.
Phibee Network Operation Center wrote:
Hi
Now, my Cisco AS5300
This is a very strange discussion.
MAC addresses can only be discovered for peers that are on the same
broadcast segment - which is the realm within which ARP lookups
participate.
Any peers not on the same logical Layer 2 network are reached through
a Layer 3 hop. MAC addresses behind that
Try throw the following options into your sip.conf peer:
port=5060
insecure=invite,port
Phibee Network Operation Center wrote:
Phibee Network Operation Center a écrit :
Hi
Now, my Cisco AS5300 sent call to my asterisk, but two problems:
When i call the phone number, i have:
[Oct
I am talking about the SIP.
Now the new mobiles (Nokia, Erecson, Panasonic, iPot, ... etc) all of them
support SIP capability. They are able to register to any SIP server (by giving
the IP address, username and password). Fring is one of the software that can
be installed on the mobile devices
Hi Matt,
That is exactly what I am doing now and it has solved my problem. Now all
the calls originate instantly with no noticeable delay.
--
Zeeshan A Zakaria
On Wed, Oct 28, 2009 at 12:18 AM, Matt Riddell li...@venturevoip.comwrote:
On 28/10/09 3:52 AM, Danny Nicholas wrote:
This might
Hello, when I remove a peer from my sip.conf and just do a reload, the peer is
still ping with SIP OPTIONS until I restart Asterisk, I use
Asterisk 1.4.27-rc2. Is it normal? Thanks
As an example, I have added and after removed this lines and
;[sip_trk_vm]
;host=88.191.80.8
;type=peer
Have you set the realm in the sip settings in the mobile? Default one is
asterisk . It's important too, defining Registration to Always on, because
if not, it doesn't enable the wifi connection. Finally, don't enable
compression and security
--- El mié, 28/10/09, bilal ghayyad
Hi people, when I try to get the billsec in the dialplan, it is 0... but if
after that I check the database, it is right (not 0).
I'm trying to get it in the h extension, like:
exten = h,1,Noop(End)
exten = h,n,Noop(Time is ${CDR(billsec)})
Is it updated after the extension h is executed?
Hi
We're using asterisk 1.4.17 with RealTime so our port and fullcontact
values in out DB get updated dynamically.
We use snom handsets and always set the network identity (port) in each
phone to something in the 1 range, so that each phone in a single
location has a different port.
When
If it is SIP use following softphones:
1) X-lite
http://counterpath.com/x-lite.htmlactive=4
2) SJPhone
http://www.sjlabs.com/sjp.html
3) Snom
http://www.snomindia.com/snomsoftphone.htm
On Wed, Oct 28, 2009 at 3:36 AM, Steve Edwards asterisk@sedwards.comwrote:
On Tue, 27 Oct 2009, giancarlo
Title: Private Message from Ravindra
Ravindra K has sent you a private message Click to read messagePlease read it or Ravindra will think you ignored this :( This message has been forwarded at the request of ravi...@gmail.com. To block all emails from FanIQ, please click
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Ravindra K wrote:
FanIQ http://FanIQ.com/user/ravibth/connect/334259105/?etid=207
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Click to read message
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Read private message
Give zoiper a try, http://www.zoiper.com (I'm working for them)
Works with SIP and IAX, and should be pretty easy to setup.
Zoa
giancarlo lombardo wrote:
I just installed an Asterisknow server
can someone suggest a software to be used for a PC - PC voice
comunication to test in easy way
Since CDR(billsec) is a live variable until the Hangup command is issued
(actually until the CDR is written), the only way to get the value (IMO)
would be after the call was completed. You could do a DeadAGI or System
call using CDR(uniqueid) to report the value from the CDR back to another
call
Mea Culpa?? Since I've only been dabbling with AMI for about 6 weeks, I
hadn't stumbled upon the Async parameter. A more correct dissertation of
the sentence would be
The AMI originate by default operates in a synchronous or threaded fashion,
unless you specify Asynchronous mode using Async:
Actually no. But i cannot get a smartnet on an ATA-188. At least not in
latinamerica.
Actually, all ata-188/186 come with sccp, i just reflashed mine to sip and
now i want it back to sccp. it was very dissapointing to learn that i cannot
download *any* sccp firmware, not even the original one.
Hi,
If you want an online option to make calls right from webpage, you can use
doddle online SIP webphone:
http://widget.doddlephone.com/
Sergio
On Wed, Oct 28, 2009 at 11:11 AM, Zoa zoach...@securax.org wrote:
Give zoiper a try, http://www.zoiper.com (I'm working for them)
Works with
Hello Anahi,
I've encountered issues with CDR function when I was using the
1.4 version and was trying to get ${CDR(duration)} in extension h.
Passing to 1.6.X.X resolved it.
I hope this helps.
Alex
From:
hello all, friends i am new in asterisk. i had just finished the
installation requirment of asterisk. i am using Centos 5.3 in which ill be
installing asterisk now guys plz guide me my requirment for deploying
asterisk is, i am having a client, (HR Consultancy) where 40 executives
work and
Go to www.asterisk.org http://www.asterisk.org/ and read the install
from YUM repo section. This will make the install pretty much automatic.
You will then want to set up a queue to route your incoming calls to your 40
extensions. You do not state what technology (SIP/DAHDI) you want to use to
Hi all,
I have a question regarding pending (zombie) SIP sessions: on Asterisk CLI,
with command 'sip show channels' , I see two channels in use with callID and
other infos detailed; also 'sip show inuse' give me same result (in terms of
channels usage):
PeerUser/ANRCall ID
Hi,
You could use:
soft hangup [channel name]
Note: You can write the first letter of the channel name, and use [Tab]
key to autocomplete.
Regards,
Aggio Alberto wrote:
Hi all,
I have a question regarding pending (zombie) SIP sessions: on Asterisk
CLI, with command ‘sip show channels’
I am sure many of you have seen my post asking question that I cannot seem to
resolve. While the responses i have been getting have been helpful i still
cannot seem to get this working 100%.
So I have waving the white flag here. I give up. I need someone to come to my
office and help me get
Might want to try these guys
http://www.bluegrassnetvoice.com/services/customerpremisePBX.html
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Wednesday, October 28, 2009 9:59 AM
To:
Does this mean its a bug in 1.4 or an enhancement in 1.6? If the latter,
can the change be back-ported?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alexandru
Oniciuc
Sent: Wednesday, October 28, 2009 8:52 AM
To:
aster...@opensourcesolution.in wrote:
hello all,
friends i am new in asterisk. i had just finished the installation
requirment of asterisk. i am using Centos 5.3 in which ill be installing
asterisk now guys plz guide me my requirment for deploying asterisk is,
i am having a
On Wed, Oct 28, 2009 at 5:05 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
Let's be realistic here. You need to 'drink the koolaid' before you
install it for a client. What I'm saying is you really need to install
Darrick,
No, he already drank the koolaid by believing in asterisk. Now
aster...@opensourcesolution.in wrote:
friends i am new in asterisk. i had just finished the installation
requirment of asterisk. i am using Centos 5.3 in which ill be
installing asterisk now guys plz guide me my requirment for deploying
asterisk is,
i am having a client, (HR
I have used ${CDR(billsec)} in asterisk 1.4.17
How I used it was
h,1,SET(BILLTIME=${CDR(billsec)})
h,2,DeadAGI(hangup.php)
My DeadAGI script could use my BILLSEC variable and it was always
consistent with the CDR too.
Danny Nicholas wrote:
Does this mean it’s a bug in 1.4 or an enhancement
Something seems to be missing here- you don't pass ${BILLTIME} to hangup.php
(as far as I can see), so it seems that hangup.php is operating (at least
somewhat) independently of the dialplan. The OP seemed to want in-line
knowledge of his billable seconds.
-Original Message-
From:
I used 1.4.21 and this(${CDR(duration)}) didn't work:
exten = h,1,Verbose( (${CDR(dst)}) # Call from ${CDR(clid)} ended at
${STRFTIME(${EPOCH},,%d/%m/%Y %H:%M:%S)}. Duration(sec): ${CDR(duration)}.)
Alex
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
Thanks,
it sounds good.
2009/10/27 giancarlo lombardo gianclomba...@gmail.com
I just installed an Asterisknow server
can someone suggest a software to be used for a PC - PC voice comunication
to test in easy way the functionalities of my server.
Thanks in advance for the help
--
I would second Steve's advice very strongly.
Steve Edwards wrote:
aster...@opensourcesolution.in wrote:
friends i am new in asterisk. i had just finished the installation
requirment of asterisk. i am using Centos 5.3 in which ill be
installing asterisk now guys plz guide me my requirment
This has been a rollercoaster ride
Building a new gateway (Asterisk 1.6.1 / Sangoma A108D 3.5.8 drivers)
Where I stand right now, I have a PRI on the gateway and circuit is
working I can make calls through the gateway
Here is my problem:
DAHDI_TEST is not returning anything and
When I make an outbound call I hear a half of a ring and than silence until
the call opens up.
It seems asterisk is sending a 183 after the 180 message. My CPE device does
not support multiple 18x messages in the same call setup. When we receive
the 180 we present ring back to the phone, but
On Wed, Oct 28, 2009 at 2:09 PM, Robert Grignon rgrig...@fleetone.com wrote:
This has been a rollercoaster ride
Building a new gateway (Asterisk 1.6.1 / Sangoma A108D 3.5.8 drivers)
Where I stand right now, I have a PRI on the gateway and circuit is working
I can make calls through the
Did you use ./Setup dahdi when installing the wanpipe drivers?
http://wiki.sangoma.com/wanpipe-linux-asterisk-dahdi
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To UNSUBSCRIBE or update options
That was a good thought. I have 3 other gateways in production and I ran
dahdi_test and zttest (older gateways) and they all said they were opening a
psedu device
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Yes I did that...
I even recompiled dahdi-linux and tools after wanpipe install... Once I
did that it recognized the card and said I could run dahdi_genconf
modules which in turn would only load the cards that it seeing.
I had the PRI running in slot 6. Once I unplugged the PRI I was able to
Upon further research I kind of answered my own question.. But I will
share...
If you are seeing multiple H.100 errors in your system log then the
hardware echo canceler does not have a good clock source. On our more
recent drivers 3.3.12 and up the first port that starts up will be the
clocking
Tim King wrote:
When I make an outbound call I hear a half of a ring and than silence
until the call opens up.
It seems asterisk is sending a 183 after the 180 message. My CPE device
does not support multiple 18x messages in the same call setup. When we
receive the 180 we present ring back
I thought that was it and tried each setting and did not see any change on
the line.
On Wed, Oct 28, 2009 at 3:58 PM, Kevin P. Fleming kpflem...@digium.comwrote:
Tim King wrote:
When I make an outbound call I hear a half of a ring and than silence
until the call opens up.
It seems
I am having a strange problem with MOH. Say I have two users, A and B. I
can set MOH in the extension for B and if A calls B and B hits hold, A will
hear B's hold music. If however A hits hold, it goes to the default music.
If I pull the setmusiconhold from extensions.conf and use musicclass in
Peder wrote:
I am having a strange problem with MOH. Say I have two users, A and B. I
can set MOH in the extension for B and if A calls B and B hits hold, A will
hear B's hold music. If however A hits hold, it goes to the default music.
If I pull the setmusiconhold from extensions.conf and
On Wed, 2009-10-28 at 14:59 +, Ott Rose wrote:
I am sure many of you have seen my post asking question that I cannot
seem to resolve. While the responses i have been getting have been
helpful i still cannot seem to get this working 100%.
So I have waving the white flag here. I give
Please post your dial peer configurations.
We have as5400 (5) working with asterisk servers also.
The cisco routers are at the edge of the network (connected to PSTN via
E1) and send calls to asterisk over SIP
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On Wed, Oct 28, 2009 at 10:16:16PM +0100, Hans Witvliet wrote:
On Wed, 2009-10-28 at 14:59 +, Ott Rose wrote:
__
Windows 7: Simplify your PC. Learn more.
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I've tested and confirm that the AGI script can do that.
i had to enable setinterfacevar=yes in the queue conf and then can
read the MEMBERINTERFACE channel variable.
Just because it can be useful for someone else.
On Fri, Oct 23, 2009 at 9:44 PM, nik600 nik...@gmail.com wrote:
Hi to all
Hello,
I have an * installation that sometimes receives a 302 "Moved
Temporarily" response to an INVITE. * sends the ACK but takes about 30
seconds to start the new INVITE to the new destination (from Contact
Header).
I have set core debugging to 20 but do not see any abnormal message.
I am having a problem with Asterisk 1.6.2.0-rc3 and Asterisk-Addons
1.6.2.0-rc1 when recording CDR to a Mysql database. All fields except
callerid are recorded properly after every call. I have both a clid
and callerid field in the database but both fields are empty. In
cdr_mysql.conf I
Any simple legacy integration will work. Search on voip-info.org
Here are some problems that I know exist with panasonic systems on
their SLT (analog) ports:
1. No CPC, Asterisk if connected using station ports on the TDA to FXO
on asterisk, will not detect hangups since the TDA will not send
Hi, all. I've got an Asterisk box installed that I'd really like to
leverage -- and installing a GUI for hunt groups would be awesome. So
long as I can have a trial copy, I could even pay money. It would have to
be able to make use of both SIP and ZAP extensions.
Suggestions?
(Note: I
Freepbx comes with setup of ring groups and queues with different hunt
strategies
Also it has Flash Operator Panel which gives you the state of the system
in real time graphical format
No money - just a small bit of installation time and learning how to use it
Cheers Duncan
Ken D'Ambrosio
I have a trunk, and its host=dynamic dns.
The problem is, when the IP changes the
Sip show peers
Still show the old IP of the DNS, I have to reload and save the
configuration again so that asterisk recognize the new IP of the DNS.
Any idea how to automate such a thing? Or how can I keep
C F thankyou very much.
when i make a call to Asterisk server recieves and works fine. But as to
make external calls we have to press nine so supposed a logic to dial 9
first then wait and then dail other number. But as i dail 9 asterisk show
the call as connected with alot of noise. Please help
If the trunk is a dynamic IP you need the other end to register to
Asterisk, so letting Asterisk know the new IP.
Regards,
Juan
B.Masoud @ SH wrote:
I have a trunk, and its host=dynamic dns.
The problem is, when the IP changes the
Sip show peers
Still show the old IP
On Wednesday 28 October 2009 17:57:49 Carlos Chavez wrote:
I am having a problem with Asterisk 1.6.2.0-rc3 and Asterisk-Addons
1.6.2.0-rc1 when recording CDR to a Mysql database. All fields except
callerid are recorded properly after every call. I have both a clid
and callerid field in
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