Folks,
My isp's router limits registrations to only 1 phone number per interface
(i.e., by MAC Address).
I am struggling to get around this limitation.
In sip.conf, I have:
rt200ne=192.168.40.1
register = 3:password:usern...@192.168.40.1/phone1
register =
I'm going to go ahead and say that while I'm not one of the
developers, I think it's safe to say that you cannot record to a file
and play it back at the same time. Probably something like file
locking (for the record, locks it from access by other processes,
etc)...
On Thu, Jul 29, 2010 at
Hi,
we are using Asterisk to record and playback. Both services are working well
independently but it seems we can't start playback of a file while we are
still recording it, even if the file is already in the hard disk.
Is it possible to playback while recording the same audio file? Or is there
Hello ,
Record the file and introduce echo this will give you effect of
recording and playing at same time ;)
On Thu, Jul 29, 2010 at 1:30 PM, Janu Mukherjee janu.mu...@gmail.comwrote:
Hi,
we are using Asterisk to record and playback. Both services are working
well independently but it
SIP s...@arcdiv.com writes:
Spammers sign up to the Asterisk mailing list and send spam once in a
while. My spam filter rejects it, and bounces the emails back to the
Asterisk list, which then drops me from the list because it got a single
bounce.
Don't ever bounce spam! You WILL get
Hi All,
I upgraded 7970 from SCCP to SIP. But the phone isn't registering.
Have you got any working XML file for 7970 phones.
Thanks..
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
thanks Jim
I will check stun server settings asap,
but i have noticed 192.168.x.x is also present in the debug of successful
call having both way audio. so i don't think this has to do anything with
this.
below is the sip debug of successful call .
---
Audio is at 79.80.154.99 port 14034
Supposedly, the filters drop it in the transaction stage. But for some
reason, every time I get dropped from the list, it's just after a spam
email was sent out en masse, so I'm not sure what's up there.
On 7/28/10 10:43 PM, jon pounder wrote:
SIP wrote:
what can you do ? simple discard
Hello,
i have Cisco Unified Communications Manager with 10 ip phone,i dont buy
license IVR of Cisco Unified Communications Manager. Can i use feature IVR
on Asterisk connect with Cisco Unified Communications Manager.
Sorry my English.Thank you.
--
Hello
I have an ericsson md110 with asterisk
Some of my clients are with ericsson and others are sip connected.
I needed a call queue application and i created it on asterisk.
I had to use static clients which are running on ericsson as follows;
Members:
DAHDI/g1/1472 (Unknown) has taken
Please do not top post.
Sherwood McGowan sherwood.mcgo...@gmail.com writes:
I'm going to go ahead and say that while I'm not one of the
developers, I think it's safe to say that you cannot record to a file
and play it back at the same time. Probably something like file
locking (for the
To provide a reliable fax solution for users connected to a Asterisk 1.6.2.6
server i have tested a few T.38 capable ATA's:
- Patton M-ATA
- Grandstream HandyTone 486
- Fritz!Box 7170
I have tried Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 and also
Asterisk 1.6.2.6 with Fax for Asterisk
WireShark does a good job showing the T38 communication. Most products you can
also set packet redundancy to send 2 packets.
Your setup was T.38 ATA to T.38 Gateway with PRI/ANALOG/PSTN/G.711. I've heard
various problems with SIP/PSTN and faxing, around jitter/packet loss and it's
not
Unfortunately, even after two more days of testing, I was not able to get
that channel info from macro into my ruby script.
As it is now - I do not see the way to do it, and this is unsolved.
If any of you good folks have any idea how to pull the dumpchan data from
macro to ruby/agi script
Hi,
I have Asterisk 1.4, trought ODBC I'm savind CDR to MSSQL. How can I change
the fields name of database?
Regards Andraz
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Strictly a shot in the dark, but have you tried
$loc=$agi.get_variable('CHANNEL') ?
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On Thu, Jul 29, 2010 at 2:19 AM, jwex...@mail.usa.com wrote:
My isp’s router limits registrations to only 1 phone number per interface
(i.e., by MAC Address).
MAC Address? Are you sure? Why would your ISP care about level 2? I
could understand IP address (level 3). If this is the case, you
On Wed, Jul 28, 2010 at 9:06 PM, bruce bruce bruceb...@gmail.com wrote:
See the jump from Jul 23rd to Jul 26th. Is this an indication of Asterisk
being down?
No, it just means there was no logger activity for those days. You
need to add a monitoring solution to your Asterisk box (IE: AMI:
- Non-Commercial Discussion'
Subject: Re: [asterisk-users] Urgent help = RUBY AGI
Strictly a shot in the dark, but have you tried
$loc=$agi.get_variable('CHANNEL') ?
__ Information from ESET NOD32 Antivirus, version of virus signature
database 5323 (20100729) __
The message
I can't even spell RUBY, so I don't have a clue as to how the AGI works. I
do know a little bit about AGI in general. The way I typically run my AGI's
is something like this:
exten = 933,1,Answer
exten = 933,n,Set(ABA=02107)
exten = 933,n,Set(city=Birmingham)
exten = 933,n,Set(state=AL)
On Thu, Jul 29, 2010 at 7:22 AM, Nguyen Quang Tri kihote...@gmail.com wrote:
Hello,
i have Cisco Unified Communications Manager with 10 ip phone,i dont buy
license IVR of Cisco Unified Communications Manager. Can i use feature IVR
on Asterisk connect with Cisco Unified Communications Manager.
Hi!
I have Asterisk 1.4, trought ODBC I'm savind CDR to MSSQL. How can I
change the fields name of database?
You will want the adaptive CDR backport to Asterisk 1.4:
https://issues.asterisk.org/view.php?id=1
http://svncommunity.digium.com/view/tilghman/branches/1.4/
Philipp
--
That looks easy. I must say that I am very frustrated as this has took my
all week, and beside dumpling that data via macro I wasnt able to
use that data in the ruby script that we have. I didnt write the script is
something old that we use but i was sure we could add few things in that
very
MSSQL Updateable Views can help you
On Thu, Jul 29, 2010 at 7:15 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
I have Asterisk 1.4, trought ODBC I'm savind CDR to MSSQL. How can I
change the fields name of database?
You will want the adaptive CDR backport
On Thu, 29 Jul 2010, Danny Nicholas wrote:
I’m a PERL weenie, so I can “shell check” my agi’s by going to
/var/lib/asterisk/agi-bin and doing
./cityweather.agi 02107 Birmingham AL 35244 en
And getting back a STDOUT output that simulates what I should get from
the CLI output.
You can
Do you just have one agi you are running? If so that will not work. Your one
agi is hung on the dial step until it finishes at which time the agi will go
away, I think. You need the one agi to cause the dial to occur and another one
to capture the information when the macro runs. You can try
On Thu, Jul 29, 2010 at 2:32 PM, Benny Amorsen benny+use...@amorsen.dk wrote:
Sherwood McGowan sherwood.mcgo...@gmail.com writes:
I'm going to go ahead and say that while I'm not one of the
developers, I think it's safe to say that you cannot record to a file
and play it back at the same
Depending on what you are recording there might be two files, one for each leg
of a call, until the call ends and the files are mixed.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 29, 2010, at 7:46 AM, Motiejus Jakštys wrote:
On Thu, Jul 29, 2010 at 2:32 PM,
I am not sure why it would be sleeping. I have never dealt with putting a
linux server to sleep. It is connected to a UPS, but I don't think it has
been put to sleep by the UPS as the USB cable from UPS is not connected to
it.
Can you please elaborate on what you mean by AMI:Ping? Is there a
Hi,
my question is how can i get channel-id of a user or peer. I tried using
ChanIsAvail(username). this works correctly when user and asterisk are on
Local LAN. But my asterisk server is on public ip and users are behind nat,
and so this method says unknow host when used on public asterisk
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nasir Javaid
Subject: [asterisk-users] How to extract channel-id of a user or peer
my question is how can i get channel-id of a user or peer. I tried using
ChanIsAvail(username). this works
Hi,
I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect
hangup tone or disconnect supervision tone from my CO. I attached the
recorded wav file which contains my telco's disconnect supervision.
I am using ,
asterisk-1.4.33.1
dahdi-linux-complete-2.3.0.1+
We will need to see your dahdi.conf and chan_dahdi.conf files as well.
--
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On Thu, Jul 29, 2010 at 10:37 AM, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nasir Javaid
Subject: [asterisk-users] How to extract channel-id of a user or peer
my question is how can i get
Hi all,
I am wondering if the Clustering concept described in Leif Madsens presentation
http://comunidad.asterisk-es.org/documentos/astricon2008/why-cluster-an-introduction-to-asterisk-clustering-and-database-integration-astricon-2008.pdf
is still up to date or if there are newer or improved
So historically I've done one of two things on systems where I've
needed to use MeetMe
* used a real Digium card, and I've only ever used a TE400 or a TE420
for that purpose, and I know they have the timing chip
* used dahdi_dummy, which works well with light load, but I had it
running on a very
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
unsero...@aol.com
Sent: Thursday, July 29, 2010 3:57 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Hi all,
I am wondering if the Clustering concept described in Leif Madsens
presentation
http://comunidad.asterisk-es.org/documentos/astricon2008/why-cluster-an-introduction-to-asterisk-clustering-and-database-integration-astricon-2008.pdf
is still up to date or if there are newer or
On Thu, Jul 29, 2010 at 5:04 PM, unsero...@aol.com wrote:
Do you know if it is possible to interconnect 1.6 with Microsoft Office
Communications Server 2007 and use the Office
Communicator as a softclient for telephone calls and the Communicator for
Instant Messaging? I believe you can set up
Do you know if it is possible to interconnect 1.6 with Microsoft Office
Communications Server 2007 and use the Office
Communicator as a softclient for telephone calls and the Communicator for
Instant Messaging? I believe you can set up a mediation
server within MOC but i don't know if
No, not until Microsoft builds a compatible soft phone. Microsoft
built software that only speaks SIP over TCP. Most SIP stacks work
over RTP.
I suspect you meant UDP, not RTP. They use TCP or UDP for SIP signaling and
RTP for the actual voice traffic.
--
No, not until Microsoft builds a compatible soft phone. Microsoft
built software that only speaks SIP over TCP. Most SIP stacks work
over RTP.
I suspect you meant UDP, not RTP. They use TCP or UDP for SIP signaling and
RTP for the actual voice traffic.
--
This is what i thought.
Thx for all the responses. Really appreciate it.
I will try putting the FQDN toIP address mapping in the /etc/hosts file to see
if that makes a difference.
I will also setup a cron to restart it every day...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi All:
If we upgrade asteriskNow from 1.4.18 to 1.7.0; just want to make sure all
the config file functionality will reamin same
That is - everything in /etc/asterisk will still work the same way.
Users.conf
Provider.conf
Extensions.conf
Sip.conf
Etc...
Thx in advance.
--
I have a problem with a Sangoma card. It worked until yesterday. Now
I keep getting this error:
Jul 29 17:45:17 pbxacura kernel: wanpipe1: Enable E1 CAS signalling
mode!
Jul 29 17:45:17 pbxacura kernel: wanpipe1:w1g1: Rx Error: No
'DeviceSelect' from target: pci fatal error!
Manmohan wrote:
Following is the output for core set verbose 5,
also i am really not able to get on the admin pin
thing? Do you mean, that with admin pin configured
we cant use recording?
You are actually running a version that has been fixed
to support recording with pin-less or user pins.
On Thu, Jul 29, 2010 at 10:15 PM, Paul Belanger wrote:
MAC Address? Are you sure? Why would your ISP care about level 2? I
could understand IP address (level 3). If this is the case, you will
need to spoof your MAC.
Actually, it is mind boggling that the isp even cares about restricting
On 07/29/2010 03:35 PM, David Backeberg wrote:
So historically I've done one of two things on systems where I've
needed to use MeetMe
* used a real Digium card, and I've only ever used a TE400 or a TE420
for that purpose, and I know they have the timing chip
* used dahdi_dummy, which works
Thanks Sherwood for all the info.
The devices are using ulaw and rfc2833. There is no transcoding on my
server, but not sure what my trunk providers are doing.
I was thinking about the frequency detection issue as it seems to be
primarily involving women so I'll try adjusting the input/output
The solution is really simple.
Make a context in your extensions.conf file:
[context]
Exten = WebCallers,1,Dial(SIP/100SIP/101)
And now on your Action script:
Action: Originate
Channel: Local/webcall...@context
..
Thanks!
Zachary Kitchen
[cid:image001.gif@01CB2F54.0CECE080]
Hi. I am using *1 in features to initiate a mix monitor recording.
However, when I hit *1, the callee hears the courtesy tone which I have,
so I know when the recording is started or stopped. This is a problem,
particularly in automated system where the beep is mistaken for a tone
or other
On Thu, Jul 29, 2010 at 4:05 PM, jwexler jwex...@mail.usa.com wrote:
On Thu, Jul 29, 2010 at 10:15 PM, Paul Belanger wrote:
MAC Address? Are you sure? Why would your ISP care about level 2? I
could understand IP address (level 3). If this is the case, you will
need to spoof your MAC.
Interesting offering, free from Twilio, this is php you install on
your own server to build a brandable VBX. Worth checking out!
Listen to tomorrow for more about this and talk to lead engineer or
Twilio CEO if you have any questions;
sip:200...@login.zipdx.com or Skype:vuc.me
IRC: #vuc on
- Carlos Chavez cur...@telecomabmex.com wrote:
I have a problem with a Sangoma card. It worked until yesterday.
Now
I keep getting this error:
Jul 29 17:45:17 pbxacura kernel: wanpipe1: Enable E1 CAS signalling
mode!
Jul 29 17:45:17 pbxacura kernel: wanpipe1:w1g1: Rx Error: No
bruce bruce wrote:
I am not sure why it would be sleeping. I have never dealt with
putting a linux server to sleep. It is connected to a UPS, but I don't
think it has been put to sleep by the UPS as the USB cable from UPS is
not connected to it.
Can you please elaborate on what you mean by
Lyle Giese wrote:
bruce bruce wrote:
I am not sure why it would be sleeping. I have never dealt with
putting a linux server to sleep. It is connected to a UPS, but I
don't think it has been put to sleep by the UPS as the USB cable from
UPS is not connected to it.
Can you please elaborate on
Hi Everyone,
I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The phones
occasionally go into No Service mode. The POE switch doesn't seem to be
the problem as it's tested fine. I think the router sometimes gives up and
comes back quickly. Or something of that nature. However,
This was a static IP. Further checks into the server prevails that there are
no logs of what happened on the 24th and 25th even in /var/log/messages.
This makes me believe that a hardware lockup has happened and according to
people on CentOS forum this is VERY HARD to diagnose as there will be no
Hi,
I changed all settings in users.conf to chan-dahdi.conf here is
configuration settings.
chan_dahdi.conf:
--
signalling=fxs_ks
busydetect=yes
busycount=3
busypattern=300,300
answeronpolarityswitch = no
hanguponpolarityswitch = no
callprogress = no
progzone = in
Hi Dan,
I did added the record option in user options as well.
$Mod_Options = array(array(_(Announce), I), array(_(Record), r));
$User_Options = array(array(_(Announce), I), array(_(Listen Only),
m), array(_(Wait for Leader), w), array(_(Record), r));
And the gre8 news is, it did worked this
On 7/29/2010 6:51 PM, Travis Langhals wrote:
Thanks Sherwood for all the info.
The devices are using ulaw and rfc2833. There is no transcoding on my
server, but not sure what my trunk providers are doing.
I was thinking about the frequency detection issue as it seems to be
primarily
On Thu, 29 Jul 2010 20:47:58 -0500 (CDT), Tim Nelson wrote
- Carlos Chavez cur...@telecomabmex.com wrote:
I have a problem with a Sangoma card. It worked until yesterday.
Now
I keep getting this error:
Jul 29 17:45:17 pbxacura kernel: wanpipe1: Enable E1 CAS signalling
mode!
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