Hi guys this is my Ist mail on this group, I am running asterisk with CentOS
4.4 machine. When i initiate a call then error message apears. calling
Number is provided to Asterisk by the php application. Error message appears
like this
Got SIP response 500 "Internal Server Error" back from 209.47.9
> Is it possible to configure a PAP2 to
> auto-answer for either paging or intercom?
No. You cannot force the connected device (phone) to auto-answer.
Imagine you have a plain old phone attached to it, who's going to lift
the receiver?
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Hi list,
I'm about to install Asterisk on an Old HP NetServer LC2000 Server (year
2001), it has 2 Pentium III 1GHz CPUs (Coppermine FSB 133MHz 256K L2 Cache),
768MB PC-133 ECC RAM, 3 UltraSCSI LVD2 18.2GB 10K RPM HDD in RAID5, 100Mb
NIC for server.
This Server will support 35 SIP phones (users) a
We've got our Polycom phones auto-answering
for paging.
Is it possible to configure a PAP2 to
auto-answer for either paging or intercom?
If so, how?
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asterisk-users mailing list
T
I was wondering how everyone here is giving users (say via the BLF on a
Polycom, or the sidecart/buddies) the ability to see how many channels they
have in their group and how many are in use. Since so many users are used to
seeing Line 1, 2, 3 etc on a key system I have been trying to think ab
Hi list,
I'm evaluating a private telephony scenario of about 20
locations - 300 phones, 50 FAX machines.
Initial overview points to the installation of asterisk at three
locations connected to the PSTN via ISDN PRI.
All other locations, small by themselves, would get SIP
phones ma
Are you using zap channels with 'aggressive' echo suppression enabled?
That will make calls pretty half-duplex.
Moj
jamespev wrote:
>
>Hello. We are very successfully using asterisk (in the form of
> trixbox 2.2/asterisk 1.2). We run a few conference lines for customer
> teleconferences
If all the services are for internal use and authorized external use
then there would be no problem with doing this. Deny all ports on the
external facing interface except 1194 or whatever you want to run
OpenVPN on and you can connect remotely over the VPN and be totally safe
from the outsid
Nonsense! I'm a Security Expert (TM) and I say run EVERYthing on your
firewall
And...uh... what was your IP again? ;)
N.
Steve Prior wrote:
>> GNUbie wrote:
>>
>>
>>> By the way, my Asterisk PBX server is also my wireless access point,
>>> web server, file server, music server, VPN
I have not noticed this here at all -- although too much of talking
over each other makes a mess, but in both 1.2 and 1.4 I have not
noticed any such behavior. What are you using for a carrier?
on Wednesday 10/10/2007 jamespev([EMAIL PROTECTED]) wrote
>
> Hello. We are very successfully us
Hello. We are very successfully using asterisk (in the form of trixbox
2.2/asterisk 1.2). We run a few conference lines for customer teleconferences
which mostly work well but they seem to operate at half duplex. If a person
starts talking they will cut off others on the call. Is this no
My opinion:
1.4 is a branch.
current trunk should be called 1.5
1.5 should be 1.5.1.1, 1.5.1.2 ,1.5.1.3,1.5.2
In the above, X.X.Y denotes the "stable" version. Any changes to that code,
would use the next point value. 1.5.1.Z
You do not change to 1.5.2."0" until it has been tested, thus 1.5.2 wou
Hi guys, I'm not sure here is the best place to ask, but, anyone has
some news regarding to this bug? I'm having problems with this in one
customer.
Thanks
Carlos Barros
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asterisk-
Hi
I am new to Asterisk, I am writing a softphone but facing few problems:
1. Call is successfully established between two clients but I am unable to
receive RTP packets. All PCs are in same network domain.
One of the client is X-lite and other client is my softphone.
Vinay
_
> GNUbie wrote:
>
>> By the way, my Asterisk PBX server is also my wireless access point,
>> web server, file server, music server, VPN server, database server,
>> firewall and router.
>>
Repeat after me - NEVER NEVER NEVER run other servers on your
router/firewall machine!!! That machine need
Awesome. Thanks all. I am still gonna work on some other possible logic. It
would really be cool to have all of that functionality in Asterisk.
Reg
>>> "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]> 10/10/2007 3:24 PM >>>
Reggie Payne wrote:
> The call is recorded after a key sequence
On Wed, Oct 10, 2007 at 02:10:54PM -0400, SIP wrote:
[snip]
> I think that using 1.5.x as the name for a release candidate for 1.6 is
> pretty close to as unintuitive as it can possibly be.
> 1.6.Xrc-Y is a strikingly MORE intuitive naming scheme for 1.6 release
> candidates.
mutt uses the x.y
I setup Trixbox on an Dell Precision 360. I ported my old POTS line over
to a pay-as-you-go through Teliax because we weren't using more than 500
minutes a month on the home line.
When a caller rings in, I screen the call with time-of-day routing. In
general, if the call comes before 7:30 AM or af
Reggie Payne wrote:
> The call is recorded after a key sequence has been pressed.
>
> Example:
>
> SIP/101 makes an outbound call to 5551212
> 5551212 starts to get rowdy
> SIP/101 enters *99 to start recording the call
> After the call is ended the recording is sent to the voicemail of 101
>
Reggie Payne wrote:
> The call is recorded after a key sequence has been pressed.
>
> Example:
>
> SIP/101 makes an outbound call to 5551212
> 5551212 starts to get rowdy
> SIP/101 enters *99 to start recording the call
> After the call is ended the recording is sent to the voicemail of 101
>
Péter Tóth wrote:
> When i try ztmonitor as follows, it gives strange output...
>
> [EMAIL PROTECTED] zaptel-1.2.10]# ./ztmonitor 1 -vv
>
> Visual Audio Levels.
>
> Use zapata.conf file to adjust the gains if needed.
>
> ( # = Audio Level * = Max Audio Hit )
> <--
Russell Bryant wrote:
> I have been having discussions with various members of the development
> community
> in regards to changes to the way we manage open source Asterisk releases. The
> changes that we eventually decide on will determine how we manage the 1.6
> version of Asterisk. I will be
Typically, echo isn't heard on the _far_ end, unless it is created by
acoustic effects within the phone hooked up to the IAXy. Can the
microphone hear the speaker? You said you've tried numerous analog
phones, so that kind of rules that out, but curious...
Sean Dennis wrote:
> From what I
rc1, rc2 is the best choice for me.
Best Regards. Emiliano Vazquez.
- Original Message -
From: "Russell Bryant" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, October 10, 2007 2:54 PM
Subject: [asterisk-users] Opinions on Release Nu
On Wed, Oct 10, 2007 at 12:54:42PM -0500, Russell Bryant wrote:
> What is your opinion? I certainly want the release naming to be as obvious as
> possible.
Wikipedia has something to say on this (by which, of course, I mean me
:-)...
The traditional approach to this is, roughly
1.5.8
1.5.9
1.5.
On Wednesday October 10 2007 2:15 pm, Doug Lytle wrote:
> Russell Bryant wrote:
> > I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc.
> >
> > What is your opinion? I certainly want the release naming to be as
> > obvious as possible.
I would say the rc-1, rc-2 is about as obviou
Russell Bryant wrote:
> I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc.
>
> What is your opinion? I certainly want the release naming to be as obvious as
> possible.
>
>
Then I think that would be the rc1,rc2 style then.
Doug
--
Ben Franklin quote:
"Those who would
Russell Bryant wrote:
> I have been having discussions with various members of the development
> community
> in regards to changes to the way we manage open source Asterisk releases. The
> changes that we eventually decide on will determine how we manage the 1.6
> version of Asterisk. I will be
Russell Bryant wrote:
> I have been having discussions with various members of the development
> community
> in regards to changes to the way we manage open source Asterisk releases. The
> changes that we eventually decide on will determine how we manage the 1.6
> version of Asterisk. I will be
The call is recorded after a key sequence has been pressed.
Example:
SIP/101 makes an outbound call to 5551212
5551212 starts to get rowdy
SIP/101 enters *99 to start recording the call
After the call is ended the recording is sent to the voicemail of 101
>>> Razza <[EMAIL PROTECTED]> 10/10/20
I second calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc.
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On 10/10/2007, Reggie Payne <[EMAIL PROTECTED]> wrote:
>
> Hello All! I am new to the list. Does know how to record a call on
> demand? What I would like to do is setup something that during a call
> someone can hit a button a the call is recorded the after the call is over
> the recording is se
I have been having discussions with various members of the development community
in regards to changes to the way we manage open source Asterisk releases. The
changes that we eventually decide on will determine how we manage the 1.6
version of Asterisk. I will be posting much more detailed inform
Eric "ManxPower" Wieling wrote:
> Steve Totaro wrote:
>> Eric "ManxPower" Wieling wrote:
>>> Steve Totaro wrote:
Steve Totaro wrote:
> I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it
> worked fine except for audio issues that I believe are directly related
>
And is there a way the automon can send the result to voicemail? I
hadn't found that yet.
Moj
Reggie Payne wrote:
> Ok. I know you have to use touch monitor but what I am after is the
> variables that need to be specified and where in the extensions.conf to
> configure for users?
>
>
Thanks. It make perfect sense. I was just curious why the manager app is
needed. Since the phone can see 4 AP at the same time, when it wants a call to
be handed over to a different AP, couldn't it just send a re-invite to Asterisk
and call it a day?
>Wai,
>
>The IP address is really on th
Ok. I know you have to use touch monitor but what I am after is the variables
that need to be specified and where in the extensions.conf to configure for
users?
>>> Brian West <[EMAIL PROTECTED]> 10/10/2007 12:00 PM >>>
Look at features.conf
/b
On Oct 10, 2007, at 10:48 AM, Reggie Payne wrote
Asterisk Project Security Advisory - AST-2007-022
++
| Product | Asterisk |
|+
The Asterisk Development Team has released version 1.4.13.
This release fixes a couple of security issues in the implementation of IMAP
storage for voicemail. One of the issues is remotely exploitable. Any systems
that do not use IMAP storage for voicemail are not affected by these issues.
For m
if you have allow=g729,ulaw and you want to use g729 but the current
channel is ulaw it will pick ulaw over g729 because it wants to
escape doing any transcoding if possible.
The best way to do this is setup different peers with different allow
lines to force the outbound leg to the codec yo
On Oct 10, 2007, at 11:12 AM, Ex Vito wrote:
> On 10/9/07, Senad Jordanovic <[EMAIL PROTECTED]> wrote:
>> zoachien wrote:
>>> Google for mexuar.
>>>
>>> Zoa
>>
>> Or look at one that works with MS Windows, Linux or Apple
>> http://www.bicomsystems.com/products/C/P/319/382/
>>
>
> FYI, Mexuar's
On 10/9/07, Senad Jordanovic <[EMAIL PROTECTED]> wrote:
> zoachien wrote:
> > Google for mexuar.
> >
> > Zoa
>
> Or look at one that works with MS Windows, Linux or Apple
> http://www.bicomsystems.com/products/C/P/319/382/
>
FYI, Mexuar's solution -- Corraleta SDK -- *works* with
win, linux a
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I have license for g729a audio codecs and I would like user to use them and
when the limit of 10 is reached, I would like the others to use ulaw...
Do youu know how to do it...
I have put:
allow=g729,ulaw
disallow=all
But ulaw is always chosen
H
Look at features.conf
/b
On Oct 10, 2007, at 10:48 AM, Reggie Payne wrote:
> Hello All! I am new to the list. Does know how to record a call
> on demand? What I would like to do is setup something that during
> a call someone can hit a button a the call is recorded the after
> the call
Yes, Ben you are right. Asterisk is a B2BUA. When the call passes
through the ingress and egress sip call ids are different. By using
$SIPCALLID I can easily get the sip call id that User A sends. The
question is how to "accessing SIP callid of the INVITE sent to User B"?
By senting Manager interf
Wai,
The IP address is really on the access points, since they are the SIP
part of the solution. Let me see how well I can explain this, The
access points register to a manager application, running on one AP,
and the phones have a hard coded DECT id and register to the same
manager app. The manage
Hello All! I am new to the list. Does know how to record a call on demand?
What I would like to do is setup something that during a call someone can hit a
button a the call is recorded the after the call is over the recording is sent
to their voicemail. Anyone?
Thanks,
Reg
did anyone think about how many concurrent call runs on DS3 and how may call
single asterisk instance can handle ?!
That board does not have any DSP, Who will do trans-coding ? echo
cancellation ?
Well, keep us update
2007/10/9, Tim King <[EMAIL PROTECTED]>:
>
> If it hasn't already been done I
On 10/9/07, Brian West wrote:
> [...] All I did was click edit in frontpage and alert them
> of anonymous publishing priv. were on their servers
> and they called the FBI [...]
I believe you.
The astonishing security holes that were engineered
by MS so their web editing-publishing-br
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Ok, I've downloaded the i386 module and it works, I have the module loaded...
Thanks for the command!!
Rafael Canchola a écrit :
>
> Hi:
>
> You can check the next command: show g729
> and you should see some like this "0/0 encoders/decoders of 2 li
Steve Totaro wrote:
> Eric "ManxPower" Wieling wrote:
>> Steve Totaro wrote:
>>> Steve Totaro wrote:
I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it
worked fine except for audio issues that I believe are directly related
to IAX2 in version 1.2.x. I have four P
I have an Asterisk box behind a firewall at home with an IAX2 trunk to a
provider. When I loose the Internet connection I have to perform an "iax2
reload" to bring the trunk back up. This is because of the firewall
configuration. I do not have a port translation through the firewall so
that the
Hi:
You can check the next command: show g729
and you should see some like this "0/0 encoders/decoders of 2
licensed channels are currently in use"
or
the command show translation
or check the asterisk log may be the module is not for you processor version.
Best Regards
At 09:06 a.m. 10/10/
Andrew Kohlsmith wrote:
> On Tuesday 09 October 2007 14:32:38 Matt wrote:
>> http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
>
> And your point, precisely, is what?
>
> Someone who has a criminal record can't be a technical authority? Someone
> can't have a criminal record without being a
Michiel van Baak wrote:
> On 16:32, Tue 09 Oct 07, Steve Totaro wrote:
>> For a small investment of time and money, you can setup OpenVPN and have
>> your own network with no NAT issues whatsoever. That would be my first
>> choice over IAX.
>
> Or wait till the ipv6 branch is ready for producti
Hope you don't mind I jump in here. I am interested in DECT's handover
of live calls. My question is, does the IP address on the phone change
when moving from on access point to another?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: W
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Hash: SHA1
Good Morning,
Any help would be grateful to help me understanding what's wrong...
I have bought 2 g729a licenses to digium and I would like to have them works...
My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4
processors)
so I h
Luis,
Like Ron, I have tested deploying several different handsets and have
been disappointed. I am currently testing a deployment with a DECT
system by Aastra that uses multiple access points the talk SIP to
Asterisk and DECT to the handset. Being based on DECT they have good
battery life and han
Just as a follow up on this thread, I decided to go for the Digium 412P
quad port card.
Thanks to everyone who commented, positively and negatively - it helped
provide a balanced view in the end.
Julian.
Matt Florell wrote:
> On 10/6/07, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
>> Julian
Hello John,
we have a number of customers using each of the solutions you mention and
they all seem to be working correctly. Unless you need a very unusual or
extremely large setup, my suggestion is to go for the one that better fits
your problem space / usage needs.
I hope this helps
l.
I'm expanding our tiny asterisk setup with a couple of external SIP
phones, and I've just noticed the issue of the callerid not being
displayed on an attended transfer.
This bug seems to deal with it:
http://bugs.digium.com/print_bug_page.php?bug_id=8824
I'm surprised that this hasn't been deal
Hello,
When I click on User menu, I get loading screen status. It runs indefinitely
without showing me
the user list and the user admin menu.
Any thoughts ?
Thanks,
Sanjoy.
Pinpoint customers who are
Hello All,
I've got the following warning messages a couple of days back:
/chan_sip.c: Maximum retries exceeded on transmission for
seqno 1 (Critical Response).
/Have got the warnings repeatedly for one Callid. If maximum retries
have exceeded why should it give me those warnings again n again
Yes - use the manager API to do an Originate by setting variable $CMD to
the shell code you want to execute, and then call a piece of dialplan
where the shellout will be executed through the System( $CMD ) command.
Note that this would enable an attacker to execute arbitrary commands with
http://www.didww.com/ will provide numbers. They even have a neat
test thing on their website where you can set it up to work with your
box. I haven't subscribed to them, but they seem ok.
Here is the voip-info link with the full DID provider list.
http://www.voip-info.org/wiki/view/DID+Serv
Also, how do you acces the second SIP call ID from the dialplan? Any
simple way to do this?
Benjamin Jacob wrote:
> Hello Steve,
> I think Ray was talking more like the following setup (do correct me
> if I am wrong):
>
> User A (SIPcallId1) ---> Asterisk (SIPcallId2) --> User B
>
> In this
Hello Steve,
I think Ray was talking more like the following setup (do correct me if
I am wrong):
User A (SIPcallId1) ---> Asterisk (SIPcallId2) --> User B
In this case, the INVITE SIP callId received by Asterisk from User A is
different to that sent in the INVITE to User B.
I can get User
Thanks for your answer, see details below:
U 10.10.10.10.67:5060 -> 10.10.10.107:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0..v:
SIP/2.0/UDP
10.10.10.67:5060;branch=z9hG4bK0264a8da;rport..f:
"2519494"
;tag=as1d5e5664..t:
..m:
..i: 503f1f3a
[EMAIL PROTECTED]: 102
INVITE..User-Agent: Asterisk P
Vitaly,
Can you provide details of what is going on in the packet capture exactly?
What is the Contact: URI that the peer provides in the 302 Moved response?
What does Asterisk do subsequently?
Cheers,
-- Alex
On Wed, 10 Oct 2007, Vitaly wrote:
> My asterisk should follow 302 redirect which i
My asterisk should follow 302 redirect which it
receives from other sip server(10.10.10.10). By
running network sniffer I see, that asterisk receives
302 answer, but doesn't follow it.
My config is:
sip.conf:
...
[out4]
type=peer
host=10.10.10.10
canreinvite=no
promiscredir=yes
insecure=very
d
Luis,
I strongly recommend that you test the setup before deployment.
I have done a lot of tests with WiFi VoIP, handover, security,
and though I don't have experience with the hardware you mention,
I know WiFi VoIP is very brittle, especially in combination
with WPA and handover. Battery life is
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