I've heard about this, but I really can't seem to find anything on it. I've
got a strange setup that exists only because of firewall issues, and
everything about it seems fine. The setup:
SIP clients - Asterisk (office) - IAX - Asterisk (colocation) - SIP PSTN
Termination
All the extensions I
I used the macro-stdextenion that comes with every Asterisk install, and
added a new option - s-CHANUNAVAIL which then dialled the other server
via IAX. Worked really well and only took a few minutes.
PaulH
On Tue, 2007-08-14 at 23:51 -0700, Nicholas Blasgen wrote:
I've heard about this, but I
Greetings list,
I've been using DUNDi for some time now to prevent calls between users going
out via PSTN if there's no need, set up as follows:
[macro-dundi-e164]
exten = s,1,Goto(${ARG1},1)
include = dundi-e164
[dundi-e164]
include = in-e164
switch = DUNDi/e164
My outbound call macro tries
On Tue, 14 Aug 2007, Nicholas Blasgen wrote:
I've heard about this, but I really can't seem to find anything on it. I've
got a strange setup that exists only because of firewall issues, and
everything about it seems fine. The setup:
SIP clients - Asterisk (office) - IAX - Asterisk
Zeeshan,
First off, if your fear of being sued is what stops you from doing
business then get out of the industry or get over it. Its a risk we all
take everyday (not just in VoIP). You build up a core of Insurance and
Defensive Patents to protect yourself. Risk is just part of doing
On Mon, 2007-08-13 at 16:15 +0530, [EMAIL PROTECTED]
wrote:
Hi,
I have successfully configured DIGIUM card and successfully communicated
through it to the another E1 card running application. Can anybody tell me
does TE120P support MFC/R2 protocol.
As far as I know the card is not the
Hi Mike,
Consider ARA
www.voip-info.org/wiki/index.php?page=Asterisk+RealTime
www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
or you can use dialplan add extension cli command from Asterisk
Manager Interface. see
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action
Hi all,
There is a parameter called nonce included in every register request that
a UA sends to asterisk. I have read sip debug a lot and only found out that
the nonce parameter value which is used in register request was generated
by asterisk server in a previous sip response.
As you can see in
- Rizwan Hisham [EMAIL PROTECTED] wrote:
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=584760da
Authorization: Digest username=bernart48, realm=asterisk, algorithm=MD5,
uri=sip:[EMAIL PROTECTED]:9060, nonce=584760da,
response=948d3923bf2df47eca17c572713af2c7, opaque=
Hi all,
I've just being wondering if Dundi has the same purpose as ENUM. I
don't know much (actually almost nothing) about these technologies. As
far as I know they are a kind of DNS resolver used in the VoIP context.
For example, user [EMAIL PROTECTED] has the extension namber 1001. This
thanx for the reply. what i have understood from ur reply and from googling
is that for every authorisation there is a unique nonce (or new nonce), and
previous nonce is expired. but i have seen in sip debug on my atserisk cli
that :
for the first register request, server sends an unauthorisation
Same problem. I've tested a Linksys/PAP2-3.1.9(LSc) and tried with
INBAND configuration in both, asterisk and linksys EP, and it works.
But, just was a test, dont know if I would let it in INBAND config.
Lastly I tried with INFO in linksys, and rfc2833 in Asterisk, and
works too.., no problem.
On
On 08:33, Wed 15 Aug 07, Ronaldo wrote:
Hi all,
I've just being wondering if Dundi has the same purpose as ENUM. I
don't know much (actually almost nothing) about these technologies. As
far as I know they are a kind of DNS resolver used in the VoIP context.
For example, user [EMAIL
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
Hisham
Sent: Wednesday, August 15, 2007 7:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] why is
You have on your hands a broken UA, since it is not responding to the
changing nonce value.
- Brad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
Hisham
Sent: Wednesday, August 15, 2007 7:52 AM
To: Asterisk Users
I'm trying to install wanpipe on my new 2.6.21-2-amd64 core 2 duo
machine (Debian's amd64 works with EMT64 too) to run Asterisk. I'm
getting compilation errors when trying to install the wanpipe utilities.
We are intending to use a Sangoma A102 card for ISDN30 in the UK.
I've tried both the
Hi,
Is it possible to configure asterisk so it doesn't play MoH from certain
phones?
Regards,
Jan
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Rory Campbell-Lange wrote:
I'm trying to install wanpipe on my new 2.6.21-2-amd64 core 2 duo
machine (Debian's amd64 works with EMT64 too) to run Asterisk. I'm
getting compilation errors when trying to install the wanpipe utilities.
Sangoma says that 2.6.21/22 is not supported yet, just
well in this case im using asterisk as a client(UA) to connect to my other
asterisk server. So is this a bug in asterisk. im using asterisk 1.4.2 both
as a client and server.
On 8/15/07, Watkins, Bradley [EMAIL PROTECTED] wrote:
You have on your hands a broken UA, since it is not responding to
You can define a new class in musiconhold.conf with an empty directory.
Create a directory with nothing in it /var/lib/asterisk/moh/empty
Add this class to musiconhold.conf
[empty]
mode=files
directory=/var/lib/asterisk/moh/empty
Then for the phone' entry in sip.conf add:
musiconhold=empty
Be
William McCloskey [EMAIL PROTECTED] writes:
I need a quick bit of advice from the list.
We purchased an asterisk based phone system back about 6 months ago and
we are using Cisco 7940G phones (I know, not everyone's favorites). We
are using the second line on the phones for paging with a
Gordon Henderson [EMAIL PROTECTED] writes:
On Tue, 14 Aug 2007, Nicholas Blasgen wrote:
I've heard about this, but I really can't seem to find anything on it. I've
got a strange setup that exists only because of firewall issues, and
everything about it seems fine. The setup:
SIP clients
Hi all,
I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1
Doing a select in the CDR table I noticed there are some calls with
billsec greater than duration, duration is always 0 in those calls.
How can this happens ? Am I missing something ?
Tnx in advance
Regards
In sip.conf I set this option:
maxexpiry=10800
(3 hours)... The phones re-register once an hour.
On 8/14/07, Steve Langstaff [EMAIL PROTECTED] wrote:
What did you change?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 14 August 2007
got it working... looks like the tar file is corrupted or something
redownload it again and installed it.
Thanks!
On 8/15/07, Anthony Francis [EMAIL PROTECTED] wrote:
looks broken, is there an apps dir in the source directory?
Mark Quitoriano wrote:
is there a new way to install asterisk?
Hi everyone,
I have been dealing with a certain issue with a particular customer site
for months now. The problem occurs when there is an error with caller
id as shown in the following:
WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error
on channel 'Zap/3-1'
When this
Am Mittwoch, den 15.08.2007, 10:14 -0400 schrieb Lee Jenkins:
Hi everyone,
I have been dealing with a certain issue with a particular customer site
for months now. The problem occurs when there is an error with caller
id as shown in the following:
WARNING[16036]: chan_zap.c:6309
How difficult is it to change the firmware load on a Polycom phone from
MGCP to SIP? I have a number of 500/600 phones and see some used phones
being offered with MGCP installed. I have the SIP firmware but have
never had to migrate between content loads. Any expected gotcha's?
Michael Graves
--
Thanks Mark for your detailed email. I have no plan to hide. I am in
business and am ready to face any challenges. Just wanted to know where I
stand if it comes to deal with patent issues, because there are companies
out there who are going thorough this issue once they started to make good
Can anybody point me in the right direction please? I'm having some issues
getting iaxmodem and hylafax to talk to each other.
I have no doubt that someone has had this type of issue before but I can't
find anything useful in the archives or on Google.
Under RH9, with chan_capi 7.1, Asterisk
Hello All,
I don't understand where is the problem...
I have Callback setup and it works fine when tested within US. Works
fine meaning the DTMF tones are passed when prompted to enter the phone
number. But when I test with some international countries, callback
works but DTMF tones are not
Hi All,
Can anybody send me a complete list of sip events. i know only 3 of those
whihc are register, message-summary, message notification. message-summary
event is causing some problems actually. My client sends a bad-event
response if it recieves a message-summary event in a NOTIFY sip packet.
http://www.faqs.org/rfcs/rfc3261.html
Rizwan Hisham wrote:
Hi All,
Can anybody send me a complete list of sip events. i know only 3 of
those whihc are register, message-summary, message notification.
message-summary event is causing some problems actually. My client
sends a bad-event
At least with my Manager API, I have the ability to simply set a default
event handler and using that I can dump all events as the pass though. Then
I setup a case switch and act on the ones I want.
But the manager events I like are LINKED and HANGUP.
Hi all,
I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1
Doing a select in the CDR table I noticed there are some calls with
billsec greater than duration, duration is always 0 in those calls.
How can this happens ? Am I missing something ?
Tnx in advance
Regards
Ah, I correct myself. I see, you wanted to know the headers for each SIP
packet. Makes a lot more sense now.
On 8/15/07, Anthony Francis [EMAIL PROTECTED] wrote:
http://www.faqs.org/rfcs/rfc3261.html
Rizwan Hisham wrote:
Hi All,
Can anybody send me a complete list of sip events. i know
Anselm Martin Hoffmeister wrote:
Am Mittwoch, den 15.08.2007, 10:14 -0400 schrieb Lee Jenkins:
Hi everyone,
I have been dealing with a certain issue with a particular customer site
for months now. The problem occurs when there is an error with caller
id as shown in the following:
Hi. I've got a working dial plan on my home system but there are problems
with it and I was hoping someone more comfortable with dial plans might be
able to help. In a nutshell here's what I'm currently doing on an incoming
outside phone call
[default]
Set(TIMEOUT(digit)=3
Matthew Harrell wrote:
Hi. I've got a working dial plan on my home system but there are problems
with it and I was hoping someone more comfortable with dial plans might be
able to help. In a nutshell here's what I'm currently doing on an incoming
outside phone call
[default]
Faris Raouf wrote:
The problem is that when I run faxstat, it does not show hylafax connected
to any tty.
You're probably not running faxgetty (and your later comments below
confirm this...)
And when I try to run faxaddmodem (just to see what might happen) and select
ttyIAX, I get an error
Hello,
I have a TDM400P with 4 FXO ports, currently using three. When sending or
receiving calls on this card, there is a nearly constant
popping/clicking sound, it is related to the
echo cancellation?. I adjusted my gains properly, but to no avail. I
even found that setting
Thanks Lee.
No, I'm definitely not running faxgetty - I didn't realise I was supposed to
:-( And no, I'm not using the + version.
Back to square one. At least I'm going in the right direction again!
Only I've sidetracked and am currently trying to use capi4hylafax instead of
iaxmodem which
Is iaxtel still around?
I was not able to go to www.iaxtel.com .
did the address changed?
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I've looked around a bit, and I'm still not sure I quite know what the
state of the union is with regard to configuring SkypeIn/Out and Vonage
services as trunk-side appearances on an Asterisk PBX?
Any good clear pointers?
Cheers,
-- jra
--
Jay R. Ashworth Baylink
So besides the missing ) on line 1, I have some other comments:
1) You should replace your priority numbers with 'n'. Just so much easier
to know that the issue isn't with priority numbers. And typing 'dialplan
show context' is a nice way to see if everything is setup correctly. The
'n' is a
Is there a way for voice media clients (like SIP phones and POTS/PSTN
phones) that connect their call legs to Asterisk to negotiate a common
codec that they both use at their end, so Asterisk doesn't have to
transcode? Asterisk would know which codecs each client can use, and
which each
What is the definition of billsec, just out of curiosity? Seconds since
the 200 OK from both ends / presumed media start?
On Thu, 16 Aug 2007, Jaswinder Singh wrote:
I made same thread few months ago and many people said that they dont have
such records in plain asterisk install ( no
Set(TIMEOUT(digit)=3
Set(TIMEOUT(response)=60
These are missing closing brackets for one thing...
That's just a cut and paste error. The real one has ending parens along with
some other stuff outside of default mode. Didn't even notice that when I
filled in the values
--
I made same thread few months ago and many people said that they dont have
such records in plain asterisk install ( no freepbx ) . I was also using
freepbx when i had this problem . Heres mine :
mysql select count(*) from cdr where billsec duration;
+--+
| count(*) |
+--+
|
Hi Jay,
Skype can be used successfully with the ChanSkype module on supported
platforms (Fedora Core 3, 4 or 5 or Ubuntu 6.04). It's $19USD for a single
personal license, and tends to work quite well. It's not the easiest item to
setup (the OS needs a window manager running on it, and each Skype
3-com Model 3102 IP-Phone / Sip firmware download ?
Has anyone every accomplished such?
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I have 10 SIP trunks that I'd really like to round-robin load balance.
Currently I have a macro that switches between available lines, but there
really must be a function in Asterisk to do this on its own. So my question
is just that, are there any easy ways for Asterisk to either balance between
I have been trying to track this down for a while to no avail.
I have a variety of different SNOM phones (the entire 3XX series) and have
also tried on a variety of different Asterisk versions (pretty much the
whole 1.2 and 1.4 train)
When I Page() phones in Asterisk, I only intermittently get
1) You should replace your priority numbers with 'n'. Just so much easier
to know that the issue isn't with priority numbers. And typing 'dialplan
show context' is a nice way to see if everything is setup correctly. The
'n' is a personal choice, but the longer your application the better.
[EMAIL PROTECTED] wrote:
Hello,
I have a TDM400P with 4 FXO ports, currently using three. When sending or
receiving calls on this card, there is a nearly constant
popping/clicking sound, it is related to the
echo cancellation?. I adjusted my gains properly, but to no avail. I
even
Did not work either...Thank you!
Otis
Michiel van Baak wrote:
On 15:02, Sun 12 Aug 07, OCOSA ListAcct wrote:
exten=5,2,Dial(SIP/supportSIP/support2,2,tr)
Make this line read:
exten=5,2,Dial(SIP/supportSIP/support2,,tr)
That should do the trick
Are there any nice GUIs out there for Asterisk Realtime? Google doesn't
yield much. I spent a day trying to get VoiceOne to work without much
success.
Thanks,
Mike Clark
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First, it seems I have to have a 2 - 3 second wait before the AGI call in
order to get valid CID data. Usually 2 seconds suffices for this one
setup
but during that time the caller has had two rings before the local
extension
has even begun to ring. Is there something I am doing wrong
First, it seems I have to have a 2 - 3 second wait before the AGI call in
order to get valid CID data. Usually 2 seconds suffices for this one setup
but during that time the caller has had two rings before the local extension
has even begun to ring. Is there something I am doing wrong that
On Wed, 15 Aug 2007, Matthew Harrell wrote:
The intent of this sequence is to take the incoming callerid, replace it if
known with something in the database, and branch on the state from the DB
and time of the day.
FWIW: I do something similar, but purely in dial-plan using the astdb -
First, it seems I have to have a 2 - 3 second wait before the AGI call in
order to get valid CID data. Usually 2 seconds suffices for this one setup
but during that time the caller has had two rings before the local extension
has even begun to ring. Is there something I am doing wrong that
: Check out the CDR configuration. I do my CDR via MySQL and I don't think
: that does buffering, but I know for sure the normal CSV format (and standard
: configuration file) has options for buffering before saving. I can't really
: think how that would change recieving the CDR information
Wait(2) is what I do.
Matthew Harrell wrote:
First, it seems I have to have a 2 - 3 second wait before the AGI call in
order to get valid CID data. Usually 2 seconds suffices for this one setup
but during that time the caller has had two rings before the local extension
has even begun to
Thanks. I was hoping there might be a way to detect whether the CID
routine was done or not. I've still seen occasions where it wasn't available
for callers that I know had it. Maybe my phone service is just a little
slow sometimes
Wait(2) is what I do.
Matthew Harrell wrote:
First, it
William McCloskey wrote:
The stability problems we have seem to be related to asterisk crashing
the apache install on the box when the PHP scripts are performing
functions via asterisk. Don't know exactly how they work it all, but
that's the gist of it.
Are the PHP scripts connected with
Try some of these suggestions.
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 15, 2007 12:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] TDM400P FXO
Dr. Michael J. Chudobiak wrote:
Rory Campbell-Lange wrote:
I'm trying to install wanpipe on my new 2.6.21-2-amd64 core 2 duo
machine (Debian's amd64 works with EMT64 too) to run Asterisk. I'm
getting compilation errors when trying to install the wanpipe utilities.
Sangoma says that
Please explain to me how FXO tune would fix popping and clicking sounds???
-Original Message-
From: Matthew Fredrickson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 15, 2007 2:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM400P
shadowym wrote:
Please explain to me how FXO tune would fix popping and clicking sounds???
If they are caused by a poorly-tuned echo canceller.
-Stephen-
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On 15/08/07, Stephen Bosch ([EMAIL PROTECTED]) wrote:
Dr. Michael J. Chudobiak wrote:
Rory Campbell-Lange wrote:
I'm trying to install wanpipe on my new 2.6.21-2-amd64 core 2 duo
machine (Debian's amd64 works with EMT64 too) to run Asterisk. I'm
getting compilation errors when trying to
I am trying to update a machine with a TE210P card setup as PRI. Running
Centos 4.4.
I stop asterisk, I do service zaptel stop. I look at lsmod and all
zaptel modules are unloaded.
I compile zaptel 1.2.19, I install zaptel.
when I do the service zaptel start, the machine locks up.
I reboot the
Does anyone online have an opinion on these? I've used 500/510/6001/601
models before. Need to know if these apparently lesser models can be
provisioned in the same way. Are end uers happy with them?
Thanks,
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product
One short warning - I know of a company that tried to setup a larger
installation of ChanSkype, and it didn't work very well at all. (huge
memory use, crashes, lockups)
PaulH
On Wed, 2007-08-15 at 16:15 -0400, Alex Robar wrote:
Hi Jay,
Skype can be used successfully with the ChanSkype
I did some testing on the 430, and was pretty happy with it.
Paul Hales
AsteriskIT
On Wed, 2007-08-15 at 20:52 -0500, Michael Graves wrote:
Does anyone online have an opinion on these? I've used 500/510/6001/601
models before. Need to know if these apparently lesser models can be
I have deployed a couple 330's and they will use the same provisioning
methods as the earlier models from Polycom, you just need to be sure and
have the firmawre and configs for that version. The other thing I ran into,
is the 330 did not ship with a power supply so you either go POE or buy the
On Wed, Aug 15, 2007 at 09:22:45PM -0400, Jerry Geis wrote:
I am trying to update a machine with a TE210P card setup as PRI. Running
Centos 4.4.
What is the output of:
uname -r
I stop asterisk, I do service zaptel stop. I look at lsmod and all
zaptel modules are unloaded.
I compile
hot to asterisk multiport...???
example 5060, 5061, 5080
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Off the cuff, I can't recall if asterisk can listen for (in this case
I assume) SIP on multiple ports. It would be quite easy to do this
redirection with iptables, though.
On 8/15/07, Walter Willis [EMAIL PROTECTED] wrote:
hot to asterisk multiport...???
example 5060, 5061, 5080
--
Erik
Hi, folks:
I've come into some Mitel 5020 IP phones. A client has made a
significant investment in them and we want to see if we can use them in
a new system.
Are these even SIP sets? I haven't been able to find out. Mitel's site
barely covers them (I was only able to find some user guides,
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