If your BG 101 is in intranet, try to adjust your qualify parameter to
60.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Matthew B
Marlowe
Enviado el: viernes, 27 de febrero de 2004 2:08
Para: [EMAIL PROTECTED]
Asunto: RE:
Hi,
good day i just install successfully asterisk and when i try iax client
to connect to my asterisk server im getting a Call reject by Remote
this is the content of my iax.conf:
register = test:[EMAIL PROTECTED]
[test]
type=friend
secret=mypass
deny=0.0.0.0/0.0.0.0
Stephen,
Thanks for the suggestion but my problem is with inbound calls from the PSTN (coming
in via a AS5300) into the SIP based platform and how the * chan_sip identifies that a
PSTN originated call should have the number withheld or not.
Rgds,
Adam
-Original Message-
From: Steve
Impressed. Does some countries have laws on SIP implementations? Wow. ;-)
We operate a large traditional telephone network in several countries and as I am sure
you are aware lawful intercept is a requirement on traditional networks. We've
extended our network to provide VoIP gateways
I need some tips on configuration of voicemail with mysql...
here is my voicemail.conf
**voicemail.conf***
[general]
dbhost=localhost
dbname=asteriskvmusers
dbuser=root
format=wav
serveremail=asterisk
attach=yes
- Original Message -
From:
Frederic
Olivie
To: [EMAIL PROTECTED]
Sent: Thursday, February 26, 2004 2:04
PM
Subject: [Asterisk-Users] Connecting an
ISDN DECT phone base
Hi,
I own a Siemens 3070 DECT system.
It's a simple DECT base which allows
On Fri, 2004-02-27 at 11:28, Frederic Olivie wrote:
I own a Siemens 3070 DECT system. It's a simple DECT base
which allows the connection of a few DECT phones. It's a very
basic PBX. It's connected to the public network using an ISDN
bri (2B + D) plug.
How about 120? Look here:
http://www.voip-info.org/tiki-
index.php?page=Asterisk+setup+medium+office+100
I've set up 75 extensions... I'm 100. Sorry.
Would anyone care to share some experience with big installs, ie.
multiple PRI's and excess of 100-200 extensions.
Thanks
Rob
Hi!
Even though it was 100, I'm also keen to hear about large installs,
what kind of experience did you have setting it up, and what hardware
for the * server did you use?
This might help if you are interested in no. of concurrent calls
instead of number of extensions/phones:
Hi!
Can build a switchboard with TDM400P + X100P?
I need a receptionist to pick up the incoming calls and transfer them to
appropriate employee.
You might want to read the handbook draft:
http://www.digium.com/handbook-draft.pdf
Do I need those Nortel telephones for this or Panasonic KXTD
Low, Adam wrote:
Could you please point me in direction of standard documents, drafts or documentation of this?
IETF specification, draft-ietf-privacy-.02.txt, SIP Extensions for Caller Identity and Privacy.
Thank you for the pointer, as this is still a draft (a lot of SIP things are), it's
I am not a programmer so can not implement this, but I think it may be
useful.
Asterisk configured to listen on multiple IP addresses,
Then configure RTP ports for each address independently;
So I open 5 ports on one IP and then forward those ports to that IP
from my firewall.
Then on another
Hi!
good day i just install successfully asterisk and when i try iax client
to connect to my asterisk server im getting a Call reject by Remote
register = test:[EMAIL PROTECTED]
host=10.1.1.2
Registration makes only sense - and only works - if you have
host=dynamic. The sole purpose of
Well I am in mostly a Cisco enviroment and it seems that it is supported on both IOS
12.3(4)T for the AS5300 and the SIP6.2 image on our 7940's. I've not tested any other
SIP stacks but maybe others can offer some added input there ?
Ok I'll submit it to bugs.digium now ...
-Original
On Thu, Feb 26, 2004 at 11:11:05PM -0500, Alex Volkov wrote:
You must have started asterisk with asterisk -c
No, I started it with asterisk and had it running in the background.
Then, per the PDF manual, I did asterisk -r to connect to the server
and get a console. The manual says I can type
On Thu, Feb 26, 2004 at 11:01:40PM -0500, Chris Clifton wrote:
Greg,
There may very well be another way to detach from the console, but I start
asterisk on tty5 or tty6, and leave it running there. (redhat gives you 6
console tty's by default, use [alt] + [f1,f2,f3,etc.] to switch) You can
Greg Kedrovsky wrote:
You must have started asterisk with asterisk -c
No, I started it with asterisk and had it running in the background.
Suggest starting as 'safe_asterisk'
asterisk -r
exit
Always works for me...
F
___
Asterisk-Users mailing list
Hi,
Did HT-286 Bypass calls from normal PBX and Asterisk PBX to analog phones ?
To be more precisely, can i receive both call from this two kind of tecnologies using HT-286 in my office ?
I dont want change my OLD PBX (That works great) with Asterisk and lose investiment etc.
So i think in use
Greg == Greg Kedrovsky [EMAIL PROTECTED] writes:
Greg I started it with asterisk ... Then ... I did asterisk -r
Greg to ... get a console. The manual says ... type quit to
Greg disconnect ... But, [it didn't work] ...
What version of *? With recent cvs it works. Or at least exit
works.
On Fri, Feb 27, 2004 at 01:20:28PM +, Fran Boon wrote:
Suggest starting as 'safe_asterisk'
asterisk -r
exit
Thanks. Worked like a charm.
-Greg
--
Mutt 1.4.1i on Slackware 9.1 Linux
Curridabat, San Jose, Costa Rica
http://www.greg-and-sue.com/screenshot.jpg
Yahoo Instant Messenger
A number of IAX Phone users have reported a bug which causes the call to
drop the audio stream after 65 seconds. The issue only seems to occur when
both parties to the call are using IAX Phone or another iaxClient-based
phone (DIAX, iaxComm, etc.) and when one or both legs of the call traverse a
Hi,
Did anyone know if exist some adapter that give me the option to connect two kind of tecnologies ?
Something like with 1 RJ-45 port 1 RJ 11 Port (IN), and 1 RJ 11 port (OUT).
Then i can join my old PBX that works perfectly with Asterisk that works great too (But in voip mode) with my analog
I've been testing a nice little box that has precisely what you requested. Its made by
Aethra (Spain) I believe and know as the VIP3001 or VIP3002 and it runs both SIP/H323
and allows you to select if you want to send calls of the VoIP or over the PSTN. It
works great with Asterisk running SIP.
I like putting a TxxxP in your * system and connecting the systems via a
T1 cross over cable.
Hi,
Did anyone know if exist some adapter that give me the option to connect two kind of
tecnologies ?
Something like with 1 RJ-45 port 1 RJ 11 Port (IN), and 1 RJ 11 port (OUT).
Then i can join my
Oops.
I forgot to include a link to a new DLL that is required in order for the
new version of wiax2.dll to operate. Very sorry about that. Here are the
links to the _2_ dll files you need to download and copy into the working
folder for IAX Phone:
I can't believe you would add anymore digits to listen for.
I have thought about speeding up the digit play back.
It seems to take forever when waiting for 7.0.1 smile
Jim Sneeringer wrote:
Actually, it works fine as long as the parkpos values are numbers. If you
put in a * or #, it
Hello. I'm running an asterisk system where the voicemail box numbers
match the extensions to which they belong. The phone numbers from the PSTN
which access the system are mapped to specific extensions, and if there's
no answer, they forward to their respective mailboxes so callers can
I have some mobile users that would prefer to have a 'real phone' instead
of a computer headset. I've been looking around at the USB phone setups,
which is (it seems) simply a softphone with a USB handset. The only
ones I've found seen to be locked to a particular service provider.
Has anyone used
Title: RE: [Asterisk-Users] Lucent Definity CallerID {Scanned}
This doesn't seem to be it, maybe its the definity release I am using but this seems to be set up properly. There must be a flag elsewhere that doesn't pass internal extentions cid informaiton. Any more suggestions?
Matt
Title: Agent Queuing on multiple machines
Hi,
I was wondering if anyone had any experience with agent queueing on multiple machines, because of redundancy in our solution I'm not sure which machine the agents will queue to since they need to log in over zap channels, and which machine the
Has anyone used these, and are there any ones that can work as a
general softphone, like X-Lite?
I have been using an IPP200 from Eutectics
http://www.eutectics.com/
It works with most softphones (it's just a USB audio device with an
additional set of libraries for monitoring the hook state).
Again, I hosed up some thing. Wrong URL. Here's the proper URL for
Eutectics:
http://www.eutecticsinc.com/usbPhones/usbPhones.html
Perhaps I should try sleeping. They say it's good for you...
Steven Sokol
Owner/Manager
Sokol Associates, LLC
Phone: 816.822.1807
IaxTel: 700.613.9004
Web:
I did come across a PDF explaining how to set up a cisco 3600 series gateway
with a Definity. Maybe it would help. Here is the link
http://www.cisco.com/application/pdf/en/us/guest/products/ps278/c1237/ccmigration_09186a00800e7631.pdf
-Art
- Original Message -
From: Matthew Branton
On Fri, 27 Feb 2004 10:52:29 -0500, Tim Sailer [EMAIL PROTECTED] wrote:
I have some mobile users that would prefer to have a 'real phone' instead
of a computer headset. I've been looking around at the USB phone setups,
which is (it seems) simply a softphone with a USB handset. The only
ones I've
On Fri, Feb 27, 2004 at 10:09:49AM -0600, Steven Sokol wrote:
Again, I hosed up some thing. Wrong URL. Here's the proper URL for
Eutectics:
http://www.eutecticsinc.com/usbPhones/usbPhones.html
I figured it out. :)
Perhaps I should try sleeping. They say it's good for you...
Really? I
I am really a newbie on *, but I think that you can answer the line and wait
some time (like 2 seconds) if the caller dont press anything like * (for
example) he will be moved to the voicemail, but if he press, he will go to
VoicemailMain to check their messages.
Somebody correct me if necessary.
Is it possible to send SIP NOTIFY messages to * users through asterisk
through an external application? Does that external application have to
be a registered * user in order to send the NOTIFY message to other
users. I have tried sending unsolicited NOTIFY messages to * but the
application
I had my first production system Asterisk crash today with no apparent
reason for the crash. This was on a production server that hasn't had
anything changed on it for 3 weeks and is rebooted every night. The load was
low when the crash occured and the logs give no indications as to what
caused
mattf wrote:
I had my first production system Asterisk crash today with no
apparent reason for the crash. This was on a production server that
hasn't had anything changed on it for 3 weeks and is rebooted every
night. The load was low when the crash occured and the logs give no
indications as
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello everybody,
has anybody managed to call a (old fashioned) phone using Sipgate.de and
asterisk? (yes I have money on my account :-) )
The configuration I got from the sipgate.de people is at the botton of
the mail
Here is mine:
sip.conf:
PS: You just got the driver only option?
[Steven Sokol]
Yep. I did order the API as well. They make you sign an NDA (pretty basic
one). The API covers the hook-switch integration and the keypad integration
for their IPP5xx series phones.
What client are you going to use it with?
Regs,
I have a need to purchase a 2-4 port FXO gateway for use with *. I have no
PCI slots left in my * machine so I can't use a X100P. So what is the best
FXO gateway to get?
Thanks,
Scott
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Try typing an ! followed by the enter key at the CLI prompt amd see what
happens.
- Original Message -
From: Fran Boon [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, February 27, 2004 7:20 AM
Subject: Re: [Asterisk-Users] exit
Greg Kedrovsky wrote:
You must have started
Hi,
I would be tempted to get rid of the slash and number on the register line,
unless your asterisk extension is 02115800.
dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
Sent: 27 February 2004 16:47
To: [EMAIL PROTECTED]
Subject:
Ed Devine wrote:
Try typing an ! followed by the enter key at the CLI prompt amd see
what happens.
That only drops you to a prompt. It doesn't exit the console session
that was active.
Unless you're intending to run asterisk not as an actual background task
(your session looking at the actual
The Mediatrix Gateways work with Asterisk, however, no gsm support.
Thanks
-Matt
TelCom Products International
2901 Frontage Road S Hwy 10E
Moorhead, MN 56560
Phone# 218-422-9004
Fax# 218-422-9014
Support on MSN Messenger [EMAIL PROTECTED]
- Original Message -
From: Scott Weis [EMAIL
Title: RE: [Asterisk-Users] Lucent Definity CallerID {Scanned}
Yeah this combined with the earlier information did it, I think most of the confusion stemmed from running definity release 6. Now it works though, its too bad it can't be set on a per trunk basis though. Thanks very much for the
On Fri, Feb 27, 2004 at 10:48:00AM -0600, Steven Sokol wrote:
PS: You just got the driver only option?
[Steven Sokol]
Yep. I did order the API as well. They make you sign an NDA (pretty basic
one). The API covers the hook-switch integration and the keypad integration
for their IPP5xx
Hi Birk
I´m messing arround for the last 2 day with sipgate.de. My latest
configuration seems to work only when X-lite is running on a PC on my
lan (!!!) and tried to play a call. So I think that there must be some
authentification problem or so...
When x-lite in not running I also get: 403
A cisco 1760 router, with a pair of dual FXO cards in it will work
fine. We've been using a couple of these for years, and they're quite
reliable, sound good, and behave themselves with Asterisk, using SIP. Not
the cheapest, perhaps, but a good choice.
If you want to save money,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi David,
no the number after the slash is necessary (and yes this is my number)
Without that slash/number I'm not able to get a call anymore.
But thanks
Birk
David J Carter wrote:
| Hi,
|
| I would be tempted to get rid of the slash and number
Hi!
has anybody managed to call a (old fashioned) phone using Sipgate.de and
asterisk? (yes I have money on my account :-) )
extension.conf:
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
Try this instead:
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
Philipp
Abraham Lincoln wrote:
Hi,
good day i just install successfully asterisk and when i try iax
client to connect to my asterisk server im getting a Call reject by
Remote
this is the content of my iax.conf:
register = test:[EMAIL PROTECTED]
[test]
type=friend
secret=mypass
I know, I should reply to myself, but I just realized this...
Andrew Thompson wrote:
Abraham Lincoln wrote:
Hi,
good day i just install successfully asterisk and when i try iax
client to connect to my asterisk server im getting a Call reject by
Remote
this is the content of my
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello Philipp,
whis also did not help - still a:
- -- Got SIP response 403 Forbidden back from 217.10.79.9
But thanks (do you have working configuration?)
Birk
Philipp von Klitzing wrote:
| Hi!
|
|
|has anybody managed to call a (old fashioned)
Is there english version of their sipgate.de website?
-D
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Birk Bremer
Sent: Friday, February 27, 2004 7:06 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Anybody managed to call a phone
On Thu, Feb 26, 2004 at 09:18:45AM +0800, [EMAIL PROTECTED] wrote:
In the Makefile inside asterisk/channels/h323 directory, there's a line like
this:
CFLAGS += -I$(PWLIBDIR)/include/ptlib/unix -I$(PWLIBDIR)/include
try to use -I$(PWLIBDIR)/include ONLY, it should work. I've compiled it
hi...
I was playing with g726 and budgetones, here's
my quick experience:
* firmware 1.0.4.40 ... the phone just crash:
as soon as you start a call in g726, only a
squeeze is heard, all the display icons are lit
and the phone is dead :)
* firmware 1.0.4.46 : the phone survives, but the
I've posted this as a bug:
http://bugs.digium.com/bug_view_page.php?bug_id=0001124
And I found this site very informative about core dumps:
http://turing.gcsu.edu/~adimitro/viewcore/
MATT---
-Original Message-
From: Andrew Thompson [mailto:[EMAIL PROTECTED]
Sent: Friday, February 27,
Hello.
I'm very new to Asterisk, and so far I've gotten the Digium FXO card to
function correctly with a SIP phone. We're looking at running this at my
company, and we already have a few Eicon DIVA Server T1/PRI cards. I was
wondering if anyone had experience with setting this up ( or generic
Does anyone actually have a 2.4 version of gnophone that will compile?
All the copies on the ftp site have a corrupt file, as does CVS...
Tim
--
Tim Sailer Coastal Internet, Inc.
Network and Systems Operations PO Box 726
We have a situation where voicemail coming in (i.e.
FXO-Asterisk-Voicemail) through a Mediacodes MP108-FXO are getting cut
off a couple of seconds early. I recall a thread about this quite a while
back where this was happening due to silence detection on ZAP channels...
Has anyone experienced
Hi,
So it's like this. I've had siproxd working for me on an external host to
which I've established a tunnel (my SIP client is behind a NAT gateway).
Of course, I've got to have mailbox functionality at the very least, so a
friend of mine told me about Asterisk, which I grabbed from the CVS and
I would like to know if anyone has run into this problem. After
upgrading to the new 2.03y version for the Snom 200 all my mapped keys
have the lights on. They do not go off. The upper MWI is off unless
you get a call or you have voice mail waiting. But the 5 side lights
don't go off.
All
Just use control-c, you will be able to exist and leaving asterisk
continue to run in the background.
--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002
smime.p7s
Description: S/MIME Cryptographic Signature
Has anyone backended a Fujitsu 9600 with an asterisk system? Does
anyone know anything about Fujitsu's em link signaling interface (T1)?
Mike
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
In the contrib/scripts directory I have been trying
to figure out the format of the entries in the MySQL table. I had seen
several posts from a while back, but everyone seemed to understand what I am not
getting.
The info in the file itself
(retrieve_sip_conf_from_mysql.pl) says to make a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Here's how I did it:
exten = 1305/1231231305,1,Macro(checkvm,isc,${EXTEN})
exten = 1305,1,Macro(stdexten,isc,${EXTEN},${PT},SIP/ISC_0007853569F1_1)
Then I set up the Cisco conf file to have the extension dial, so pressing the
messages button calls
hi guys finally i got my wisip this week and im very happy with it. It
works but i was wondering anyone know where can i find new firmware,
updates or a wish list? I cross emails with jeff pulver about having a
small http browser for auth on starbucks hotspots mcdonalds or prodigy
movil(mexico).
I'm still in a test mode with a new Mediatrix 1204 fxo gateway, and been
having an issue with the 1204 properly detecting callerid.
Two pstn lines installed, both with callerid.
One pstn line rings with a standard US ring (long ring)
Second pstn line is a CO Centrex and rings with a long+short
On Fri, 27 Feb 2004, Michael Graff wrote:
Here's how I did it:
exten = 1305/1231231305,1,Macro(checkvm,isc,${EXTEN})
exten = 1305,1,Macro(stdexten,isc,${EXTEN},${PT},SIP/ISC_0007853569F1_1)
Then I set up the Cisco conf file to have the extension dial, so pressing the
messages button
Can someone refer me to an example of an automated broadcasting
operation that sends a canned voice message to a list of phone #'s?
--
Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED]
Thanks for putting up with my spam filter!
After struggling with the carrier access channel bank for a few weeks,
I finally gave up on it, and got myself three X100P cards instead, for
my incoming lines. The plan is to use the channel bank just for
internal lines.
I installed the cards and at first they were mostly OK, except
Hi,
Does anyone know how to check the status of a queue from within
extensions.conf. If a queue has no one logged into it I want to redirect the
call to a manager phone.
Any ideas would be appreciated.
Thanks
John Bittner
Simlab.net
___
Check out the sample call file in the source directory. You can get the
system to call a number and then connect it to an internal extension.
The extension can be set to play a file and then hang up. If you cannot
find the samples, get a hold of me and I will send you something.
Darren
Hi, all
I wonder when passing calls through asterisk with H323, is there anyway to
find out what codec the calls are using, anyone can help please, thanks alot
!
TC
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.590 / Virus
Hi,
Whats involved in getting H323 working on Asterisk with Redhat 9???
Cheers,
Carl.
One way to do it is to use a sniffer, such as ethereal, to capture the
traffic. You should see it in capability exchange, but also easily see in
RTP packets. There might be better ways. But if you're interested in
pursuing it this way and not sure how to do, please follow up with another
Has anyone had a similar issue with Asterisk
Voicemail being unable to detect the digits sent from an SJ Phone connection. I
have included dtmfmode=inband and it works fine when calling other phones though
not with Voicemail. Voicemail doesn't regonise the password.
Is there a way to not
I can't offer you an explanation Rob, only thanks.
We were going nuts trying to track this with SIP
debugging, when in fact we had exactly the same problem on two mailboxes. In our
case it was msg0015.txt causing the MWI to stay lit.
-Darren
-- Darren NickersonSenior Sales Support
Nate Carlson wrote:
Caller ID to work. I searched the archives, and found some people saying
that outgoing Caller ID shows up as Out of Area (that's what I get), and
another person saying it worked 75% of the time for him. I've tried
calling 3 different area codes (612, 952, and 253), so I've
Anyone here with experience on the Cisco ATA 188 and *?
Is it as good as ATA 186?
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To UNSUBSCRIBE or update options visit:
Is Asterisk capable of handling video
conference? I am wondering if there is anybody in the list who tried it
with NetMeeting(s). If it is possible, is the * required to register in
the GK for this purpose? or making it as h323gw only is enough.
Has anyone had a similar issue with Asterisk Voicemail being unable to
detect the digits sent from an SJ Phone connection. I have included
dtmfmode=inband and it works fine when calling other phones though not with
Voicemail. Voicemail doesn't regonise the password.
I am using SJPhone, and
Ron McMillan wrote:
One way to do it is to use a sniffer, such as ethereal, to capture the
traffic. You should see it in capability exchange, but also easily see in
RTP packets. There might be better ways. But if you're interested in
pursuing it this way and not sure how to do, please follow
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