On Jun 18, 2004, at 1:05 PM, Steven Critchfield wrote:
On Fri, 2004-06-18 at 11:06, Artur Jasowicz wrote:
I am running asterisk on an old PowerComputing Mac clone running
YellowDog 3.0 (Red Hat clone for PowerMacs) I've decided to try adding
a generic winmodem card and compile zaptel-0.9.1 for it.
Hi,
No matter what I set
in my sip.conf, I always get '3' as amaflags in my mysql
cdr.
(a) How do I make
amaflags correctly set in mysql cdr
(b) Seperate note,
how can I set amaflags from agi
Thanks
SW
Kyle Hagan wrote:
Brian Buhrow wrote:
Hello. I've seen this behavior. What happens is that the
Grandstreams forget to continue registering with Asterisk after a
while. I
bet when you find this happening, that sip show peers doesn't show
ext/ext
ip address for the one that isn't working.
Title: Message
I am looking to
install a new PBX into a small business. We have 18 internal extensions,
and 6 phone lines. I have been looking at Asterisk as a possible solution
and would like to hear from people already using it. Digium recommended I
post to this list for responses.
My
I've just finished a general purpose configuration utility for the GS
phones:
1) Generates files from scratch (using MAC), from HTML config listing,
or by directly downloading from the phone.
2) Does multiple simulteneous edits.
3) Can reboot as many or as few phones at a time as you like.
-Messaggio originale-
Da: Kevin Walsh [mailto:[EMAIL PROTECTED]
I don't quite understand your Caller*ID dilemma.
In your sip.conf, you'd have a block for each user, say [abc123].
That's your random username, yes? The same block would also
define the password and other directives.
I'm trying to compile cdr_addon_mysql but keep getting this error. Again, searching
the Wiki and the mailing list archive didn't bring up anything useful. Any help? Yes,
I'm using MySQL 4.0. Maybe I have to switch back to 3.23?
# make
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql
I've got everything up and running, but I've hit an
issue and I'm not sure how to go about resolving it.
Here's our config:
LG GDK-186 PBX --PRI--- TE405P/Asterisk
---PRI--- Telewest (Telco provider)
I can make and receive calls from the GDK, and
make/receive calls from the VOIP
From: Stephen R. Besch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 18, 2004 10:13 PM
Subject: [Asterisk-Users] Grandstream CFG file generator
I've just finished a general purpose configuration utility for the GS
phones:
1) Generates files from scratch (using MAC), from HTML
I've the same problem with the Cisco ATA's and Cisco 5300. The cisco
sends the: SIP/SDP Status: 183 Session Progress , with session
description, asterisk forwards is to the phone: SIP/SDP Status: 183
Session Progress, with session description after that the SIP Phone stops
ringing.
People
I read in the archives a post from last year about the Dialogic drivers not
being free for use with Linux/Asterisk.
So, I have a VFX/41JCT-LS to try with *
Suggestions? Purchase digium boards is not an option. We want to test the
app before buying any other hardware.
thanks,
erick,
hi,
i'm attempting to develop an enhanced privacy manager using AGI.
the flow is roughly as follows:
1) intercept calls for users w/privacy manager enabled and execute
an AGI.
2) play a notification to the caller, and prompt them for their
name, recording their voice.
3) loop an awaiting
On Fri, 2004-06-18 at 15:39, Erick Perez wrote:
I read in the archives a post from last year about the Dialogic drivers not
being free for use with Linux/Asterisk.
So, I have a VFX/41JCT-LS to try with *
Suggestions? Purchase digium boards is not an option. We want to test the
app before
Thanks for the advice, checked it with a VOM and sure enough, it had
remote disconnect supervision coming down the line, but we didn't have
the dip switches on the FXO card on the ADIT switched correctly. Once we
did that all is good.
Thanks again.G
Gregory P. Scasny
Golden Technologies
Furthermore, even if you assumed that spandsp was as stable as HylaFAX,
there is a vast feature-set difference between them as far as the
faxing itself goes. Steve has already made it clear that he sees no
future in fax, and that he does not intend to bridge that feature-set
gap at all.
sorry but i did not understand your answer.
Erick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Friday, June 18, 2004 4:00 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dialogic VFX/41JCT-LS Linux RH9 Drivers
On Fri,
If you read the wiki... disabling subscribe to MWI should fix this
problem.. it did for me.
-Original Message-
Kyle Hagan wrote:
Brian Buhrow wrote:
Hello. I've seen this behavior. What happens is that the
Grandstreams forget to continue registering with Asterisk after a
Don't reply all to the list please. I do not want a copy of your mail in
my INBOX when I get a second copy via list mail in the proper folder.
On Fri, 2004-06-18 at 16:22, Erick Perez wrote:
sorry but i did not understand your answer.
fleaBay == eBay
eBay an item means to sell it.
I suggested
ok Steven, so i dump dialogic. but my question remains. Are there any
free-available linux drivers for the * pbx/dialogic or do i really have to
dump my card.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: Friday, June 18, 2004 4:22 PM
I am new to asterisk I just downloaded it I setup some extentions I can't
seem to get them to ring I can get my ata 186 to register but having
problems with getting the phones to ring when I dial an extention
Extentions.conf
[dstech4]
exten=104,1,Answer
exten=104,3,Goto(dstech4,104)
[dstech3]
In article [EMAIL PROTECTED],
Manuel Wenger [EMAIL PROTECTED] wrote:
I'm trying to compile cdr_addon_mysql but keep getting this error. Again, searching
the Wiki
and the mailing list archive didn't bring up anything useful. Any help? Yes, I'm
using MySQL
4.0. Maybe I have to switch back to
Andrew P Cook wrote:
I am looking to install a new PBX into a small business. We have 18
internal extensions, and 6 phone lines. I have been looking at
Asterisk as a possible solution and would like to hear from people
already using it. Digium recommended I post to this list for
responses.
In article [EMAIL PROTECTED],
Jon Radon [EMAIL PROTECTED] wrote:
If you read the wiki... disabling subscribe to MWI should fix this
problem.. it did for me.
I have both my phones with subscribe to MWI disabled, running 1.0.4.68,
and one of them still occasionally stops registering, but can
Erick Perez [EMAIL PROTECTED] wrote:
ok Steven, so i dump dialogic. but my question remains. Are there any
free-available linux drivers for the * pbx/dialogic or do i really have
to dump my card.
I believe that Dialogic supply a Linux driver. I don't think it's free
though (binary only). I
Jeremy Kenney wrote:
I am new to asterisk I just downloaded it I setup some extensions I
can't seem to get them to ring I can get my ata 186 to register but
having problems with getting the phones to ring when I dial an
extention
Extentions.conf
[dstech4]
exten=104,1,Answer
i guess i am not being sufficiently clear.
the sipura seems to act one way and the cisco another. the
sipura, x141, is happily served by
[in-206-sipura]
exten = s,1,SetVar(areacode=206)
exten = 141,1,GoTo(in-int,s,1)
[in-int]
exten = s,1,Answer
exten =
I was able to get app_prepaid working, but unfortunately I am getting one
way audio on the phone that I was placing the call from. It is behind NAT.
It appears that the app_prepaid is not taking this into consideration since
I see:
Jun 18 17:46:25 DEBUG[1133742896]: chan_sip.c:4130 build_route:
Hiyall.
Just wondering how people test your emergency dialing in the UK.
Obviously you need to dial the 999 for emergency services, but am a bit
unsure if this would go down too well with the operator with a 'sorry
just testing' call. (you do all /test/ your emergency dialing dont
you!?:-) )
Issue has been resolved.
The short answer:
The format of caller ID information on em_w trunks (maybe zap channels
too) in zapata.conf can make it so the sipura will not anwser a call.
Calls to the Sipura 2000 work fine from another sip device connected
through *, from either an fxo or fxs
On Fri, 2004-06-18 at 16:35, Erick Perez wrote:
ok Steven, so i dump dialogic. but my question remains. Are there any
free-available linux drivers for the * pbx/dialogic or do i really have to
dump my card.
If you read the discussions in the archive about the dialogic drivers
you will find out
On Fri, 2004-06-18 at 16:35, Erick Perez wrote:
ok Steven, so i dump dialogic. but my question remains. Are there any
free-available linux drivers for the * pbx/dialogic or do i really have to
dump my card.
List of Asterisk supported hardware is at
You just need to ensure that the first thing you say is This is an engineer
making a 999 test call and clear down as soon as they have confirmed.
-Original Message-
From: Wayne [mailto:[EMAIL PROTECTED]
Sent: 18 June 2004 23:07
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Testing UK
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Found user 'dstech3'
Looking for 104 in dstech3
Reliably Transmitting (NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.50:5060;received=68.61.52.152
From: Anonymous sip:dstech3@:0;tag=2110378357
To:
Hi Charles,
Blocking the 183 is undesirable, because messages
from the PSTN indicating that e.g. a number has been changed,
will be lost. Instead, do what's necessary to get audio
back to the caller. On the ATA, set bit 19 of ConnectMode
(see table 5-8 of manual). On the 5300, see
On Jun 18, 2004, at 3:02 PM, Randy Bush wrote:
the sipura seems to act one way and the cisco another. the
sipura, x141, is happily served by
[in-206-sipura]
exten = s,1,SetVar(areacode=206)
exten = 141,1,GoTo(in-int,s,1)
[in-int]
Ahh. Okay, I think I see. The Cisco isn't doing
Ahh. Okay, I think I see. The Cisco isn't doing anything weird; the
sipura is. Why is it sending its own extension first?
bingo! thank you. fixed.
the sipura dialplan was hacked and left in a bad state.
randy
___
Asterisk-Users mailing list
[in-206-cisco]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,PlayTones(dial)
exten = s,5,SetVar(areacode=206)
exten = _001,1,SetVar(mailbox=001)
exten = _001,2,Macro(fwd-call,${EXTEN})
exten =
On Jun 18, 2004, at 4:10 PM, Randy Bush wrote:
Unless Asterisk does something weird with it that I haven't seen
before,
then you'll only get 's' in this context if you get the cisco to dial
without specifying a number.
oops! then how do i get a per-incoming-context SetVar?
I've done something
I am using a IPDialog siptone II. I can take incoming calls, but when I try
and make an outgoing call I get a SIP 407 error.
Can some kind soul explain to me what I am doing wrong?
Here's what I found in the wiki:
If a proxy does not accept the credentials sent with a request, it SHOULD
hi;
i have the following topolgy:
asterisk box set with public ip address 1.2.3.4
i have a snom200 sip phone that resides on a subnet with 192.168.0.10
address which i know have worked previously with vocal
now my sip.conf file looks as follows:
==
[general]
port = 5060 ;
Hi,
I am trying to setup an overhead paging system with asterisk. I have
followed some of the advice from the list and have oss.conf set for
autoanswer. The sound card and speakers work because they can play mp3s
just fine. When I call the extension, the asterisk console looks like
everything
Its not an apps place to take nat int account. WHERE did you get the idea
that it was?
bkw
- Original Message -
From: Brian Rathman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 18, 2004 5:03 PM
Subject: [Asterisk-Users] app_prepaid NAT issue
I was able to get
How do you register?
do this /msg NickServ help
or /msg NickServ register
[yourpassword]
You will be required to /msg NickServ IDENTIFY
[yourpassword] before
you can join #asterisk.
I'm sorry we had to do this but the spambots that
join and part 100+ times
per hour were getting way out
On Sat, 2004-06-19 at 06:13, Stephen R. Besch wrote:
So, if someone could brief me on the GPL issue, and (perhaps someone
else) offer a distribution point, it's free for the asking, VB sources
and all.
Stephen R. Besch
Alright, I've waited a long time before offering this. Anyway, the
Hi Glen, I have had the same problem for quite awhile, since around
February, all cvs codes that I have tried, and with h323, I have been
getting no audio. I am forced to stay with mid-Jan version of the cvs
because of this. I tried using ulaw, g729, but same results, I have in a few
occasions
Citat Klaus-Peter Junghanns [EMAIL PROTECTED]:
Am Fr, 2004-06-18 um 19.56 schrieb Lee Howard:
Firstly, I'm not just talking about receiving faxes.
If my choices are between HylaFAX and spandsp and if I want outbound
queueing and a client-server interface for networked usage, then
On Fri, Jun 18, 2004 at 09:27:52PM +0100, Steve Hanselman wrote:
LG GDK-186 PBX --PRI--- TE405P/Asterisk ---PRI--- Telewest (Telco
provider)
--snip---
Any ideas on where to start?
asterisk is get timing from your telco, and provide timing for you gdk
pbx.
Wayne [EMAIL PROTECTED] wrote:
Just wondering how people test your emergency dialing in the UK.
Obviously you need to dial the 999 for emergency services, but am a bit
unsure if this would go down too well with the operator with a 'sorry
just testing' call. (you do all /test/ your emergency
Let's try again, missed a line in the last reply...
On Fri, Jun 18, 2004 at 09:55:24PM -0400, Yifang Dai wrote:
On Fri, Jun 18, 2004 at 09:27:52PM +0100, Steve Hanselman wrote:
LG GDK-186 PBX --PRI--- TE405P/Asterisk ---PRI--- Telewest (Telco
provider)
--snip---
Any
Guys,
just for an informational update: I re-worked the chan-sccp so that it
compiles cleanly under CVS-HEAD of Asterisk.
Also await the soon-to-be Cisco 7970 7935 Support after i received the
hardware.
What might be taking a bit longer is the support of the Cisco Extension
Box for the 7960,
[inside]
exten = 2000,1,Dial(foo)
exten = 2001,1,Dial(bar)
...
[inside-sip]
exten = _.,1,SetVar(areacode=206)
exten = _.,2,Goto(inside,${EXTEN},1)
doh thanks
randy
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Lee Howard wrote:
Furthermore, even if you assumed that spandsp was as stable as
HylaFAX, there is a vast feature-set difference between them as far as
the faxing itself goes. Steve has already made it clear that he sees
no future in fax, and that he does not intend to bridge that
feature-set
On 2004.06.18 20:02 Steve Underwood wrote:
Lee Howard wrote:
So, show me a T.38 channel driver for Asterisk. And if you think
that using t38modem is ugly, then show me a T.38 driver for HylaFAX.
I really want to see spandsp talking through T.38 across the
internet. T.38 and fax to/from e-mail
Updated to the latest code release of *
today. After compiling and reinstalling the SIP dialout connections
through our media gateway stopped working. Finally tracked
down the issue. In chan_sip.c in transmit_invite there was ;rport added to
the INVITE via line of the msg:
snprintf(p-via,
Modified Files:chan_sip.c Log Message:Enable support for
RFC3581 (bug #1862)
bkw
- Original Message -
From:
Todd Graham
To: [EMAIL PROTECTED]
Sent: Friday, June 18, 2004 11:05
PM
Subject: [Asterisk-Users] current code
release chan_sip problem/question rport
i am using the freebsd port, which seems to not yet have WaitExten(),
which i kinda want to use thusly
[ext-666]
exten = _.,1,SetVar(areacode=666)
exten = _.,2,Background(zz-in-who) ; give them list of extns
exten = _.,3,WaitExten(10) ; let them enter extn to call
Thanks Brian. Makes sense now. I guess I need to forward this extension
info on to our media gateway vendor to get them to include in their SIP
stack.
Todd
- Original Message -
From: Brian K. West
To: [EMAIL PROTECTED]
Sent: Friday, June 18, 2004 9:09 PM
Subject: Re:
hello asterisk list,
i've installed asterisk-0.9.0 on fedora core 1, i've been receiving error 401 when i
connect my voip gateways on asterisk (welltech fxo, antek fxs)
here is my sip.conf
[general]
port=5060
bindaddr=0.0.0.0
context=from-sip
I've install the asterisk in my Redhat 9 and it work properly with the SIP
phone. Then i install the festival as mention in
http://www.voip-info.org/wiki-Asterisk+festival+installation. the problem is
when i dial the extension 555 in the asterisk console it show like :
-- Executing
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