ok im sorta confused.
when I save a * email'ed voicemail and I check the properties on th file It says
BitRate 13kbps
Channnels 1 mono
Audio Sample Rate 8 kHz
Audio Format GSM 6.10
when I look at the sox'd files from you script I see
BitRate 128kbps
Audio Sample Size 16bit
Channnels 1 mono
Almost... right now it lets you press * to cancel and enter a different
mailbox, but that just lets you leave a message rather than ringing the
extension. I guess exit vm and go back to the automated attendant is a
typical type of feature. Maybe it would be cool if there were a way to quit
Does anyone out there know if it is possible to discover the dialed
number when a line in an analog hunt group rings? I can't get a
straight answer from our IT folks. We have a 5ess switch delivering 4
analog lines which are in a simple hunt group servicing our lab. I
would like to have a
On Tue, 26 Aug 2003 17:48:55 -0500
Don Pobanz [EMAIL PROTECTED] wrote:
*This message was transferred with a trial version of
CommuniGate(tm) Pro*
On Tuesday, August 26, 2003 4:55 PM, Stephen R. Besch
[SMTP:[EMAIL PROTECTED] wrote:
Does anyone out there know if it is possible to discover
the
Again, not near my asterisk box so I can't check this out,
but is it possible to have the different ports drop into *
in a different context for each line? That way you could
just set up an 's' extension in that context for the
different attendants.
Yup. Set up different contexts in
What part of IN OTHER WORDS: Run Open H.323 v1.11.7, nothing newer, nothing older if
u want
this to work. don't you understand?
Jeremy McNamara
Jan Rychter wrote:
I have hit a problem where chan_h323 sometimes doesn't hang up properly
and stays stuck in the Up state, with asterisk consuming
Does chan_h323 support phone number calling via a gateway? ie.,
something like calling 5000 forwarded to:
exten = 5000,1,Dial(h323/[EMAIL PROTECTED])
if so - what format should the exten be in? Thanks.
Regards,
Steven Thomas
___
Continuing my problems with h323. I think I am getting closer.
SJPhone works direct to the gateway - calls and answers fine on the pstn.
So the gateway is working.
Inbound calls from PSTN = Gateway = Asterisk = Phone work great!
Outbound from Asterisk = Gateway = PSTN still remains a
Hello all !
How can I make conference authorization
based on pin number ?
I have:
exten = 1,1,Meetme,1234|ps|
where is a pin number
and this doesn't works
Where do I have to add information about pin number ??
Greetings
Andrzej Radke
___
Well, going by apps/app_meetme.c, for some of it, we see
inpin = strchr(inflags, '|');
if (inpin) {
*inpin = '\0';
inpin++;
/* XXX Need to do something
Title: Message
Does anyone have the
same issues and is there any work arounds.
I have a SNOM 200
which seems to work fine for so long but after an undetermined time when I make
a call I hear no audio. If I reboot the SNOM all is fine
again.
Also when I reboot
the SNOM it only ever picks
Stuart Hirst wrote:
Does anyone have the same issues and is there any work arounds.
I have a SNOM 200 which seems to work fine for so long but after an
undetermined time when I make a call I hear no audio. If I reboot the
SNOM all is fine again.
The same here. Version sip-1.16w. You have
Does anyone have the same issues and is there any work arounds.
I have a SNOM 200 which seems to work fine for so long but after an
undetermined time when I make a call I hear no audio. If I reboot the
SNOM all is fine again.
Also when I reboot the SNOM it only ever picks up the NTP
Hi,
I use asterisk-oh323 and a gatekeeper (gnugk) and
netmeeting
In asterisk i can have Caller ID when I do "show
channel " I have (N/A).
Does anyone know how I can have this caller ID
?
Notice that in the gatekeeper I can see the user
login of the netmeeting caller.
Regards
Rattana
Perhaps you should check out the AGI module. Write a perl script to compare
DTMF(pin) with any data storage(text file, Database). See this doc
http://home.cogeco.ca/~camstuff/.
The other solution is of course modify the source code to check for pin.
You can also use the Autheticate module.
I
Setup as follows: [private*] - Natting Router - [public*]
[private*] cannot register via IAX2 correctly while [public*] is running.
Status remains UNKNOWN even after minutes, calls from [public*] to
[private*] are not possible.
Console output of [public*]:
| *CLI iax2 show peers
|
Use extension logic.
Is there an echo in here?
Jeremy McNamara
[EMAIL PROTECTED] wrote:
Well, going by apps/app_meetme.c, for some of it, we see
inpin = strchr(inflags, '|');
if (inpin) {
*inpin = '\0';
Well, considering I replied as soon as I got it, it could always be a delay in
smtp or whatever...
Anyways, the documention advertises a feature which isn't present in the
module it indicates it is in. This would normally be classifed as a bug,
or do you feel it doesn't need to be raised?
-
I have on Chan_h323 with G729 and X100P trying to connect to
a Planet VOIP400 gateway box(http://www.planet.com.tw)
I uncommented g729 in the Makefile and I'm setting g729 in h323.conf
I'm receving in my side:
1:20.906 H225 Caller:810f070 h323ep.cxx(1537)
H323 Clearing
Is the order important in IAX.CONF?
I DID have this:
[iaxtel]
type=user
host=12.37.165.130
context=iax-caller
callerid=Guest IAX User
accountcode=iaxuser
;auth=rsa
;inkeys=iaxtel
[other]
host=dynamic
type=friend
context=default
auth=md5,rsa
secret=secret
inkeys=testinkey
outkeys=testoutkey
On Mon, 2003-08-25 at 18:05, Ernest W. Lessenger wrote:
At 11:09 AM 8/25/2003 -0500, you wrote:
Perhaps some day there will be a client side product/widget/whatever
for Asterisk, but right now it doesn't exist, to my knowledge that
is.
I believe the Asterisk Manager will do everything
hi,
how can I add or remove this line "include
=context"by the command CLI ?
regards
Rattana
Don Pobanz wrote:
On Tuesday, August 26, 2003 4:55 PM, Stephen R. Besch
[SMTP:[EMAIL PROTECTED] wrote:
Does anyone out there know if it is possible to discover the dialed
number when a line in an analog hunt group rings? I can't get a
straight answer from our IT folks. We have a 5ess switch
Hi
The endpoint seems to be running Radvision h323 stack, and I know
chan_h323 works with Radvision, there could be a couple of reasons!!
1) You dont have G729A in the capabilities of remote endpoint
2) The packetization interval is way off
The best way would be to run ethereal or dump323 and
Hello All,
Does anyone use a Plolycom SIP-based phone with Asterisk?
Does anyone use a Polycom SoundPoint 500 with Asterisk using SIP?
If so, please share your experiences, both good and bad.
Thanks,
Tim
___
Asterisk-Users mailing list
[EMAIL
On Wed, 2003-08-27 at 09:59, Timothy Soos wrote:
Hello All,
Does anyone use a Plolycom SIP-based phone with Asterisk?
Does anyone use a Polycom SoundPoint 500 with Asterisk using SIP?
If so, please share your experiences, both good and bad.
I tried to contact Polycom regarding their VoIP
I just received an unlocked ADSI phone and I am
playing with the ADSI script.
I was wondering how I can include Voicemail functions
(Check new messages, Delete message) into the soft
buttons.
I checked in app_voicemail.c and it looks like these
functions have already been programmed.
Is there a
Hi,
one question:
What you mean with unlocked ?
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von jerk face
Gesendet: Mittwoch, 27. August 2003 18:31
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] ADSI Programs
I just received an unlocked ADSI
Some ADSI phones come locked to a certain service provider. You cannot
load your own adsi scripts into these phones - you need one that isn't tied
to a specific company or pbx.
-d
At 06:35 PM 8/27/2003 +0200, you wrote:
Hi,
one question:
What you mean with unlocked ?
-Ursprungliche
check 'help'
include contexta in contextb
regards
Martin
On Wed, 27 Aug 2003, Rattana BIV wrote:
hi,
how can I add or remove this line include = context by the command CLI ?
regards
Rattana
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi List,
I am trying to locate some detailed documentation and sample configs. I
downloaded and compiled Asterisk, and I haven't been able to find much
detailed docs on the config files. The distribution I compiled and installed
doesn't have any config files, and the handbook is good but doesn't
I know all about most ADSI phones being locked.
The first line of my email was I just received an
unlocked ADSI phone and I am playing with the ADSI
script.
I have a Cybiolink P-I, and it is completely unlocked.
--- denon [EMAIL PROTECTED] wrote:
Some ADSI phones come locked to a certain
Gustavo Villaran wrote:
Hi, im new in the list and i want to buy a BRI card that works with
Asterisk PBX software for testing purpose, but i dont know which one
works with that software.
If someone knowns something that can help me, please write to me.
Thanks
Gustavo
My X100P card seems to have interrupt clashes with my Sound card, any ideas
to prevent this ?
Thanks and Regards
Ajit
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Anyone know offhand what the default flash time is? Where to find and
adjust if necessary? Going to test out some analog sets with * and wanted
to know.
Sincerely,
Andy Hester
Consero
(817)375-1244
(817)937-7977
___
Asterisk-Users mailing list
[EMAIL
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ajit M
Kallingal
Sent: Wednesday, August 27, 2003 1:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PCI X100P card interrupt problems
My X100P card seems to have interrupt clashes with my Sound
zaptel.h:#defineZT_DEFAULT_FLASHTIME750 /* 750 ms default
flash
time */
Martin
On Wed, 27 Aug 2003, Andy Hester wrote:
Anyone know offhand what the default flash time is? Where to find and
adjust if necessary? Going to test out some analog sets with * and wanted
to know.
hello,
Is this configuratoin possible:
--FXO
--FXO
ADTRAN
TA 750
- T1Card --- ASTERISK
-FXOT1
line
Yes it is possible.
Please describe what you want in the future. As you can see below your
mail looks like crap and wasted all your time drawing this mess out. You
really should look at the source to your last message and see how nasty
it was.
On Wed, 2003-08-27 at 14:38, Bartosz Jozwiak wrote:
Yes.
Jared Smith
On Wed, 2003-08-27 at 13:38, Bartosz Jozwiak wrote:
hello,
Is this configuratoin possible:
--FXO
--FXO
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