thanks I 'll try ... :-)
jl
2005/7/4, Jean-Louis curty [EMAIL PROTECTED]:
Hi,
I'd like to understand what should i do to use G729 codec in a legal way,
how do I order licences ? to whom ? how do I install them on asterisk etc ?
thanks in advance ,
jl
Ok,
Now you get registration, but still you cannot complete the call.
I don't see your dialplan on your files. However, the error log shows that
ztdummy is not working, so there is no timming.
Lsmod shows the module already installed?
Carlos
-Original Message-
From: [EMAIL PROTECTED]
Yes i we know it but as it is an intermediate patched version i think it's
better way for the final release
A++
-Message d'origine-
De : Javier Ergas [mailto:[EMAIL PROTECTED]
Envoyé : vendredi 8 juillet 2005 18:51
À : [EMAIL PROTECTED]
Objet : RE: [Asterisk-Users] Sipura SPA-841
G279 on Asterisk works great.
Jean-Louis curty wrote:
thanks I 'll try ... :-)
jl
2005/7/4, Jean-Louis curty [EMAIL PROTECTED]:
Hi,
I'd like to understand what should i do to use G729 codec in a legal way,
how do I order licences ? to whom ? how do I install them on asterisk etc ?
Greetings-
As most of you who monitor this list know, I've been messing about
with Asterisk -HEAD, Cisco Callmanager, and the nufone H323 channel
driver here for some time- with pretty decent success. I'm hoping to
cash in a chip here- I've run into something that is probably a very
simple
Rich Adamson wrote:
The iax2 show peers is indicating the phone is not registering
properly. (If the phone never changes IP addresses ever, then he
could put a host= statement in the iax.conf, but that is very
non-standard and will only serve to confuse people.)
In my reply to time bandits
Hi,
Does anybody knows if ADSI could be used from the SIP channel?
Alejandro
___
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
wassim darwish
Sent: Friday, July 08, 2005 1:15 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] changing Nobody picked up in 3 m
i dont know how to edit the the time for
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael L Smith
Sent: Thursday, July 07, 2005 12:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] URGENT: hardware spesifications needed
Who are
Well the telco may be handeling it :P the telco gives a fast busy
(reorder tone) for numbers that aren't configured.. I'd rather give a
message :) The solution offered above fixed the problem.
On 7/8/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Friday 08 July 2005 09:37, Matt wrote:
What
On Thu, 2005-07-07 at 07:36 -0500, Guillermo Salas M wrote:
Hi,
I've oh323 chan installed and working to make calls from SIP to H323
devices. The problem is can no hear sound with the H323 device. I think
this is some related with codecs o nat, because the H323 have one public
IP from a
Ha!
I think I found my phone - see point 2 on my previous post (copied below).
Although the sticker on the bottom of my phone does have ALDU, the
firmware version is 02.09.07
So, it should cope with IAX only (and no SIP at all), yet I cannot get
it registered (yet)
Carlos Alperin wrote:
Ok,
Now you get registration, but still you cannot complete the call.
I don't see your dialplan on your files. However, the error log shows that
ztdummy is not working, so there is no timming.
Lsmod shows the module already installed?
Carlos
I did have it on a
Ha! (again)
Hopefully this helps you guys a bit (to help me )
Ta yet again,
Zoltan
From the wiki page about my phone, I changed my iax.conf to:
(note 6 lines after CallerID)
* sip.conf **
[general]
port=4569
bindaddr=0.0.0.0
bandwidth=medium
disallow=LPC10
If I write try with sip, everybody will try to hang me in the highest tree.
So, If was on your position, I'll try SIP.
I'm trying to help, no to argue with anyone, and of course not to dissinform
or missinterpretate anything. Just help.
Regards,
Carlos
-Original Message-
From:
The timing is provided by Zaptel, if you don't have that there is no timing.
So, after we get the registration, the next step is get the TDMoE working,
if not you cannot generate any calls.
If I confuse you, I'm sorry but was trying to help not to make you loose.
I don't use ztdummy but you
In extension.conf
exten = XXX,1,Dial(SIP/XXX|30)
Change the 30 to 40 and the phone will ring for 4ms. The |30 is how
long to ring the interface. I'm using SIP here. This is one way to
change that amount but I don't know what your configuration looks like.
-
Chris Stinson
If putting a host= statement in * does anything in terms of completing
the registration, there is a serious infrastructure issue. That isn't
going to lead him to resolving the registration problem.
Why not try 'iax2 debug' and let's take a look at what's going on to
resolve the registration
My question: How do
I configure AAH via AMP to make a connection through our legacy PBX to the
PSTN?
Details:
We're trying out
Asterisk through Asterisk @ Home.
Our legacyPBX
has a modem type dial tone port that we hooked a Digium FXO
to.
Now I can
dialfrom the XTEN client on my
Hello.
I've been trying to access www.soft-swtch.org for a couple days, but it
seems to be down.
Anyone got the same?
I need to talk to Steve Underwood, could someone give me his email/phone.
Please, in private of course.
Thanks.
Denis.
___
The iax2 show peers is indicating the phone is not registering
properly. (If the phone never changes IP addresses ever, then he
could put a host= statement in the iax.conf, but that is very
non-standard and will only serve to confuse people.)
In my reply to time bandits suggestion,
Carlos Alperin wrote:
If I write try with sip, everybody will try to hang me in the highest tree.
:-)
(you have a nice attitude)
So, If was on your position, I'll try SIP.
I did actually have a quick try - and got similar problems - but I'll
have to have a good look to know if the
I'm having sound quality problems on the remote side with anything
higher than 3.1.2(d).
3.1.3(a) oscillates and is just too quiet.
the pre-release of 3.1.4(a) is staticy according to multiple folks
that I called.
___
Asterisk-Users mailing list
Thanks for the thorough reply.
I'm aware that there necessarily are inconsistencies between termination
providers; I was just curious to find out if there's some form of
standard one should follow, which may either result in more consistent
behavior, or at least shift culpability to the
Ed,
Please let me know how you make out. I am sort of keeping tract of what
asterisk needs for speech.
I am not working a project yet, just trying to get a feel for what
people need before I start making new stuff.
Race the tyrant Vanderdecken
Race at code tyrant dot com
-Original
If you need to modify the C source code for voicemail to add 0 or other
numbers I can do that for you.
Race the tyrant Vanderdecken
Race at code tyrant dot com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos
Alperin
Sent: Friday, July 08, 2005
No one is going to hang you, Carlos. Trying sip probably isn't a bad
idea at all, but that effort still wouldn't resolve the iax issue. ;)
If I write try with sip, everybody will try to hang me in the highest tree.
So, If was on your position, I'll try SIP.
I'm
Prospective user question
What is the simplest/inexpensive board to use in order to be able to
receive faxes in Asterisk.
I have a couple of cards I bought off E-bay think they were TX-1000 (or
supposed to be anyways)
but I assume I need some form of fax card though.
Thanks
Well, the stable (1.0.7) version supports it as well. Just make sure
you have an o (the letter , not the number zero) defined as an
extension in the last macro/context that called the voicemail app.
If all you want to do is let them leave a message and after they end
recording (by pressing #) they
Carlos Alperin wrote:
The timing is provided by Zaptel, if you don't have that there is no timing.
So, after we get the registration, the next step is get the TDMoE working,
if not you cannot generate any calls.
If I confuse you, I'm sorry but was trying to help not to make you loose.
I
If I write try with sip, everybody will try to hang me in the highest tree.
:-)
(you have a nice attitude)
So, If was on your position, I'll try SIP.
I did actually have a quick try - and got similar problems - but I'll
have to have a good look to know if the probs are
Eric Wieling aka ManxPower wrote:
Andrew Sayman wrote:
Okay, so I believe I've gotten rid of the IRQ conflict, and it's still
not working.
All analog FXO ports are considered answered when the dialing is
finished.
Thanks for this bit of information.
I went back with a Vonage modem (I'm
Ha! (again)
Hopefully this helps you guys a bit (to help me )
Ta yet again,
Zoltan
From the wiki page about my phone, I changed my iax.conf to:
(note 6 lines after CallerID)
* sip.conf **
[general]
port=4569
bindaddr=0.0.0.0
bandwidth=medium
I agree 100%, and as you can see now they complete the registration.
The problem that was hidden is that there is no timing working on the
system. So, I don't think that dialing or anything else is going to work.
Now, what happen with the network, I don't know because it looks like
everything is
Zoltan,
Rich is right, try to use Ethereal to see what the phones are sending out.
You said that you can see the phones, so watch their traffic, disregards the
http (just they're web servers) and see what happening on the port that
belongs to SIP IAX. You never knows what you 're going to find.
Hi Rich,
See debug output after your post below.
Thanks,
Zoltan
Rich Adamson wrote:
If putting a host= statement in * does anything in terms of completing
the registration, there is a serious infrastructure issue. That isn't
going to lead him to resolving the registration problem.
Why not
Thanks,
I'm going to check first, and I'll let you know.
Regards,
Carlos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Friday, July 08, 2005 2:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
Thanks for the thorough reply.
I'm aware that there necessarily are inconsistencies between termination
providers; I was just curious to find out if there's some form of
standard one should follow, which may either result in more consistent
behavior, or at least shift culpability to the
hello,
i'm currently using asterisk (cvs-head) as PSTN gateway. the routing
logic is mostly done in OpenSER. the problem is that i'm not able to
transfer calls between the PSTN and another SIP peer (when the PSTNSIP
connection goes over asterisk but the SIPSIP connection does not).
there are 2
Cablevision's DNS says it doesn't know of this domain name.
[EMAIL PROTECTED] wrote:
Hello.
I've been trying to access www.soft-swtch.org for a couple days, but it
seems to be down.
Anyone got the same?
I need to talk to Steve Underwood, could someone give me his email/phone.
Please, in
Thanks,
Now I know that I don't need to change my name address. (Joke)
My only concern was to be sure that every device is seen each other.
I only use IAX2 for trunks between servers, which pretty good, however I
hear every kind of argues about why I shouldn't use.
But for me, it never
Bill Wesson wrote:
My question: How do I configure AAH via AMP to make a connection
through our legacy PBX to the PSTN?
Details:
We're trying out Asterisk through Asterisk @ Home.
Our legacy PBX has a modem type dial tone port that we hooked a Digium
FXO to.
Now I can dial from the XTEN
Rich Adamson wrote:
I still think it's a ztdummy issue - is that also needed with SIP?
Zaptel timing (eg, ztdummy) isn't needed for iax or sip. Its only
needed on a couple of application items like the meetme, etc, which
are documented on the wiki.
You're quite right about SIP not
Rich Adamson wrote:
Well, if I were working with those I'd fire up Ethereal and look to
see exactly what the phone was doing. If they really are iax capable,
you should see at least some iax packets coming from them. Then, the
real answers to why the phone doesn't work can be resolved from the
Cablevision's DNS says it doesn't know of this domain name.
Sorry... I mistake was done... The correct address is :
http://www.soft-switch.org
Thanks.
Denis.
[EMAIL PROTECTED] wrote:
Hello.
I've been trying to access www.soft-swtch.org for a couple days, but it
seems to be down.
Rich Adamson wrote:
Thanks for the thorough reply.
I'm aware that there necessarily are inconsistencies between termination
providers; I was just curious to find out if there's some form of
standard one should follow, which may either result in more consistent
behavior, or at least shift
Just got a reply from sipura support confirming the problem and
recommending to use this firmware:
http://www.sipura.com/download/temp/phone/spa841-03-01-03-a-vol-fix.zip
while they're fixing it and until they release the version 3.1.4
thumbs up for their fast reply.
Hugh L. Johnson wrote:
Alex:
Thanks this solved the problem.
Syed Akbar
Alico Systems Inc
www.alicosystems.com
Tel: 562-436-1510
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Friday, July 08, 2005 6:06 AM
To: Asterisk Users Mailing List -
Rich Adamson wrote:
Yes, the above indicates the phones did in fact register at least
one time, as indicated by the IP address in the Host colume.
The Status = Unreachable is saying the phones no longer are
reachable by asterisk. So, either the phone is going to sleep
on its own, or, for
[EMAIL PROTECTED] wrote:
I've been trying to access www.soft-swtch.org for a couple days, but it
seems to be down.
Anyone got the same?
Indeed, same issue here.. If we can help, let us know. (dns/website hosting)
Cheers,
Kristof.
___
Asterisk-Users
Bill Wesson wrote:
My question: How do I configure AAH via AMP to make a connection
through our legacy PBX to the PSTN?
Details:
We're trying out Asterisk through Asterisk @ Home.
Our legacy PBX has a modem type dial tone port that we hooked a Digium
FXO to.
Now I can dial from
-Original Message-
My question: How do I configure AAH via AMP to make a connection through our
legacy PBX to the PSTN?
snip
--I cannot help with the AMP way but what you need to do is something like
this in extensions.conf
Exten = _9NXX,1,Dial(zap/X/w9w${EXTEN:1}
That
On Fri, 2005-07-08 at 15:19 -0400, Shawn Porter wrote:
Prospective user question”
What is the simplest/inexpensive board to use in order to be able to
receive faxes in Asterisk.
I have a couple of cards I bought off E-bay think they were TX-1000
(or supposed to be anyways)
but I assume
Hi all.
I am seeing incoming calls from digital lines (mobiles e.g.) dialling
my main number + 3-digit extension just fine (Accepting voice call
from '11234567' to '250' on channel 0/1, span 1). The problem however
is with calls from analog lines:
Accepting voice call from '13331846' to '25' on
I've read that the auto fax detection for asterisk is built into the
chan_zap software, however i've been experiencing odd behavior. I have
two digium cards, a TDM and a TE110p, and the only time i am getting the
fax detection is on outgoing calls from the TDM card.
zapata.conf has =both.
I am thinking of having a pots line with multiple numbers on it, and having
Asterisk dial my desk if the outside caller dials xxx-xxx-xxx1, and ring
another desk if the person called xxx-xxx-xxx2, etc.
Can Asterisk do this?
--
Jeff Ramsey
MIS Administrator
Tubafor Mill, Inc.
Carlos Alperin wrote:
Zoltan's problem looks more to be a timing issue, just he is not using
Zaptel but ztdummy modules that are not loading.
Regards,
Carlos
Is this how the modprobes are supposed to respond??
gl0:/home/zls # modprobe zaptel
gl0:/home/zls # lsmod | grep z
Module
Zoltan,
The debug stuff at the bottom of this email looks normal. Do the
'iax2 debug' again, but include the registration (reboot the phone),
copy/paste that, then let the debug run until something different
then those POKES show up. To save a little time, dial from one
iax phone to another and
Jeff Ramsey wrote:
I am thinking of having a pots line with multiple numbers on it, and having
Asterisk dial my desk if the outside caller dials xxx-xxx-xxx1, and ring
another desk if the person called xxx-xxx-xxx2, etc.
Can Asterisk do this?
Not really with POTS. There is some basic support
On Friday 08 July 2005 17:01, Jeff Ramsey wrote:
I am thinking of having a pots line with multiple numbers on it, and having
Asterisk dial my desk if the outside caller dials xxx-xxx-xxx1, and ring
another desk if the person called xxx-xxx-xxx2, etc.
Can Asterisk do this?
Asterisk can detect
Rich Adamson wrote:
Yes, the above indicates the phones did in fact register at least
one time, as indicated by the IP address in the Host colume.
The Status = Unreachable is saying the phones no longer are
reachable by asterisk. So, either the phone is going to sleep
on its own, or, for
Well, if I were working with those I'd fire up Ethereal and look to
see exactly what the phone was doing. If they really are iax capable,
you should see at least some iax packets coming from them. Then, the
real answers to why the phone doesn't work can be resolved from the
packet trace.
Greetings,
We installed a number of SPA-841 in a office environment, originally
firmware was 0.9 something and the audio in the headsets worked really
well, with virtually no muting of received audio.
When I first upgraded a couple of the phones to 3.1.2 the headset audio
went to crap, muting
Hello everyone,
We have recently turned up a new T1 from McLeod (Midwestern CLEC). It
is configured like so:
/etc/zaptel.conf:
loadzone=us
defaultzone=us
span=1,0,0,esf,b8zs ;(also tried 1,1,0,esf,b8zs)
bchan=13-23
nethdlc=1-12
dchan=24
/etc/zapata.conf:
switchtype=national
Title: RE: [Asterisk-Users] Asterisk/Grandstream Budgetone disconnect issue
Upgraded tover 1.0.6.7. Problem is still happening.
Thanks.
-Original Message-
From: Storm D. J. Petersen
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: 7/7/05 10:01 PM
Subject:
I don't see any packet coming from 192.168.0.201 192.168.0.202.
But I see also packets from other networks plus spanning tree protocol, like
if you have a switch with STP...
Is your computer on the same network that the phones?
Carlos
-Original Message-
From: [EMAIL PROTECTED]
Yes,
But it looks like they cannot open a device on the Zaptel.conf.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei
Sent: Friday, July 08, 2005 5:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hello, I'm new on this.
I've recently installed Asterisk on VMWare and it
installed OK, I could enter to the main site, I've entered Sugar but when I want
to enter to Asterisk Management Portaland use the default admin and
password but it does not work. Does anyone had this problem?
Thanks
**Snip**
pbx*CLI pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, Active
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T313 Timer: 4000
N200
Craig wrote:
Greetings,
We installed a number of SPA-841 in a office environment, originally
firmware was 0.9 something and the audio in the headsets worked really
well, with virtually no muting of received audio.
When I first upgraded a couple of the phones to 3.1.2 the headset audio
went to
I've found the same problem. I have to log in using
'maint' and 'password'.
--Bill
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fabrizzio
ValenciaSent: Friday, July 08, 2005 2:37 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users]
On Jul 6, 2005, at 9:07, Kevin P. Fleming wrote:
Ian White wrote:
The use of the nonce looks right to me. Can somebody point out
what is going wrong here?
Yes, I agree, it looks correct. However, what version of Asterisk
are you testing against? Current CVS HEAD adds 'stale=true' to
thanks a lot
now i can play with this fantastictoy
:-D
- Original Message -
From:
Bill Wesson
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Friday, July 08, 2005 6:59 PM
Subject: RE: [Asterisk-Users] admin
password does not work on APM in
Ian White wrote:
It is running the latest CVS tag v1-0 checkout. This is a production
system, and as such I'd prefer not to run CVS HEAD. If you (or one of
the other developers) have a CVS HEAD server running and want to give
me a SIP username/secret/host I'll reconfigure the phone in
Javier Chia ha scritto:
I did that, the phone logged in, but is unable to make nor recive calls.
did you disabled the skinny channel on modules.conf?
noload = chan_skinny.so
load = chan_sccp.so
if this does not work for you, you should post your sccp.conf. It should
be a config problem
When using a PRI on my asterisk system, and when calling a busy number
I often get an All Circuits Are Busy message. I've trouble shot it
and made absolutely sure all circuits are not busy... but the number
being called is in fact the issue. How do I get Asterisk to give me
a busy signal
I have installed asterisk in a 4.11 RELEASE FreeBSD, and we are using
two Zoom X5v with SIP and works fine, we can call each other and this
is OK
--extensions.conf--
[general]
static=yes
writeprotect=no
[sip]
exten = 123,1,Dial(SIP/123,20)
exten = 123,2,Voicemail(u${EXTEN})
I have an issue with silent calls when an agent gets a call from the queue
What happens is
- The system dials a call (agent call)
- The caller picks up
- Asterisk sees the person picked up
- Transfered to an agent
- Agents phone automaticly picks up (sjphone auto accept on)
-The
We are using it too, withouta problem.
SipGetHeader and realtime works like charm.
I just didn't get spandsp working... It compiled ok, but doesn't work.
Denis.
On 06 de jul de 2005, at 13:56, Kevin P. Fleming wrote:
Tony Mountifield wrote:
Anyone here in the know about when HEAD will
Remember clients send packets from a random high port number which
changes. Port forwarding on your router is pretty useless. nat=yes
combined with qualify=yes should cause enough traffic on the right
ports to keep the NAT translations open on your NAT router.
Brian McCrary wrote:
In
When I put
load = chan_sccp.so
Then Asterisk is stopped after reboot.
My sccp.conf is:
[general]
keepalive = 5
context = internal
dateFormat = D-M-Y
bindaddr = 192.168.1.83
port = 2000
[SEP0008E399E223]
description = ciscosystems
type = 7910
context = internal
tzoffset = 0
autologin = cisco
On Fri, Jul 08, 2005 at 11:12:37PM +0200, Zoltan Szecsei wrote:
Is this how the modprobes are supposed to respond??
gl0:/home/zls # modprobe zaptel
gl0:/home/zls # lsmod | grep z
Module Size Used by
zaptel239620 0
crc_ccitt 6144 1 zaptel
Chris Shaw wrote:
**Snip**
pbx*CLI pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, Active
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
About once a day I have noticed a phantom incoming call with a caller ID of
[EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the
call is disconnected. Any clues?
David Koski
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
On Jul 8, 2005, at 12:43 AM, Jay Milk wrote:
All,
I'm currently only setting CID as a ten-digit number. Has anyone on
this list tested caller-id delivery with various services? Is there
*one* usable format (i.e. 1+10, or +1+10), or does it vary from
provider to provider?
Jay, FWIW the US
how to edit the time of ring 3ms to 4ms in
astcc since it displays this on console Nobody picked
up in 3 ms when nobody picked up the phone in
3ms and then it hangup.
please help i have been asking this question from long
time and no body answered me yet.
wassim darwish wrote:
how to edit the time of ring 3ms to 4ms in
astcc since it displays this on console Nobody picked
up in 3 ms when nobody picked up the phone in
3ms and then it hangup.
please help i have been asking this question from long
time and no body answered me yet.
Besides edit sccp.conf and modules.conf, is there
anything else that I have to change?
The phone is now logged in but can´t place nor receive
calls. It keeps giving Busy tone when I try to dial a
number.
But in: Asterisk*CLI shows the following:
SEP0008E399E223: Cisco Digit: 0001 (1) on
[EMAIL PROTECTED] wrote:
http://www.serviceview.bt.com/list/current/docs/Call_Charges.boo/00165.htm
Of course these are BT retail rates but I fully expect wholesale
rates based on call prefix will be available for carriers / ITSP
In some countries there's a company (companies?) providing
I try to get H323 to run, but have so far only partial success:
There is a Gatekeeper GK, where asterisk connects to.
The Gatekeeper sees Asterisk, and Asterisk sees the gatekeeper.
From the Network on the GK, asterisk is reachable via the number
07033. I have an extension on asterisk
Hi.
I have the following line in the default context of all my internal extensions:
exten = 9876,1,Transfer(125)
When I dial extension 9876 from any sip phone, * dutifully transferrs it to extension 125, which is just what I want.
Unfortunately when I dial 9786 from my Zap connected analogue
Hi,
I found the reason. Asterisk did not recognize DTMF-event because my
SIP phone sent DTMF-event with wrong rtp payload type.
In short, Asterisk is not guilty.
When the SIP phone calls, it will advertise RTP payload type 96 for
DTMF-Event; Asterisk answers with 96 and expects 96.
All,
I use various IAX providers to terminate my outbound calls. I set the
caller-ID to one of several DIDs, based on the called number. There
doesn't seem to be any rhyme or reason as to what the called user sees,
however.
Calls to most cell-phones show *exactly* the number I submit.
Calls to
On Thu, 2005-07-07 at 12:01 -0500, Matthew Boehm wrote:
Sahil Gupta wrote:
Hi,
I spent quite a few days with this and in the end I find that the 1.07
release is by far the most stable.
I had a lot of trouble with the CVS release.
Ofcourse, thats just in my case, what do the
Hello
Go back to the firmware before and all will be ok
or wait until the next one
Best Regards
Thierry
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
de Vahan Yerkanian
Envoyé : vendredi 8 juillet 2005 07:23
À : Asterisk Users Mailing List -
First of all sorry for the little offtopic post :-)
For one of our customers i need to make a vpn between the Netherlands
and Hungary.
Over this vpn two * machines are gonna talk IAX and employees in Hungary
are gonna use the Exchange server located in the Netherlands.
So far no problem...
The
I've been asked to integrate some simple speech recognition with an IVR.
Is there anything that people are using with Asterisk for this? Where
should I start reading?
/edg
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Lee Azzarello schrieb:
The README in the source code states:
app_conference doesn't have DTMF-activated features or anything like
that.
I'm curious how you got audio working on your compliation. I am running
CVS HEAD + app_conference in a Xen virtual machine. I can connect to the
channel but
i dont know how to edit the the time for ringing
3ms to 4ms,please help me.
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Why don't you just use Dial(SIP/125)??
Or better, if you have your extensions defined in context e.g.
[from-internal], just do:
exten = 9876,1,Goto(from-internal,125,1)
Julian.
On 7/8/05, Mark Edwards [EMAIL PROTECTED] wrote:
Hi.
I have the following line in the default context of all
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