.
just be aware that g711 will use 80Kb up and down...
gsm and g729 wil use 30/40Kb
then :
disallow all
allow = gsm
allow = g729
Olivier
kurt x a écrit :
I have an Asterisk server that I use at work. I have a phone that is
at home that logs into
the Asterisk server at work. My home phone
I have an Asterisk server that I use at work. I have a phone that is
at home that logs into
the Asterisk server at work. My home phone is hooked up via DSL
through a Linksys router. You can see the my sip.conf for the phone
blow.
The problem is each time the phone rings I can hear/be heard 50%
A 488 can mean a codec miss match. Check that your Asterisk box is
configured for g729.
Kurt
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debug ccsip message
Kurt
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I have four DIDs. 2400,2401,2402, and 2403
There is no phone attached to 2400 but the other three DIDs do have
phones attached
All the four DIDs have their own voicemail and voicemail works on all
the DIDs. When you dial 2400 it rings the other three numbers. If no
one picks up, it goes to the
Does anyone know if Asterisk supports any Voice recognition software
or is there a third
party out that has one available for Asterisk.
What I want to do with Voice recognition.
When some calls my * IVR instead of the caller spelling the name via
the buttons I want the user to be able to say the
Is it possible to have a bunch of people call a meetme room then have
that room call
into another conference off net. T
Kurt
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I am trying to get a IP 4000 to register to Asterisk. I can make
outbound calls from the IP 4000 but not to it. When I implement "sip
show peers" it lists the extension but with no IP address
(unspecified). I am configuring the phone via the web interface. I
am not using ftp or tftp to configur
Matching the Correct Inbound POTS Dial Peer for DID
For DID to work correctly, make sure the incoming call matches the
correct POTS dial-peer where the command direct-inward-dial is
configured.
If your PRI has DIDs you need the command.
Kurt
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I configured queues.conf and just added a bunch of "member =>
SIP/" numbers to
the bottom. I set up my extensions.conf with the access number to the
queue. Everything works but the phones on the lists display a ANI of
"911" out of area. Is there away to change that ANI to something
else.
Tha
I need to able to ring 30 phones at once on * plus another 10 that are
not on Asterisk.
I know I can use the
Dial(SIP/1&SIP/2…SIP/30&SIP/[EMAIL PROTECTED]&SIP/[EMAIL PROTECTED]/109) but
this
seems cumbersome. Is there an easier way to do achieve this?
Kurt
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Is it possible in * to set the Packetization period. For example: If
I want G711 to be at
10ms. Is that possible in *?
Thanks,
Kurt
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Upon setting up and configuring the my extension.conf, meetme.conf and
following the instruction outlined at this web page:
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
I get the following errors when calling the meetme number.
Executing Wait("SIP/216.53.118.2-f41196e0", "1") in new stack
I think I configured the MeetMe right. Since I am using SIP for
inbound calls I followed the
instruction, for 2.6 kernel, from this web page:
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
When I call the MeetMe number I get the greeting to enter in your
conference room. I do and get inv
I would like to know if any body is using the Polycom Soundstation IP
4000 SIP conference phone with Asterisk. I am thinking of purchasing
one.
Kurt
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A user has their unavailable message played and once that message
is over the Comedian
message is played right after. Is there any way to prevent the
Comedian message being
played if the user's unavailable/busy message is being played.
Thanks,
Kurt
My problems is when I log into the web page to get my voicemail I see that there
nothing being listed. I know there is vmail their because I can
retrieve the messages from the
phone.
I changed the following line in vmail.cgi so I do not need to login
with my extension plus context.
$context="loc
I have the below script that works but for one problem. The call
cannot last longer then 4 minutes when the script is utilized.
However, when I configure my extension.conf to not call the script the
call will stay up until I hang-up.
I call the script as follows:
exten => _24XX,1,AGI(internal.a
I noticed when I call certain IVR systems, such as 1800calldhl, that
Asterisk will not
barge the prompt. Would this imply that Asterisk has an Early media
detection problem.
Is anyone else experiencing this problem. Is there a fix?
Kurt
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Does Asterisk support preference for the dial peers.
For example:
I have two outbound peers in *. The first is a SIP dial peer and the
second peer is to
the PSTN via a T1.
The SIP dial peer is the main outbound peer for all calls. However, if
the my SIP providers network goes down, I need to
e to configure your dialplan to hunt to the next extensions?
> How else would * know to try 94207 if 4207 is busy?
> - Original Message -
> From: "kurt x" <[EMAIL PROTECTED]>
> To: "Asterisk"
> Sent: Thursday, April 21, 2005 3:08 PM
> Subject: [Ast
I have aSIPURA 841 that is working on L1 with phone # 4027. L2 is
configured for 94027.
Both numbers register with Asterisk. When issuing the command "sip show peers"
both numbers have the same IP address but 94027 show its sip port at
5061. Which I expect is right. When I dial 4027 it works bu
I have the following extension (7700) that can dial out with the below config.
exten => _1nxxnxx/7700,1,Dial(SIP/[EMAIL PROTECTED])
exten => _1nxxnxx/7700,2,Hangup
If I change it to
exten => _1nxxnxx/77XX,1,Dial(SIP/[EMAIL PROTECTED])
exten => _1nxxnxx/77XX,2,Hangup
It does no
I have the Digium S100i IAXy device hooked up to my asterisk server.
When I pick
up the phone I do get dial tone but it does not stop when I start to
dial a number. The
dial tone is alway heard and it does not make the call.
It does register with Asterisk
I can make a call to the IAXy device a
I,ve gotten the GotoIf statement working now. I hard coded the value
10 in place of the ${DIGITS} varible. Worked like a charm.
Thanks to everyone who helped.
Kurt
On Wed, 09 Mar 2005 12:07:51 -0600, Chris Wade <[EMAIL PROTECTED]> wrote:
> kurt x wrote:
> > [globals]
>
AIL PROTECTED]> wrote:
> kurt x wrote:
> > [globals]
> > Setvar(DIGITS=10)
>
> Try this instead...
>
> [globals]
> DIGITS=10
>
> -Chris
>
>
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IGITS}]?4:5)
exten => s,4,Setvar(${CALLERIDNUM}="Unknown")
exten => s,5,Voicemail(u${ext})
exten => s,6,Hangup
Kurt
On Wed, 09 Mar 2005 07:34:24 +1100, Howard Lowndes <[EMAIL PROTECTED]> wrote:
> On Wed, 2005-03-09 at 05:29, kurt x wrote:
> > I am tryin
ECTED]> wrote:
> Can you post your dialplan for that extension. Also, NoOp works great for
> debugging these issues.
>
>
> On Tue, 2005-03-08 at 12:29, kurt x wrote:
>
> I am trying to test how the GotoIf and $LEN functions work but am not
> succeeding is this ven
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is
this venture. When I dial and access voicemail with an ani of 3000
the gotoif statement does not push the call to s|6. Its goes through
each line( 1,2,3,4,5,6,7) . In additon when I dial with a 10 digit
ani the
I would like to know if the following lines represent the RTP traffic
going across,
the CODEC being used is G711ulaw, or both. The complete trap is below
the dotted lines
Thanks
Kurt
asterick*CLI>
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 004 Type: VOICE Subclass: 4
Timestamp: 02570ms S
Can someone explain in greater detail the following two Control
frames. The IAX2
draft document had extremely brief explanations.
LAGRQ = Lag request
POKE = Poke request.
Thanks,
Kurt
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htt
I have two * boxes running two differnet versions of *.
Box A is running:
Asterisk CVS-HEAD-07/14/04-16:28:29 built by
[EMAIL PROTECTED] on a i686 running Linux
Box B is running:
Asterisk 1.0.3 built by [EMAIL PROTECTED] on a i386 running FreeBSD
I can make a IAX call from B to A but not from
When I receive voicemail notification via e-mail I noticed that the
${VM_CALLERID) puts the IP address of the * box when callee info is
not present. Is there a way to have the field put Unkown caller in
instead of the IP address of the * box.
Kurt
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When I access the Directory() and use it to call an extension, the
origination hears garbled or inconsistent ringing. The termination
side rings normally and the conversation is clean in both directions.
Kurt
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The Directory command is working properly but the ringing herd in
the origination phone is either garbled or herd infrequently. The
termination phone does ring with consistency. Any suggestion on what
might be happening.
Kurt
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:
Unable to open silence/10 (format ULAW): No such file or directory
Kurt
On Mon, 24 Jan 2005 12:53:11 -0500, Roger Gulbranson
<[EMAIL PROTECTED]> wrote:
> On Mon, 2005-01-24 at 12:36 -0500, kurt x wrote:
> > . Once the .gsm file is finished playing you can not select any of the
>
I set up an IVR systems that plays a message for 15 seconds but
once the message is over you can not select any of the prompts.
If you select something within 10 seconds the IVR system works.
I even set the "ResponseTimeout" cmd to 25 secs but that does
not work.
Jan 24 09:54:29 NOTICE[-12226448
Is there any way to encrypt the PIN numbers in voicemail.conf.
I looked at the Wiki page for voicemail.conf but it did not mention
anything about that topic.
I am not using MySQL or any other thrid party database.
Kurt
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L PROTECTED]> wrote:
> If you put the following in your Dialplan, pressing * should break you
> out of voicemail and call VoiceMailMain
>
> exten => a,1,VoicemailMain,EXTEN
> exten => a,2,Hangup
>
>
> On Wed, 19 Jan 2005 11:33:23 -0500, kurt x <[EMAIL PROTECTED]&
Other then the standard "sip debug" is there any other
sip debug bugs like for errors, events, etc.
Kurt
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I want to know if there is way to break out of the voicemail message.
for example:
On my Noterl PBX when you dial you number from any where
you get your recorded voice mail message, but during the message I
press 81 and break out of that message. It then
prompts me for my PIN thus allowing me to
That was the ticket. The Extra ")" was the problem.
Thanks Sean.
Kurt
On Tue, 18 Jan 2005 08:13:31 -0800, Sean Kennedy <[EMAIL PROTECTED]> wrote:
> kurt x wrote:
>
> >What I am trying to do is the following: A call is sent to the * box
> >via a SIP invite.
What I am trying to do is the following: A call is sent to the * box
via a SIP invite. The * box answers via an IVR menu system with "
Enter the extension you want to dial" so I enter in my 5 digit
extension and get the below message.
Jan 18 10:10:03 WARNING[-1380238416]: channel.c:1860 ast_requ
I am trying to use the Directory() but am having difficulty using it.
According to Wiki page that I found you need to pass it
your voicemail.conf context. My vm-context is [local]. So when
I setup the cmd (Directory(local)) I can search on the three letters
of the last name find that user. But
Question: What is your reasoning for using Cisco Voice Mail instead
of Asterisk's voice mail.
IMHO it would make more sense to keep everything on Asterisk.
Kurt
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You need to either download 12.3(11)T or 12.3(10)LD.
Kurt
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See if you have the below configure under your "dial peers" or "voice
service voip".
If you do, then issue this command " no signaling forward unconditional"
signaling forward unconditional
Kurt
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[EMAI
I am interested to know how one would calculate the amount of PSTN
connection needed for backup on an Asterisk PBX that is being setup to
receive its DIDs via a VoIP provide. To sum up what I am
implementing: I am porting my DIDs to a VoIP provide so I will need a
back up plan in place if the Dat
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