Thanks!
I got it now!
Here is a sample for a delayed callback after a caller gets to a users
voicemailbox. Purpose: Reminder for people that they got a message on their
v. box.
exten = 1002,1,Answer
exten = 1002,2,Set(CHANNEL(musicclass)=default)
exten =
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
From: Jared Smith [EMAIL PROTECTED]
There is a SIP timers patch in the bug tracker (see
http://bugs.digium.com/view.php?id=10665) that currently implements
this, and it's being
The rtptimeout feature has a few limitations:
. It is ineffective when the RTP is not terminated on Asterisk.
. It can cause false session hangups if the remote SIP UA does not support
silence suppression
. The companion rtpholdtimeout can cause false hangups if you make incorrect
judgment on
Phil Knighton schrieb:
Thanks for your reply Stefan :-)
I'm using Asterisk 1.4.10 now, I was using 1.2.16. My config hasn't
changed between the two, both had the hints set in extensions.conf with
entries such as exten = 510,hint,SIP/510.
Each of the Snom phones has function keys programmed
Hello,
2008/1/1, Steve Underwood [EMAIL PROTECTED]:
Hi Rob,
Rob Hillis wrote:
Well that answers that question. I see that t38modem provides an H232
modem - is this unsuitable for HylaFAX's purpose? (ignoring the fact
that it requires a kernel recompile on most newer distros.)
Steve
This is the whole point behind using VLAN on the phone. Tagged VLAN
for your phone with QoS configured accordingly on your switch and
untagged VLAN for your PC, both on the same wire. This way you can
always guarantee enough bandwidth for your VoIP packets.
Thanks,
Wojtek
On 2-Jan-08, at
I have an analog GSM Gateway that is connected to a normal SIP ATA device.
Basically what it does is this : when you call the extension nr. of the
SIP ATA port, the GSM Gateway will pick up the phone and presents a (new)
dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia
Sorry, should have put that in my reply.
core show hints shows me all the hints as I would expect to see them,
for example:
[EMAIL PROTECTED]: SIP/510 State:Idle
Watchers 6
and core show subscriptions also shows me all the subscriptions, again
as I would expect to see them
On Thursday 03 January 2008 15:28:15 Remco Barendse wrote:
I have an analog GSM Gateway that is connected to a normal SIP ATA device.
Basically what it does is this : when you call the extension nr. of the
SIP ATA port, the GSM Gateway will pick up the phone and presents a (new)
dial tone,
Steve Totaro wrote:
Jay Moore wrote:
Greetings, List.
I'm having a problem where my recorded calls are skipping every 4-5
seconds are so. I can hear the caller (or callee) just fine and then a
second or so of silence followed by the person talking again. I'm
saving my calls as .gsm
Does anybody have an idea where I can start looking to fix this?
Or is this regular behaviour of asterisk that it does not show an
extension as busy when it initiated the call?
Thanks,
Lars
--
Zymurgy's Law of Volunteer Labor:
People are always available for work in the past tense.
On Thu, 3 Jan 2008, Benchev wrote:
Basically Grandstream HT286 is a single port FXS ATA.
In order to interconnect GSM gateway one would need FXO.
Are you sure it gives you new dialing tone or this is the * itself
you hear?
Yes, i am positive that i get a new dialtone from the GSM Gateway.
Hi list,
Attempting to get an ISDN-BRI line connected using an HFC-S PCI card
together with Asterisk v1.4.14 and Zaptel 1.4.7 on a Debian etch
system, I find that I can't access the card's resources because the
channels are always be busy. An attempt to call out results in the
following
On Thu, Jan 03, 2008 at 04:08:10PM +0100, Jaap Winius wrote:
Hi list,
Attempting to get an ISDN-BRI line connected using an HFC-S PCI card
together with Asterisk v1.4.14 and Zaptel 1.4.7 on a Debian etch
system, I find that I can't access the card's resources because the
channels are
Remco Barendse wrote:
I have an analog GSM Gateway that is connected to a normal SIP ATA device.
Basically what it does is this : when you call the extension nr. of the
SIP ATA port, the GSM Gateway will pick up the phone and presents a (new)
dial tone, and then dials whichever DTMF tones
Hello Phil,
please check the following details in your asterisk configuration and on
your phones. These are the settings that work for me:
sip.conf
[general]
limitonpeers=yes
allowsubscribe=yes
notifyringing=yes
notifyhold=yes
useclientcode=yes
canreinvite=yes
[user1]
secret=user1
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
What is the output of:
pri show spans
PRI span 1/0: Provisioned, Down, Active
PRI span 2/0: Provisioned, Down, Active
Do incoming calls work?
I haven't configured that yet.
Interesting... which one of those two is used?
Good question.
Hello everybody,
I'd like to have more detailed records for calls related to queues. For
instance, if A enters in queue X, waits for Y secs and then talks to peer Z for
T seconds, I'd like to have two entries in my CDR:
- src: A, dst: X, duration: Y, state: ANSWERED
- src: A, dst: Z, duration: T,
On 15:38, Thu 03 Jan 08, Remco Barendse wrote:
On Thu, 3 Jan 2008, Benchev wrote:
Basically Grandstream HT286 is a single port FXS ATA.
In order to interconnect GSM gateway one would need FXO.
Are you sure it gives you new dialing tone or this is the * itself
you hear?
Yes, i am
Jay Moore wrote:
Steve Totaro wrote:
Jay Moore wrote:
Greetings, List.
I'm having a problem where my recorded calls are skipping every 4-5
seconds are so. I can hear the caller (or callee) just fine and then a
second or so of silence followed by the person talking again. I'm
Quoth Phil Knighton [EMAIL PROTECTED]
I've incorporated the kind responses from other list members, such as
setting call limits but to no avail! I've checked the function key
settings on the Snom, and adjusted it to match the suggestion from
another list member - nothing. Not one lamp on any of
On Thu, Jan 03, 2008 at 12:22:57PM -0500, Steve Totaro wrote:
How much uptime was on the server? If and when it happens again, run
top and look at CPU and memory usage.
Just the obvious comment here:
In top I see:
Mem:483588k total, 475448k used, 8140k free,64856k buffers
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Not sure where to report this...
http://www.asterisk.org/downloads
Right hand download box, Asterisk 1.4.17 points to 1.4.1
Just a heads up.
Stu
- --
And all I can do is keep on telling you, I want you, I need you,
But there aint no way Im ever
Stuart Sheldon wrote:
http://www.asterisk.org/downloads
Right hand download box, Asterisk 1.4.17 points to 1.4.1
Sorry about that. It appears to be already fixed, though ...
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
Quoth robert boardman [EMAIL PROTECTED]
Tzafrir Cohen wrote:
On Wed, Jan 02, 2008 at 07:41:40PM +, Russell Brown wrote:
Something in the, fairly, recent series of Asterisk updates has broken
DIGITAL call passthrough.
I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1
On Thu, 2008-01-03 at 18:00 +, Russell Brown wrote:
Quoth Phil Knighton [EMAIL PROTECTED]
I've incorporated the kind responses from other list members, such as
setting call limits but to no avail! I've checked the function key
settings on the Snom, and adjusted it to match the
Andrea Spadaccini wrote:
Hello everybody,
I'd like to have more detailed records for calls related to queues. For
instance, if A enters in queue X, waits for Y secs and then talks to peer Z
for
T seconds, I'd like to have two entries in my CDR:
- src: A, dst: X, duration: Y, state:
Watch the SIP Trace page on the Snom.
1. When it boots, it should send out a Subscribe message.
2. When the other phone is getting a call, the Snom should receive a
Notify messages to tell the state.
This might be a little out of date, but the main info is there:
I used to work for Telefonica of Puerto Rico installing Asterisk, so I
have installed few of them.
I installed one last week (downloaded and installed the latest) and
everything went beautiful and every thing works fine, however, my client has
voice mail and no matter what phone I use, or what
Is there any possibilletys to klick on
a telephone nr an it will dail like the case in a mail program if you klick a
url://a.b.se it opens a browser
and in this case it would open a dailplane ??
Is there sucha thing ?
Asking just out of curisoty
/Fredrik Söderlund
I think Snapanumber might be what you are looking for.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Fredrik Söderlund
Upgraded to 1.4.17 and found that the parking slot is not announced.
Reverted back and all is well. Anyone else notice this behavior?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE
On Thursday 03 January 2008 22:15:07 William Herrera wrote:
I installed one last week (downloaded and installed the latest) and
everything went beautiful and every thing works fine, however, my client
has voice mail and no matter what phone I use, or what password I enter, or
in which way I
Brent Torrenga wrote:
Upgraded to 1.4.17 and found that the parking slot is not announced.
Reverted back and all is well. Anyone else notice this behavior?
If that is the case, put it on bugs.digium.com and it will get taken care of. I
will try to take a look at it, as I think I may have
Hi,
I'm wondering whether or not it is achievable to build a web based
click2dial application that could automatically detect that a user is
connected from office or home.
Another option is to directly ask user or let them change default option but
having this automatically detected is a bonus.
On Thursday 03 January 2008 16:38:35 Remco Barendse wrote:
On Thu, 3 Jan 2008, Benchev wrote:
Basically Grandstream HT286 is a single port FXS ATA.
In order to interconnect GSM gateway one would need FXO.
Are you sure it gives you new dialing tone or this is the * itself
you hear?
Yes,
- Original Message -
From: Lars Bensmann [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, January 03, 2008 4:24 PM
Subject: Re: [asterisk-users] BLF trouble
Does anybody have an idea where I can start looking to fix this?
Or is this regular behaviour of
- Original Message -
From: Steve Langstaff [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, January 03, 2008 11:49 AM
Subject: Re: [asterisk-users] How to automaticaly closecallswhenAsterisk
didn't receive the
The created extension its set to default (rfc2833). This is something I have
never had the need to change ... (with the older versions of Asterisk)
WH
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benchev
Sent: Thursday, January 03, 2008 4:44 PM
To:
Can you provide more details on what you are trying to do. Your
explanation is a bit confusing - sounds interesting but just want to
make sure I have your idea right.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney
Some of the grandstream phones refuse to listen to Asterisk, so you have
to set them manuallygr.
PaulH
On Thu, 2008-01-03 at 17:23 -0400, William Herrera wrote:
The created extension its set to default (rfc2833). This is something I have
never had the need to change ... (with the
That is correct; we would not recommend using just *any* CF card, as the
write speed of the card needs to be pretty high to be able support
multiple voicemail messages being written simultaneously. With that
said, though, it is possible to use a higher capacity CF card, but my
previous response
Yea, sounds like they've planned for this issue. Kevin, is there an sdk that
can be used to create our own binaries should we want to add modular support
for something? Like say mysql cdr's?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian J.
Olivier wrote:
Hi,
I'm wondering whether or not it is achievable to build a web based
click2dial application that could automatically detect that a user is
connected from office or home.
Another option is to directly ask user or let them change default option
but having this automatically
Ciao Atis,
Do you think that it's a good idea? How can it be implemented? I see that
uniqueid changes for each call in the scenario that I described, so I'm a
bit stuck.
I'm using asterisk 1.2.2x (I know that I should migrate.. This is the last
release of our product that uses 1.2).
Hello Asterisc-Users List,
I am new to the list. I joined with a question in mind: How would you
set up an asterisc box so that:
(A) Someone dials a number
(B) They are presented with a menu
(C) Entering a number, like 1, connects a call to me.
(D) I am on a mixing board, running an internet
MatsK wrote:
Olivier wrote:
Hi,
I'm wondering whether or not it is achievable to build a web based
click2dial application that could automatically detect that a user is
connected from office or home.
Another option is to directly ask user or let them change default option
but having
hello list, happy new year to all, also to digium for their great work
with asterisk .
I want to make an automatic call marking an extension from my dial
plan , an example that when marking the extension 100, tell me it
records their message, mark the hour of their automatic call and at
the end
nothing? :'(
On Dec 25, 2007 1:59 AM, Kerry S [EMAIL PROTECTED] wrote:
Unfortunately I don't have a server set up that supports G.728.
I'm asking for a software project. They also don't have the immediate
resources. The goal of the project is to have a comprehensive VoIP
conversation
On Thursday 03 January 2008 19:36:10 Shane D wrote:
Hello Asterisc-Users List,
That's Asterisk, with a K. I do have to say, I've never seen that particular
misspelling before, though.
I am new to the list. I joined with a question in mind: How would you
set up an asterisc box so that:
(A)
Asterisk doesn't support g728.
Any idea what does?
PaulH
On Thu, 2008-01-03 at 20:57 -0600, Kerry S wrote:
nothing? :'(
On Dec 25, 2007 1:59 AM, Kerry S [EMAIL PROTECTED] wrote:
Unfortunately I don't have a server set up that supports
G.728.
I'm
Would this be a firewall problem?
chan_sip.c handle_request_register: Registration from sip failed for
ACL error (permit/deny)
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or
At 23:45 1/3/2008, Doug wrote:
Would this be a firewall problem?
chan_sip.c handle_request_register: Registration from sip failed for
ACL error (permit/deny)
Nope. I just needed to reload the configuration
so that the phone could register.
dean, fredrik,
when I installed skype, ugh, it asked me if I wanted to link phone
numbers on the web page to be click2dial... I did it and every phone
number on a web page was a link... I ended up turning it off... it was
too annoying... so there are some plug-ins out there that can do that
Hi,
I have callcenter running with v 1.2 with AgentCallbackLogin and now
trying to move to 1.4 using the example doc,
doc/queues-with-callback-members.txt. From what I understand the basic
idea in the example is to
1. Authenticate a caller with VMAuthenticate
2. Get his SIP Channel number
3. Use
Hi,
I have callcenter running with v 1.2 with AgentCallbackLogin and now
trying to move to 1.4 using the example doc,
doc/queues-with-callback-members.txt. From what I understand the basic
idea in the example is to
1. Authenticate a caller with VMAuthenticate
2. Get his SIP Channel number
3. Use
56 matches
Mail list logo
