6 jul 2012 kl. 09:29 skrev Elliot Murdock:
Hello,
Thank you for the clarification.
Just a few questions:
1. What is the Timer1 used for?
Timer1 is the base for many other SIP timers and it's an estimate of the
roundtrip time for a packet
between two SIP devices or servers. TimerB is
6 jul 2012 kl. 23:18 skrev Felix Salfelder:
Hi there.
i am seriously stuck with an asterisk and sip problem.
the following sip.conf works with respect to some_peer:
[general]
bindaddr = x.y.z.w
nat = no
[some_peer]
type=peer
host=somehost
secret=somesecret
some other
7 jul 2012 kl. 21:07 skrev Mikhail Lischuk:
Thomas Perron писал 07.07.2012 21:48:
exten = s,n,Dial(SIP/16175551212)
sip.conf
[general]
;register = 125010155:funnyti...@sip3.voipvoip.com/125010155
register = 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11
;
[incoming]
9 jul 2012 kl. 15:24 skrev Sergio Serrano:
Hi all
I hope that someone of you can solve this. Right now I'm stuck!
I'm using asterisk with some SIP extensions. Basically I want to
establish a call between desktop voip phone (ext 181) and embedded sip
system (ext 182)
All I can see
On Tue, Jul 10, 2012 at 10:24:18AM +0200, Olle E. Johansson wrote:
The Asterisk SIP channel has no knowledge about interfaces and can't
bind to a specific interface for communication.
Thanks for the reply.
in the meantime i've found a sort of workaround.
[general]
host = dynamic
; take some
Hi,
I have a server running at more than two years with Asterisk 1.6, and began
presenting problem seedlings links in external SIP extensions on some links.
By doing rtp set debug on discovered the problem, he is trying to deliver
the audio directly to internal IP Extension. And sometimes shown
Hi list,
TRUNkA=Dahdi/g0 {g0=1-15,17-31}
TRUNKB=Dahdi/g1 {g1=32-46,48-62}
I have 2 gsm channel banks its E1 connection , its connected to server. I
define this 2 different trunks.
for example like TrunkA,TrunkB.
TRUNKA connected 1 gsm channel bank and TRUNKB connected 2 gsm channel
bank. if
On Tue, Jul 10, 2012 at 10:45 AM, Daviramos Roussenq Fortunato
daviramo...@gmail.com wrote:
I have a server running at more than two years with Asterisk 1.6, and began
presenting problem seedlings links in external SIP extensions on some links.
By doing rtp set debug on discovered the problem,
Channels can be in more than one group.
Make g0=1-15,17-31,32-46,48-62 and
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta
Sent: Tuesday, July 10, 2012 10:04 AM
To: Asterisk Users Mailing List -
Thank you for your reply.
not like this. because there is 2 different types of calling is there thats
why .
On Tue, Jul 10, 2012 at 7:39 PM, Eric Wieling ewiel...@nyigc.com wrote:
Channels can be in more than one group.
Make g0=1-15,17-31,32-46,48-62 and
-Original Message-
From:
Is there a limit to the number of connections
that manager can handle at one time?
In my logs I see connect error but then try again
in a few seconds and it works.
I could have quite a number of connections at one time.
How can I up the limit.
Jerry
--
Is there a limit to the number of connections
that manager can handle at one time?
In my logs I see connect error but then try again
in a few seconds and it works.
I could have quite a number of connections at one time.
How can I up the limit.
Jerry
--
I've installed 10.6.0-rc2 on two machines. On one of the machines (but
not the other) /tmp gets filled with:
...
-rw---. 1 asterisk asterisk 53661696 Jul 7 23:46
core.PBX-2012-07-07T23:46:10-0400
-rw---. 1 asterisk asterisk 53891072 Jul 7 23:48
- Original Message -
From: sean darcy seandar...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, July 10, 2012 10:42:20 AM
Subject: [asterisk-users] 10.6.0-rc2: tmp full of core.PBX
I've installed 10.6.0-rc2 on two machines. On one of the machines
(but
not the other)
On 07/10/2012 11:44 AM, Matthew Jordan wrote:
- Original Message -
From: sean darcy seandar...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, July 10, 2012 10:42:20 AM
Subject: [asterisk-users] 10.6.0-rc2: tmp full of core.PBX
I've installed 10.6.0-rc2 on two machines. On
Hi,
The flowroute website mentions that they set callerid on outbound calls
based on the presence of (in order of preference):
P-Asserted-Identity, Remote-Party-ID or From:.
I've been trying to make outbound callerid work via flowroute to no
avail. Does anyone have an extensions.conf /
On Tue, Jul 10, 2012 at 9:04 AM, mahesh katta maheshka...@flexydial.comwrote:
Hi list,
TRUNkA=Dahdi/g0 {g0=1-15,17-31}
TRUNKB=Dahdi/g1 {g1=32-46,48-62}
I have 2 gsm channel banks its E1 connection , its connected to server. I
define this 2 different trunks.
for example like
SIPAddHeader() comes to mind. :-)
-- Alex
--
Sent from my Samsung mobile, and thus lacking in the refinement one might
expect from a proper keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web:
On 10-07-12 18:29, Alex Balashov wrote:
SIPAddHeader() comes to mind. :-)
Yup I got that far :) I tried things like (with correct name number):
exten = _1ZX,1,SipAddHeader(P-Asserted-Identity: Global
Minties Corp sip:19995551212@AST_BOX_FQDN)
But that did not work as flowroute
At 09:20 AM 7/10/2012, you wrote:
I've been trying to make outbound callerid work via flowroute to no
avail. Does anyone have an extensions.conf / sip.conf snippet howto
make this work? This is for Asterisk 1.4.44.
This is a section of code I use to choose outgoing callerid for my
Flowroute
Check your users.conf - this looks like an override issue to me.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists
Sent: Tuesday, July 10, 2012 11:45 AM
To: asterisk-users@lists.digium.com
Subject:
On Tue, Jul 10, 2012 at 11:45 AM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 10-07-12 18:29, Alex Balashov wrote:
SIPAddHeader() comes to mind. :-)
Yup I got that far :) I tried things like (with correct name number):
exten =
On 10-07-12 18:49, Warren Selby wrote:
You can't* set the outbound name. That's defined in the national caller
id name database that the receiving phone company dips into. As far as
I know, Flowroute does not add entries to this database, nor do they dip
it when you receive a call to pass the
- Original Message -
Thanks Tim.
One of the problem that I am facing is the complicated generated
configuration for the FreePBX, is it the same thing in the Elastix?
To understand this complicated generated commands, is there a
documentation to explain this for FreePBX or Elastix?
On Tue, Jul 10, 2012 at 12:34 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
Thank you for your feedback Warren. I removed the outbound name but still
get random numbers VOIP CALLER on outbound calls. Googling I tried some
more:
SipAddHeader(P-Asserted-**Identity:
On Tue, Jul 10, 2012 at 12:39 PM, Tim Nelson tnel...@rockbochs.com wrote:
Most of the predone projects (Elastix is my favorite at the moment)
include some sort of endpoint manager that will generate configs for your
phones. I'm not sure specifically on Cisco phones, other than they are a
huge
On 10-07-12 18:48, Danny Nicholas wrote:
Check your users.conf - this looks like an override issue to me.
Thank you for your feedback Danny. users.conf is default and has not
been touched.
Regards,
Patrick
--
_
--
On 10-07-12 18:47, Ira wrote:
At 09:20 AM 7/10/2012, you wrote:
I've been trying to make outbound callerid work via flowroute to no
avail. Does anyone have an extensions.conf / sip.conf snippet howto
make this work? This is for Asterisk 1.4.44.
This is a section of code I use to choose
On 10-07-12 19:48, Warren Selby wrote:
On Tue, Jul 10, 2012 at 12:34 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl
mailto:asterisk-l...@puzzled.xs4all.nl wrote:
Thank you for your feedback Warren. I removed the outbound name but
still get random numbers VOIP CALLER on outbound
I'm currently trying to decide on which GUI-enabled version of Asterisk to
use for one particular installation, where we will need good telecommuter
support. We've made it so easy for people to work remotely that the
customer is downsizing their real estate and will have 90% remote workers
with
On 07/10/2012 01:42 PM, Carlos Alvarez wrote:
I'm currently trying to decide on which GUI-enabled version of Asterisk
to use for one particular installation, where we will need good
telecommuter support. We've made it so easy for people to work remotely
that the customer is downsizing their
- Original Message -
I'm currently trying to decide on which GUI-enabled version of
Asterisk to use for one particular installation, where we will need
good telecommuter support. We've made it so easy for people to work
remotely that the customer is downsizing their real estate and
On 07/10/2012 03:24 AM, Olle E. Johansson wrote:
The Asterisk SIP channel has no knowledge about interfaces and can't
bind to a specific interface for communication. In fact, it's a well known
bug that if you have multiple interfaces with different IP networks,
Asterisk will send from the wrong
On Tue, Jul 10, 2012 at 11:46 AM, Kevin P. Fleming kpflem...@digium.comwrote:
This can be done using Digium phones; they have built-in support for
selecting which 'user' they should be when they are reconfigured. It's
slightly more complicated than a simple login/logout because it requires
On 10-07-12 20:42, Carlos Alvarez wrote:
I'm currently trying to decide on which GUI-enabled version of Asterisk
to use for one particular installation, where we will need good
telecommuter support. We've made it so easy for people to work remotely
that the customer is downsizing their real
On 07/10/2012 01:50 PM, Carlos Alvarez wrote:
On Tue, Jul 10, 2012 at 11:46 AM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
This can be done using Digium phones; they have built-in support for
selecting which 'user' they should be when they are
The Asterisk Development Team has announced the release of Asterisk 1.8.14.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.14.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team has announced the release of Asterisk 10.6.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 10.6.0 resolves several issues reported by the
community and would have not been possible
On 10/7/12 7:46 pm, Tim Nelson wrote:
Not to sound like a broken record or anything... but I'd say give Elastix a go.
It is top notch in terms of release quality and features. And, being based on
FreePBX, you can set it to 'Device and User' mode instead of the default
extensions mode so users
Recent Polycom firmware versions (4.x, I think) also have support for user
sort of stuff. See the 4.x Admin Guide.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Tuesday, July 10,
On July 12, 2012 from approximately 11:00AM to 11:30AM (Central Daylight
Time, GMT-5), the core routers that provide connectivity through to all
Asterisk community services will be swapped out.
This will mean that these services will be unavailable during most, if
not all, of this time
I went for admin/module admin and I search for custom contexts but did not find
it. How I can get it?
Regards
Bilal
---
The module is custom contexts - its a third party option in
the module admin
But you can write contexts in the extensions_custom.conf if
you want to
I
Please don't top-post.
On Tue, 10 Jul 2012, bilal ghayyad wrote:
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Tuesday, July 10, 2012 5:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FreePBX: using
Dear Warren;
I did not understand the example below well.
What the Verbose will do? It will write in the CDR or the database? Really this
did not understand.
Also did not understand this lineL same = n,Goto(${EXTEN},from-internal,1) How
it will work? Can u plz explain?
Regards
Bilal
- Original Message -
From: bilal ghayyad bilmar...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, July 10, 2012 4:25:40 PM
Subject: Re: [asterisk-users] FreePBX: using context other than the default
context and the generation for the
configuration
Dear Warren;
On Mon, Jul 2, 2012 at 12:13 AM, Olle E. Johansson o...@edvina.net wrote:
No.
This is probably because you are using phone numbers as names of devices with
type=friend in sip.conf.
That's generally a bad idea.
The SIP channel matches an incoming call this way:
1. Take the From: user
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