On Tue, 14 Dec 2010 11:19:48 -0600, Lyle Giese l...@lcrcomputer.net
wrote:
You are setting up a SIP trunk from your VOSP provider(whatever VOSP
is). It dials your phone number. So whatever you dial from your cell
phone is the extension that this trunk should land at.
's' is not an extension. It's
The system that hosts the Asterisk community mailing lists
(lists.digium.com) experienced some failures yesterday, and as a result
the lists have been moved to a new system (with the same name).
During this process, it is possible that outbound messages queued for
delivery to some list
Hi
We are using asterisk 1.4.17 for the apt repository on an Ubuntu server
and we're getting an odd problem with one customer using a Queue
The queue is called in the dialplan with the options Tn
The queue only has one member.
Occasionally and starting to get more frequently the caller ends up
Hi All;
I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8?
For example, when to decide that I have to go for 1.6 or I have to go for 1.8?
Regards
Bilal
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Hi All,
Anyone out there successfully tested Asterisk 1.8 with Web-Meetme 4.0v in my
case my asterisk got crashed when i dialing conf room number.
Best,
S
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Hi Tilghman,
I am indeed still seeing this issue (emails missing in sequence, and
therefore voicemail box not readable), and I have absolutely no third-party
vendor solution playing with voicemails.
How do I find whether this was a simple bug that was found and fixed in
between official
Hi All,
I have issue with res_odbc.so module Asterisk 1.8 not allowing me to enable
because its depended on generic_odbc and ltdl
I did install unixodbc and ltdl but still same error
Thanks,
Satish
--
Hi list,
For a telecom project I need to setup a SCADA solution. I don't have any
previous experience in this type of monitoring and automization. I'll be
using SNMP data from asterisk servers and endpoints. If anybody has any
suggestion which SCADA software can fit in such a VoIP solution, your
The Asterisk Development Team has announced the release of Asterisk 1.8.1.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.1.1 resolves two issues reported by the community
since the release of Asterisk
Greetings folks-
I'm experiencing issues with a freshly installed box. When a call comes in via
PRI (Sangoma AFT-A104), I see this in my logs:
[Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo
cancellation on channel 12 (Invalid argument)
[Dec 15 14:26:10] WARNING[23546]
On Wednesday 15 December 2010 13:53:12 satish patel wrote:
I have issue with res_odbc.so module Asterisk 1.8 not allowing me to
enable because its depended on generic_odbc and ltdl
I did install unixodbc and ltdl but still same error
Make sure you re-run ./configure after you add/remove
Le 15/12/2010 15:21, Gilles a écrit :
[...]
;IMPORTANT: outgoing must be BEFORE incoming
[vosp_outgoing]
type=peer
host=myvosp.com
username=myaccount
secret=mypasswd
fromuser=myaccount
fromdomain=myvosp.com
nat=yes
canreinvite=no
[vosp_incoming]
type=peer
host=myvosp.com
nat=yes
canreinvite=no
What are your needs? Perhaps use stable 1.4 if the provided features
suffice you.
Am 15.12.2010 15:46, schrieb bilal ghayyad:
Hi All;
I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8?
For example, when to decide that I have to go for 1.6 or I have to go for 1.8?
there is no reason not to use 1.8 when you start a new installation.
1.8 is the new five years long term support version
Kristijan
2010/12/15 bilal ghayyad bilmar...@yahoo.com:
Hi All;
I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8?
For example, when to decide
Hi,
Looking at dmesg, I can see :
[ 45.213205] wcb4xxp :01:07.0: Span 5 has invalid sync priority (5),
removing from sync source list
[ 45.213205] wcb4xxp :01:07.0: Span 6 has invalid sync priority (6),
removing from sync source list
[ 45.213205] wcb4xxp :01:07.0: Span 7 has
Hello
At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT
set up with a VOSP trunk that I can use to make/receive calls to/from
the PSTN.
Now, I'd like to be able to call any number on the Net that is
advertised as sip:u...@domain.com, such as those:
Tried the following but no luck:
exten = _53.,1,Set(CALLERID(num)=473520)
exten = _53.,n,Dial(SIP/${ext...@ss74)
I am still passing IMSI310410381554227 as the CALLERID.
My peer is setup as follows:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
I had two asterisk servers connected with each other, both were 1.4.22
but I've upgraded one to 1.4.37 and now I get a message when I try call from
asterisk-1.4.22 to asterisk-1.4.37
ERROR[4539]: chan_iax2.c:4330 handle_call_token: Call rejected,
CallToken Support required. If unexpected,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Wednesday, December 15, 2010 8:47 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Which version to use: 1.4 or 1.6 or 1.8
Hi
I'm currently using Web-Meetme 4.0.2 with Asterisk 1.8.0 with no
problems. Maybe you could post more information about the crash?
On Wed, Dec 15, 2010 at 11:46 AM, satish patel satish...@hotmail.com wrote:
Hi All,
Anyone out there successfully tested Asterisk 1.8 with Web-Meetme 4.0v in
my
On Wed, 2010-12-15 at 14:33 +, Ishfaq Malik wrote:
Hi
We are using asterisk 1.4.17 for the apt repository on an Ubuntu server
and we're getting an odd problem with one customer using a Queue
The queue is called in the dialplan with the options Tn
The queue only has one member.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eddie Mikell
Sent: Wednesday, December 15, 2010 7:39 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Two asterisk servers,two different service
1. Per http://www.voip-info.org/wiki/view/Asterisk+video: Asterisk does not
provide any video transcoding capabilities
2. You can turn off video support on a peer like this:
disallow=h261
disallow=h263
disallow=h263p
From: asterisk-users-boun...@lists.digium.com
Hi there!
Anybody knows why I am receiving this output from CLI:
No such command 'dialplan reload' (type 'core show help dialplan reload' for
other possible commands)
Look like asterisk dont see dialplan?
Is it possible to restart it ?
Thansk
Att,
Flavio Roberto Miranda
3 for advanced options does not have an effect _while_ the
vm-message is played. all other options like 7 delete or 6 next
message are
working. after the vm-message is played, it's working. the question:
is this intentional? or a bug?
the available documentation does not describe this case.
On 10-12-15 09:46 AM, bilal ghayyad wrote:
Hi All;
I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8?
For example, when to decide that I have to go for 1.6 or I have to go for 1.8?
It depends on your required usage (features available in version) and your
required
On Wed, Dec 15, 2010 at 1:53 PM, satish patel satish...@hotmail.com wrote:
Hi All,
I have issue with res_odbc.so module Asterisk 1.8 not allowing me to enable
because its depended on generic_odbc and ltdl
I did install unixodbc and ltdl but still same error
Thanks,
Satish
Try
Here is how I would do it:
First, come up with a numbering scheme. For instance, all extensions in
location 1 are 1xxx, extensions in location 2 are 2xxx. External calls are
9xxx-xxx-
In Location 1:
Sip Trunk 1 - goes to Location 2. The dial plan uses it as outgoing trunk for
all calls
Hi all,
Last night I went to replace an Asterisk 1.4 + mISDN + b410p box with
Asterisk 1.6.2 and DAHDI BRI - to no avail.
I had two servers so copied network setting etc from the working one,
moved the card across, ran dahdi_genconf etc and it didn't work.
Here's the console output with
Hi all,
(Sorry if this is the second email - didn't see the first)
Last night I went to replace an Asterisk 1.4 + mISDN + b410p box with
Asterisk 1.6.2 and DAHDI BRI - to no avail.
I had two servers so copied network setting etc from the working one,
moved the card across, ran dahdi_genconf
Just add something like this to your dialplan:
exten=1234,1,Dial(SIP/u...@domain.com)
Then, when you dial 1234 on your XLite, it will connect you to u...@domain.com.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On Thu, 16 Dec 2010 10:06:35 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Why 2 context? Todays Asterisk versions only needs one peer context for
incoming/outgoing. Something like
I tried combining the two sections in sip.conf, but get a BUSY signal
for incoming calls from the PSTN. Could
Dave,
Can you capture the cli output and the sip debug of the call not doing what
it's supposed to?
Thanks,
--Warren Selby, dCAP
On Dec 16, 2010, at 6:52 AM, dave george dgeo...@teletoneinc.com wrote:
Tried the following but no luck:
exten = _53.,1,Set(CALLERID(num)=473520)
exten =
linuxinnovations.com is also a good place to seek out the differences
between the versions.
On Thu, Dec 16, 2010 at 7:00 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 10-12-15 09:46 AM, bilal ghayyad wrote:
Hi All;
I need to know which version of asterisk to use, if to be 1.4 or 1.6
Hi
Does anyone knows how to find out a call in a asterisk is external
incoming ,external out going or internal
Thanks
Nikhil
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New to Asterisk?
Hi,
Did you use libpri 1.4.11.5 or 1.4.12-beta ?
Recently l tried 1.4.11.5 on a live system and it failed (Asterisk 1.6.1.18
and dahdi trunk, Junghanns QuadBRI, PtmP lines).
Going back to 1.4.11.2 solved it.
Unfortunately, I couldn't note what error message were then generated.
Hope this helps.
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