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From: Phuong Hoang ducphuongbk200...@gmail.com
Date: Thu, Nov 18, 2010 at 9:16 AM
Subject: Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit
To: asterisk-users@lists.digium.com
Hi all,
I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5
Hi all,
I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but
not successful, Can anyone help me to do it?
Thanks and best regards.
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On 18 Nov 2010, at 10:36, Phuong Hoang wrote:
I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but
not successful, Can anyone help me to do it?
How is this different to the other two posts? Please stop repeatedly sending
messages! If nobody replies you're probably not
On 18 Nov 2010, at 10:33, Phuong Hoang wrote:
I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but
not successful, Can anyone help me to do it?
Given that you haven't given any error messages, any logs, or your sip.conf, or
the manner in which it is not working
Hi people,
Who knows as I can do Announcement Transfer with call-limit = 1 in Asterisk 1.6.
Thanks so much.
Renato
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New to Asterisk? Join us for a
I think these will be helpful, if not for solving the problem, at least
for trying to rephrase your questing in a meaningful manner:
http://www.asteriskdocs.org
http://www.asteriskguide.com
and particularly
http://www.asteriskguide.com/pdf/GettingStartedWithAsterisk.pdf
Bert
On 18/11/2010
Hi All
Does anyone know about any tool that does to Asterisk what mod_jk does for
JBoss/Tomcat: a load-balance/failover server that is constantly connected to
Asterisk backend servers and is capable of identify loaded or down servers?
Regards
Antônio Theóphilo
smime.p7s
Description: S/MIME
On Wed, Nov 17, 2010 at 6:17 PM, Matthew Fredrickson cres...@digium.com wrote:
On 11/17/10 2:44 PM, Cary Fitch wrote:
In regard to #2, any T1 card should work. But the problem is you need SS7
software and SS7 connectivity in addition to the T1 card.
Asterisk (as of version 1.6.0 or greater)
You could use a sip proxy front end like Kamailio.
Sent from my iPhone
On Nov 18, 2010, at 7:39 AM, Antônio Theóphilo anto...@freeddom.com wrote:
Hi All
Does anyone know about any tool that does to Asterisk what mod_jk does for
JBoss/Tomcat: a load-balance/failover server that is
On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneen mden...@gmail.com
wrote:
Are you saying ADSL as in a generic term for broadband router or do
you really mean that the router also acts as a DSL transceiver?
Sorry about that. Ideally, the unit should be both an ADSL modem +
router, but apparently,
On 11/18/2010 07:52 AM, Gilles wrote:
On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneenmden...@gmail.com
wrote:
Are you saying ADSL as in a generic term for broadband router or do
you really mean that the router also acts as a DSL transceiver?
Sorry about that. Ideally, the unit should be both
On Tue, Nov 16, 2010 at 8:28 AM, Gilles codecompl...@free.fr wrote:
Hello
For users who 1) don't have a QoS-capable ADSL router and 2) would
like to run Asterisk with a couple of SIP trunks, I was wondering what
hardware is recommend to run any of the main open-source *WRT projects
to which
On 11/18/2010 10:02 AM, Chris Gentle wrote:
On Tue, Nov 16, 2010 at 8:28 AM, Gilles codecompl...@free.fr
mailto:codecompl...@free.fr wrote:
Hello
For users who 1) don't have a QoS-capable ADSL router and 2) would
like to run Asterisk with a couple of SIP trunks, I was wondering
Thank you for the answer Darren.
In fact I have an application that requests a call to a real person through an
AMI interface and get some client information. Using a SIP Proxy is an option
but I prefer that the interface between the app and the Asterisk could be the
AMI (or HTTP).
Regards
On Thu, Nov 18, 2010 at 9:20 AM, jon pounder j...@inline.net wrote:
I have a similar setup in an office but sip directly back to the main
server - not sure what the value add to the local asterisk is, except
intercom calls would not have to leave the lan, but isn't that the purpose
of
On 11/18/10 7:40 AM, Matt wrote:
On Wed, Nov 17, 2010 at 6:17 PM, Matthew Fredricksoncres...@digium.com
wrote:
On 11/17/10 2:44 PM, Cary Fitch wrote:
In regard to #2, any T1 card should work. But the problem is you need SS7
software and SS7 connectivity in addition to the T1 card.
On 11/18/10 10:07 AM, Matthew Fredrickson wrote:
On 11/18/10 7:40 AM, Matt wrote:
On Wed, Nov 17, 2010 at 6:17 PM, Matthew Fredricksoncres...@digium.com
wrote:
On 11/17/10 2:44 PM, Cary Fitch wrote:
In regard to #2, any T1 card should work. But the problem is you need
SS7
software and
Hi all,
We have been using asterisk 1.8 for some while now, together with asterisk 1.6.
We have the following problem. In asterisk 1.8 when you leave a voicemail, the
person at that extension is notified by email that he has a voicemail. The
voicemail is attached to that email along with
Hi,
I`ve been using Asterisk parking lots (multiple parking lots) with relative
success on 1.6.2.X. The problem that I just found was the following: When I do
park someone, and the call is parked for the duration of the timeout, the
person who parked the call gets back the parked calls
Mike wrote:
I get a
no extension ‘t’ in context ‘’
Add this to your dial plan:
[park-dial]
exten = t,1,Where to send calls go here
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
That is correct. Feel free to ask me any questions if you have any
issues come up along the way. The sample chan_dahdi.conf has a section
with an example of an SS7 setup in it, for reference on configuration.
Oh yeah, and also, there's an asterisk-ss7 mailing list at
lists.digium.com where
I tried thator I think I did something similar, but that may or may not
apply (depending on my understanding of parking lots)
Here is my relevant contexts. The SIP phones are registered under this context:
[some_context]
include = parkinglotA
include = outboundcalls
exten =
On 18 November 2010 17:43, Mike l...@net-wall.com wrote:
I tried thator I think I did something similar, but that may or may not
apply (depending on my understanding of parking lots)
Here is my relevant contexts. The SIP phones are registered under this
context:
[some_context]
Mike wrote:
So my questions:
1) Why won't this work?
I'm still running under 1.4.x
And more importantly:
2) what's this park-dial context you speak of ? Is this a hardcoded context
calls go back to?
It is under 1.4
Doug
--
Ben Franklin quote:
Those who would give up
Hi,
I tried all combinaisons of parkinglot contexts. I always get the same CLI
message:
no extension t in context
I can't find a way to define what context Asterisk should be going back to.
The park-dial context that Doug suggested didn't help, neither did my
variations on parkinglotA.
Hi,
With a dynamic Meetme using: MeetMe(|DsMrc)
How do I control which context MOH uses, other than default ?
Asterisk: 1.4.15
Thanks,
Adrian
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On Tue, Nov 16, 2010 at 9:28 AM, Gilles codecompl...@free.fr wrote:
Hello
For users who 1) don't have a QoS-capable ADSL router and 2) would
like to run Asterisk with a couple of SIP trunks, I was wondering what
hardware is recommend to run any of the main open-source *WRT projects
to which
Hi all,
I'd like that each analog trunk of my TDM410p was received in different
extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a
different context and in my extensions.conf, under [default] I put such
contexts and an especific estension to answer it. therefore,
For some reason we are seeing Avoiding deadlock for channel in our
Asterisk logs, the logs are getting filled up with an amazing speed around
12000 lines a second, and all of them are Avoiding deadlock. What could be
the potential reason for this to be happening? The Asterisk is used as auto
On Thu, 18 Nov 2010, Flavio Miranda wrote:
I'd like that each analog trunk of my TDM410p was received in different
extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each
trunk in a different context and in my extensions.conf, under [default]
I put such contexts and an
Is anybody here familiar with the meaning of INVAL packets for IAX2?
Every few days I get a dropped outgoing call in the middle of the
conversation (the outgoing call has been connected for few minutes) when
an incoming call comes in. The log reads the following when this happens:
[Nov 17
Hi Steve,
thanks for the tips Better bait = better fish !
As you said, I am in the right track.
Looking to dahdi show channles , I realized that all the trunks was in the
same context. So, I have changed this and everything works!
thanks you !!
Att,
Flavio Roberto Miranda
On Thu, 18 Nov 2010, Flavio Miranda wrote:
Looking to dahdi show channles , I realized that all the trunks was in
the same context. So, I have changed this and everything works!
That's why I prefer to work from what Asterisk parsed the file as, not
what the poster thinks :)
--
Thanks in
AstLinux on various embedded hardware really works well. I have several
on older HP Thin Clients, 55xx series, some with only 128 Meg of ram. A
replacement flash from Transcend and a 55xx of eBay can be easily under $50.
the GUI in AstLinux makes life simple for users who want to make minor
Hi everybody,
since some time I am looking for a current and reliable solution to send
and receive faxes (probably fax-2-mail and mail-2-fax) in conjunction
with Asterisk.
For testing I am using a HFC-ISDN passive PCI-card, in production a
Digium Dual T1/E1 PCI-card will be used.
I run CentOS
On Thu, 18 Nov 2010, John Novack wrote:
AstLinux on various embedded hardware really works well. I have several on
older HP Thin Clients, 55xx series, some
with only 128 Meg of ram. A replacement flash from Transcend and a 55xx of
eBay can be easily under $50.
the GUI in AstLinux makes
Im using
asterisk-1.6.2.13
asterisk-addons-1.6.2.2
dahdi-linux-complete-2.4.0+2.4.0
libpri-1.4.11.4
spandsp-0.0.6
Sangoma Hardware, using wanpipe-3.5.17
Extensions.conf:
[fax-in]
exten = s,1,Answer()
exten = s,n,Wait(1)
exten =
On Thu, Nov 18, 2010 at 8:52 AM, Gilles codecompl...@free.fr wrote:
On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneen mden...@gmail.com
wrote:
Are you saying ADSL as in a generic term for broadband router or do
you really mean that the router also acts as a DSL transceiver?
Sorry about that.
Hello
For users who 1) don't have a QoS-capable ADSL router and 2) would
like to run Asterisk with a couple of SIP trunks, I was wondering what
hardware is recommend to run any of the main open-source *WRT projects
to which Asterisk has been ported:
Jeff LaCoursiere wrote:
On Thu, 18 Nov 2010, John Novack wrote:
AstLinux on various embedded hardware really works well. I have several on
older HP Thin Clients, 55xx series, some
with only 128 Meg of ram. A replacement flash from Transcend and a 55xx of
eBay can be easily under
On Thu, Nov 18, 2010 at 9:26 AM, Darrick Hartman
dhart...@djhsolutions.com wrote:
On 11/18/2010 07:52 AM, Gilles wrote:
On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneenmden...@gmail.com
wrote:
Are you saying ADSL as in a generic term for broadband router or do
you really mean that the router
On Thu, Nov 18, 2010 at 12:37 PM, Adrian Marsh
adrian.ma...@ubiquisys.com wrote:
Hi,
With a dynamic Meetme using: MeetMe(|DsMrc)
How do I control which context MOH uses, other than “default” ?
Asterisk: 1.4.15
Thanks,
Adrian
--
On Thu, Nov 18, 2010 at 12:37 PM, Adrian Marsh
adrian.ma...@ubiquisys.comwrote:
Hi,
With a dynamic Meetme using: MeetMe(|DsMrc)
How do I control which context MOH uses, other than “default” ?
Asterisk: 1.4.15
In 1.4.x you would use SetMusicOnHold(class) before you called your
On 11/18/2010 10:38 PM, Tilghman Lesher wrote:
On Thursday 18 November 2010 14:01:49 Sebastian wrote:
Is anybody here familiar with the meaning of INVAL packets for IAX2?
Every few days I get a dropped outgoing call in the middle of the
conversation (the outgoing call has been
Hi,
I tried to perform call forward in asterisk by writing the following in the
dial plan.The data base is getting updated with the caller ID number how
ever the call is not getting forwarded.
[apps]
exten = _*21*XX,1,Set(DB(CFIM/${CALLERID(number)})=${EXTEN:4})
exten = _*21*XX,2,Hangup
exten =
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