Greetings Users!
On March 24th we're hosting the first of a new series of webinars entitled
"Asterisk Tech-Tips". Our goal is to present a new "episode" or "issue" or
"webisode" (or whatever you want to call it) every other week. Here's the idea:
Asterisk Tech-Tips are all about helping peopl
Hi Steve
I agree - past experience has been hit and miss for me too - I have to do a
dialplan reload after an external database update and have been advised
against using AMI as this sometimes hangs and a full reload is needed to get
the system going again - something I cant do with. Any ideas on
On Mon, 14 Mar 2011, Paddy Grice wrote:
I am having trouble running the command
siptest:~# asterisk -rx 'dialplan reload'
I assume the problem is timing but any ideas on how to fix it
I'm just a 1.2 Luddite, but it's been my experience that issuing shell
command lines and parsing the outp
Hi List
I am having trouble running the command
siptest:~# asterisk -rx 'dialplan reload'
most times it does what I expect and I get a response as below
siptest:~# asterisk -rx 'dialplan reload'
Dialplan reloaded.
every now and then I get no response i.e.
siptest:~# asterisk -rx 'dialpl
I did a google talk,skype, SIP, asterisk, IRC async event driven voice/IM mashup
for the voip user's conference - (see http://wp.me/pgOOh-4a for a description)
I've ended up with a thing that could (with some work) be turned into an
asynchronous
asterisk event->action framework.
The basic premi
On 03/14/2011 11:24 AM, satish patel wrote:
Thanks Kevin,
I test page application and it works but i am worried about i have 200
SIP phone. Do you think asterisk page application can handle that number
of page ?
Just worried about my asterisk. I don't want to crach :(
It doesn't matter whethe
We don't have multicast network configuration in our LAN :(
From: steve-li...@geekinter.net
Date: Mon, 14 Mar 2011 16:29:55 +
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom
On 14 Mar 2011, at 16:24, satish patel wrote:I test page appli
Oops - from the very bottom of that page (no pun intended) : "So far as
we can tell, Polycom sets do not support multicast. We certainly were
not able to find a way to use it."
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.co
...http://ofps.oreilly.com/titles/9780596517342/ch11.html if you're not
sure on Multicast (near the bottom).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven
Howes
Sent: 14 March 2011 16:30
To: Asterisk
On 14 Mar 2011, at 16:24, satish patel wrote:
> I test page application and it works but i am worried about i have 200 SIP
> phone. Do you think asterisk page application can handle that number of page
> ?
Do they support multicast?
S--
_
If I was worried I'd record the 'page' first - and then play it back to
50 handsets at a time (using a loop).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish
patel
Sent: 14 March 2011 16:25
To: asterisk
Thanks Kevin,
I test page application and it works but i am worried about i have 200 SIP
phone. Do you think asterisk page application can handle that number of page ?
Just worried about my asterisk. I don't want to crach :(
-Satish
> Date: Mon, 14 Mar 2011 11:18:36 -0500
> From: kpflem.
On 03/14/2011 10:01 AM, satish patel wrote:
Hey Guys,
I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi
stopped working look like asterisk 1.8 did some changes in manager apps
i am doing following.. my phone is ringing but not auto answer could you
give me some issue what i am do
On 03/14/2011 04:36 PM, Bruce B wrote:
Thanks for the input.
I can see the module loaded but yet the command "core show channel
types" doesn't show H323 in channel list. Maybe ooh323 is not supposed
to show in that list?
server55667*CLI> module show like 323
Module Descr
On 14 Mar 2011, at 15:58, Jonas Kellens wrote:
> dialplan :
>
> exten => 67121212,1,NoOp()
> exten => 67121212,n,Set(CALLERID(all)="3259" <3259>)
> exten => 67121212,n,SIPAddHeader(P-Preferred-Identity:
> )
> exten => 67121212,n,SIPAddHeader(Privacy: id)
> exten => 67121212,n,Dial(SIP/32
Hello,
none of the 2 SIP headers are sent...
dialplan :
exten => 67121212,1,NoOp()
exten => 67121212,n,Set(CALLERID(all)="3259" <3259>)
exten => 67121212,n,SIPAddHeader(P-Preferred-Identity:
)
exten => 67121212,n,SIPAddHeader(Privacy: id)
exten => 67121212,n,Dial(SIP/3259/6712121
Thanks for the input.
I can see the module loaded but yet the command "core show channel types"
doesn't show H323 in channel list. Maybe ooh323 is not supposed to show in
that list?
server55667*CLI> module show like 323
Module Description Use
C
Thanks guys,
I got the layer1 link up.
Edwin, I will make a cable from this link that you have posted and see if
that also works. Presently, I just did a 'manual' connect of the ends to get
the layer1 up.
Josue, many thanks for your response. Searching through this list archives,
I see that you
Hey Guys,
I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi stopped
working look like asterisk 1.8 did some changes in manager apps i am doing
following.. my phone is ringing but not auto answer could you give me some
issue what i am doing wrong ?
root@ubuntu-test:~# telnet 12
Try:
SIPAddHeader(P-Preferred-Identity: )
SIPAddHeader(Privacy: id)
That works for me in the UK.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 11 March 2011 15:06
To: Asterisk Users Mail
On 03/13/2011 05:27 PM, Bruce B wrote:
Indeed ooh323 is available as part of the RPMs and the correct URL is:
You are right. I stand corrected.
Installing h323 from source is not an issue for me. My post is very
specific and related to RPMs. I would like to hear from those who
installed using
On 03/14/2011 03:07 AM, Nick Ustinov wrote:
Hello
using asterisk 1.8, compiled res_fax.so and res_fax_spandsp.so - both
loaded successfuly
in sip.conf set t38pt_udptl=yes
but faxes still don't work even in passthru mode.
if i do a 'sip show channel' on the channel via which i am sending fax i
Hello
using asterisk 1.8, compiled res_fax.so and res_fax_spandsp.so - both
loaded successfuly
in sip.conf set t38pt_udptl=yes
but faxes still don't work even in passthru mode.
if i do a 'sip show channel' on the channel via which i am sending fax it shows:
T.38 supportYes
however
23 matches
Mail list logo
