Displaying 20 results from an estimated 300 matches similar to: "Compiling SVN Trunk"
2006 Apr 01
2
TO have ringing tone instead MOH
I need to avoid MOH on my asterisk box, so i need to have a ringing tone
when attendant transfer is made, or a call is on hold..
Is there any way to do that.
I did not see a simple way to do that.
Regards
2006 Nov 17
5
spc.exe
Does anyone have a copy of spc.exe they could send me?
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2006 Jun 14
1
SPA-941 Disable call waiting or Disable Call waiting via asterisk
I'm trying to disable call waiting for Linksys SPA-941, but
unfortunately as far as I have seen, there are no parameters on the web
interface regarding this feature. I just want callers to hear the busy
tone when the called party is at the phone. Probably I can accomplish
this by using the "disable call waiting" in asterisk as well, but I have
not been able to find any
2007 Apr 12
8
test
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<TITLE>Message</TITLE>
<META content="MSHTML 6.00.2900.3059" name=GENERATOR></HEAD>
<BODY>
<DIV> </DIV></BODY></HTML>
2003 Nov 01
13
Quick Question
Apologies if there is a cleanly written and searchable FAQ that I could be
directed to. I have no problem to RTFM if I can find the FM...
Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R
servers that do not support X windows under RH8.x and I prefer not to go
back to RH7.3...
BTW, where would I find a useful FM?
David
--
David J. Sussman, MBA
email:
2006 Apr 10
5
SPA-941/942 Bulk provisioning
Has anyone got any information on bulk provisioning of Linksys SPA-941/94s?
There is an overview in the admin guide but it refers to a different
provisioning guide that I haven't found anywhere.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - <mailto:kerryg@techdatapros.com>
kerryg@techdatapros.com
2006 Mar 29
3
SMS in Spain (it seems Protocol 2)
Hello,
(I have asked it some time ago in Asterisk-es mailing list, so excuse me if
anybody receive it twice.)
I am trying to send SMS in Spain using landlines. It seems that
app_sms.c only handles Protocol 1, but Spain and Italy are using
Protocol 2.
I have been searching in Internet without any results... anybody is
sending SMS from Asterisk (or any method) using Protocol 2? (so, it
seems,
2006 Apr 21
5
Separating Asterisk SIP extensions from dialing each other.
This is coming from an * noob. :)
I've got two customers, they both are replacing their phone systems with
VOIP, and we need to retain both their existing dialplans.
One has 5 extensions starting at 100, and the other has 10 extensions,
starting at 100.
Is there a way to have the same extension number twice in the same
asterisk system ?
They will have different incoming DIDs of course.
2006 Jan 12
2
Easy to Access Telephone Directory AGI
I've written myself a easy to use telephone directory
which I use at home and thought it may be of interrest
to others.
The purpose of this agi script is to provide an online
telephone directory that can be easily accessed using
the numbers on the phone dial pad.
You select entries by spelling out the name of the
person you want to contact using the phone dial pad.
Now this is normally
2007 May 23
16
WiFi SIP phones
Greetings list,
What are people's experiences with WiFi SIP phones?
When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices.
I assume things must have moved on somewhat since
2006 Apr 23
0
RE: Asterisk-Users Digest, Vol 21, Issue 130
Have you thought about making them agents, they would both be reachable by dialing there agent number then, and I know only one agent can be logged in at once. Just a thought.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com
Sent: Saturday, April 22, 2006 8:26 PM
To:
2006 Mar 27
0
Question about Polycom 601 and expansion module.
Hi, I have questions about the Polycom 601 and side card....
1) In the side card the lights all time off... But all functions it's ok.
I need help with extension module of polycom... All works fine... But lights
not work.... So... I don't know when any person or extension is busy...
Any ideas?
,
Olger
On 3/27/06 11:34 PM, "asterisk-users-request@lists.digium.com"
2004 Jun 29
5
nat problem
hello,
i have trouble with nat + sip outgoing call.when make an outgoing call to a
sip gateway, i have no sound.
i have 2 sip gateway, one is asterisk.
asterisk is on public ip and private ip
other sip gateway is on public ip
phone are cisco and grandstream on private ip on the same subnet as
asterisk.
phone are connected by sip to asterisk (i have try with or without nat=yes)
incoming call
2005 May 16
11
H323 to SIP
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2006 May 05
1
problem g729
Hello ,
I'm have this problem before copy codec in the /usr/lib/asterisk/modules
before registration ...
My asterisk is Asterisk SVN-trunk-r20297 built by xxx@ xxx on a i686 running
Linux on 2006-04-20 01:02:07 UTC
This erro :
codec_g729a.so]May 5 21:39:16 WARNING [6950]: loader.c:731 __load_resource:
misstng mod_data for codec_g729a.so
Segmentation fault
Thanks
2006 May 28
1
FreeBSD Digium g.729 codec seg faults on rev 30652
Greetings-
Was running the Digium FreeBSD g.729 codec until recently when the latest
Asterisk bits were obtained via svn:
svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
This produced:
Checked out revision 30652
This on FreeBSD 6.1-RELEASE
Attempting to start asterisk it returns:
== Registered custom function URIENCODE
[codec_g729a.so]May 27 13:29:59 WARNING[71884]:
2011 Aug 27
4
to represent color range on plot segment
Dear R community,
With an advantage of being "NEW" to R, I would like to post a very basic
query here,
I am in need of representing gene expression data which ranges from -0.09 to
+4, on plot "segment". please find below the data df, the expression values
are in df[,2]. kindly help me with the code, so that I can represent the
values with a clear color gradient (something
2006 Jan 21
3
cvs asterisk compile failed (newer libpri)
I used:
cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds
iaxyprov astcc
and in the same order I try to compile it.
Asterisk ends with the lines below. It complains of a newer libpri, but
I just did it a step before!
What do I miss?
chan_zap.c:62:2: #error "You need newer libpri"
chan_zap.c:128: error: parse error before '<<' token
chan_zap.c:133:1:
2005 Sep 14
11
RxFax/TxFax - Compile Problem
Anyone know how to fix this?
gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff
In file included from app_rxfax.c:14:
/usr/include/asterisk/lock.h: In function `ast_mutex_init':
/usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE'
undeclared (first use in this function)
/usr/include/asterisk/lock.h:302: error: (Each undeclared identifier is
reported
2017 Jun 18
2
asterisk 13.16. - sigseg during negotiation
Hello!
unchanged asterisk crashes during udptl / t.38 negotiation with telekom
- they do not support t.38 / udptl.
In detail:
fax client -> asterisk -> telekom -> easybell -> asterisk -> fax server
Fax server sends t.38 reinvite via asterisk to easybell.
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 2447581897 4 IN IP4 46.17.15.23