Displaying 7 results from an estimated 7 matches for "vontag".
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montag
2005 Jun 06
1
CLUELESS NEWBIE needs help making an outboundsip call to PSTN
...asterisk/modules/pbx_wilcalu.so:
undefined
symbol: ast_p
thread_create
Jun 6 14:33:49 WARNING[27986]: loader.c:518 load_modules
: Loading module pbx_wilcalu.so failed!
asterisk will not run!
I have no idea what this means or how to deal with it. any help is much
appreciated!
Asterisk version: Vontage:/etc/asterisk# asterisk -V
Asterisk CVS-HEAD
umm.... not really informative there.... :-) I downloaded and built it
June, 6
2:45PM Eastern STD time (US)
Here's some more info about my system just in case it is userful:
Stable compiles & runs OK.
Thanks!
Steve
Vontage:/usr/src/aster...
2005 Oct 18
0
Re: Vontage Problems
I am a newbie and want to step up to VoIP and switch from analog
connetion to my Astrisk/Lineox box.
Any suggestions on configuring Vontage and what to get/ask
when signing up?
> Has anyone experienced problems with Vontage and Asterisk. I'm using
>Asterisk (Current Stable) and Sipura-841 phones. While talking on an
>outbound call the transmission seems to fade out until the other person
>can't hear me but I...
2006 Jun 19
2
home routers
...ng for somehting like the standard house hold linksys/dlink router.
Basically it needs to have at least 1x100mbit port, wireless G capabilitys
and at least 1 x anolog voip/sip connection. I've found linksys's WRT54GP2
which appears to do what i want. Anybody use this? Does it require
vontage's service? I'm looking for any recommendations.
Thanks
--
~Shaun
2006 Jan 30
1
Cant compile asterisk #error "You need newer libpri"
..._set_useruser'
discards qualifiers from pointer target type
chan_zap.c:2537: warning: passing arg 2 of `pri_call_set_useruser'
discards qualifiers from pointer target type
make[1]: *** [chan_zap.o] Error 1
make[1]: Leaving directory `/usr/src/zaptel/channels'
make: *** [subdirs] Error 1
Vontage:/usr/src/zaptel#
I have been careful to get rid of all existing asterisk stuff
executables modules and libraries etc. off thesystem before
trying to compile this.
The release compiles just fine but I st
ill get no audio with sip-sip calls if I compile the release version.
Thanks
Steve
2005 Jun 23
2
Asterisk 'losing' upstream provider registration state during small network outages.
...nsecure=very ; To allow registered hosts to call without re-authenticating
canreinvite = no
; BV claims they support rfc2833, but for some reason passing digits
; after a connected call only works with inband
dtmfmode = rfc2833
;dtmf=inband
CVS-HEAD
Running Version:
Asterisk CVS-HEAD built by root@Vontage on a i686 running Linux on 2005-06-06
22:32:05
*CLI> show version files
File Revision
---- --------
cdr_custom.c Revision: 1.11
cdr_manager.c Revision: 1.6
cdr_csv.c Revision: 1.16
pbx_functions.c Re...
2005 Jun 02
3
CLUELESS NEWBIE needs help making an outbound sip call to PSTN
...gives you the option to either #1 change my
outgoing unavailable message or #2 press ANY key besides #1 to hangup.
Any help or pointers would be GREATLY appreciated!!!!!
I compiled and Installed Asterisk about 10 days or so ago and am running
version:
CVS-Nv1-0-7-05/19/05-11:22:20 built by root@Vontage on a i686
running Linux
Thanks!!!
Having a BLAST!
Steve Gladden
2003 Dec 20
0
X101P + TDM400P
...pected in other ways. I wanted a home PBX system, that would let me
treat different callers different ways depending on CID.
I initially bought the Digium developer's kit to try things out. That's a
single port TDM400 and a X101P. I've added another X101P.
One X101P terminates in a Vontage Cisco ATA-186. The other terminates
with Qwest. The TDM400 is connected to both a TDD and a cordless phone.
I also have a softphone connected along with 2 DID numbers through
Voicepulse. I have a second Asterisk system outside my firewall to use
for FWD.
What went bad (most are minor):
- SIP...