search for: vontag

Displaying 7 results from an estimated 7 matches for "vontag".

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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...