Displaying 20 results from an estimated 44 matches for "ipbxs".
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pbxs
2006 Dec 18
2
Digium TE405P with French E1 => Red Alert
Hi
anyone have a idea for debug my digium TE405P card ?
My zaptel.conf:
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone = fr
defaultzone = fr
My Zapata.conf:
[channels]
language=fr
context=from-E1
switchtype = euroisdn
pridialplan = unknown
signalling = pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
2009 Nov 11
1
hosted / virtual IPBX platform
Hi,
I am looking for a hosted / virtual IPBX *PLATFORM* for service provider.Such hosted IPBX platform is aimed to be as a service, so that final customers don't have to install, maintain, or upgrade any PBX hardware or software.
It should have a control panel for end users to create / edit extension, conference rooms , IVR menus, etc
Could you please indicate companies that offer such
2008 Dec 22
1
Web-driven SIP call thru Asterisk IPBX
Hi,
I think that the web-driven SIP Phone (free) doddle (beta version) can be useful with your Asterisk applications.
You can pre-fill it with your sip settings (Asterisk host name or IP /?realm / sip user), you just need to setup the HTML link as that: (Attached is the HTML page example)
?
/**************************/
simple HTML code example:
/*************************/
<html>
<head>
2006 Jan 27
0
pb with callerid
Since I passed from version 1.0 to the 1.2.3. I have Pb with the
callerid. If somebody call with presentation of the number all is well.
If somebody make call in masked number, i couldn't send a callerid to
the phone.
It is in a call center and i use the callerid to present the name of the
number called to the operator.
Before that went. To identify the sda, I use the assignment of the
2011 Sep 02
0
No subject
typing his number, though there is a 15 seconds timeout, and even if I type
the number very fast it still may happen to me.
*same => n,Read(mobileNumber,app/input-mobile,10,,2,15)*
In the logs:
When it fails:
- - <SIP/ipbx-iwred-000002e> Playing 'app/input-mobile.slin' (language 'fr')
- - User disconnected
When it succeeds:
- - <SIP/ipbx-iwred-000002e> Playing
2004 Jul 02
3
Termination for Asterisk Users - Inter-Asterisk Exchange
Folks!
Netweb Group, Inc. fully supports connectivity to any Asterisk PBX systems you have and can provide A-Z termination with immediate effect.
Any volume is good enough for us, even an amount as small as $1.00 a day will do for us.
We will provide connectivity from our Softswitch IP 216.162.116.46.
If anyone is interested, add this to your Asterisk IPBX and then email me for setting up a
2009 Jan 05
1
B410p, Ast1.4, France Télecom Numeris Double T0 problem
Hi.
When I call my RNIS numbers (with a mobile phone for example), I can
see 2 incoming calls on the IPBX, which should not happend.
I'm not sure if it's a problem with the telco France Telecom and their
ISDN setup, or if it's a problem
with the MISDN driver on the IPBX itself.
I'm stuck ...
Any advices for troubleshooting that?
Someone provide working configuration files
2006 Nov 04
0
iax2 qualify - false "peer unreachable"
I would like to ask, if someone observe also problem with peer qualify
problems,
my asterisk log is full with UNREACHABLE/REACHABLE messages, even when
two asterisks are in LAN environment,
please take a look into this debug, I can't find any problem with packet
loss, all qualify requests are replied and acknowledged,
I will submit bug report, if you will also not find any problems here...
2015 Jan 19
2
SEMI-OFFTOPIC openvox
Hi list, I write on the list looking for help, buy a openvox gw gsm
for four channels and I'm a little disappointed with the support
openvox, for some reason , The call doesn?t get trough
support tells me it was my asterisk server, but does not really work
me and my internal calls are working perfectly, I tested with another
sangoma FXO gateway and works perfectly.
the problem is that
2015 May 21
1
asterisk 13 webrtc
hi,
is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ?
or is chan_pjsip better supported?
or the recommended way for asterisk is use respoke.io?
my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js)
chan_sip.c:10496 process_sdp: Can't provide secure audio requested in
SDP offer "
sip.conf (important parts)
[vr1a882]
...
nat=force_rport,comedia
2015 Jul 29
2
PJSIP T.38 issues
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Thanks for your reply Larry.
Le 27/07/2015 01:22, Larry Moore a ?crit :
> I think the "488 Not acceptable here" is occurring because the channel
> connecting through is not T.38 capable, that will be the IAX channel
> from iaxmomdem.
This is what T38gateway is supposed to do. And I'm very happy to report
that after one more
2004 Jan 16
1
ERROR[8192]
Hi all!
I get this error when trying to start asterisk:
ERROR[8192]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk
What can be the problem?
Thank you!
Miklos
iPFONE Telefonia IP
Rua Caio Graco 735 S?o Paulo SP
iPBX +55 11 3801-3702
UK +44 870 - 3403539
FWD 64662
sip:ipfone@sipserver.com.br
www.ipfone.com.br
info@ipfone.com.br
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2014 Apr 03
1
func_odbc
Hi All
Anyone know how to do include files with func_odbc.conf?
I now have several pages of functions in my func_odbc.conf and it is
getting harder to maintain it.
I would like to break them up into files by category. The standard method
of using the #include does not seem to work .
Ideas are appreciated.
Bryant
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2014 Jul 11
2
CDR(dst) not set in AEL macro
Hi
I'm using a macro to dial in a AEL dialplan. The problem is the macro do
not set the field CDR(dst), showing only ~~s~~.
I tried various configurations, but without solutions.
This is the macro:
macro dial-out(destno,dialstring,route_descr,interno) {
__TRANSFER_CONTEXT=ipbx;
if(${interno} = 1) {
Set(__PICKUPMARK=${destno});
if(${ODBC_verify_user(${CALLERID(num)})} > 0) {
t = tT;
}
2011 Mar 10
1
Is this true for Asterisk as SBC?
*Hi All,
I have starting to reading About SBC and found one artical reagding SBC and
they gives a solutions like this.
i want to know is this true in realtime sceanario while we think of an big
implementation and is it possible with cloud computing.
i have found from
http://www.smartvox.co.uk/products_gateways_explained.htm
Asterisk as a Session Border Controller*
Equip the Asterisk server
2008 Mar 21
1
Which command line is used to send emails to notify incoming voicemail ?
Hi,
In exim4, I can see lines such as :
mainlog.9:2008-03-12 08:53:28 1JZLmC-0000E7-0A <= root at foo.com U=root
P=local S=43802 id=Asterisk-0-123413860-4174-2662 at ipbx-bs-60200
In my voicemail.conf, I see :
; If you need to have an external program, i.e. /usr/bin/myapp called when a
;externnotify=/usr/bin/myapp
; If you need to have an external program, i.e. /usr/bin/myapp called when a
2003 Aug 04
14
Mysql CDR
hello all,
I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record.
Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault.
the original version of cdr_mysql.so works fine but I need the start time and end
2003 Jun 28
0
SV: Newbie questions.....
...card to this? If so, could
> we then connect the Asterisk PBX to the callmanager?
> (Perhaps with another extension range).....and if so, how?
Yes, it's possible. I've done it to trombone calls through a
Operator system (Trio) and to interconnect with both Ericsson,
Nortel PBXs and iPBXs.
On CM you set it up as a trunk line, create a route map which
forward all calls to that specific E1 / T1 port on the 6608,
which are connected to the Asterisk Pri port.
On the Asterisk you do the same. Beware that top-down, bottom-up
is opposite on the 2 system. Then you should be able to get it...
2005 Aug 05
0
Seeking Beta testers for enterprise mystery service
A company where I work is building an enterprise-grade infrastructure
system which enhances the usefulness of VoIP systems on the public
Internet. We're looking for a few enterprises which are running
Asterisk who would be interested in being Beta customers for our
pre-rollout testing. In exchange for being a Beta customer, you
would be entitled to free service through December 2005.
2008 Jan 04
1
Remote hold on PRI
Hi everybody
We have a strange problem with several asterisk servers (Version
1.4.11) using PRI cards (tied to telco here in Belgium).
Indeed we noticed that whenever a local user places an outgoing call
through the PRI (and telco) to another IPBX (tied to telco using BRI
or PRI), if the remote party places the call on hold, the caller hears
the _local_ music on hold instead of the