Displaying 6 results from an estimated 6 matches for "callthem".
2009 Apr 22
2
Conference problem
Hello all,
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I have some issues with the MeetMe application.
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The working topology is as follows. The Asterisk (1.4.22-rc5) is connected through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are forwarded to Asterisk by the CM.
?
The problem is that some users who are calling in from PSTN are getting
2009 Jan 16
0
No subject
...dahdi show status
Description Alarms IRQ bpviol
CRC4
T2XXP (PCI) Card 0 Span 1 OK 0 0
0
T2XXP (PCI) Card 0 Span 2 RED 0 0 0
On Thu, Apr 2, 2009 at 9:40 PM, Martin <asterisklist at callthem.info> wrote:
> That's very strange ... the code when is compiling checks whether
> zaptel is present and then
> the #define HAVE_ZAPTEL is set.
>
> Since your error says No "ZAP" channel ...
>
> and the code says
>
> ast_log(LOG_WARNING, "No %s chan...
2009 Oct 20
3
troubleshooting NAT
Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at your install and they said we are having a NAT problem but didn'ttell me if it was with the asterisk conf or the Cisco ASA.
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2010 Jul 28
1
Why do Zaptel calls drop all of a sudden? Could busy detect be the problem?
Hi Guys,
I am getting a complain that call on analogue lines (Sangoam A400D) drops
all of a sudden. Here is what I see in logs:
[Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy,
avgtone: 75, avgsilence 135
[Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing
[h at macro-dialout-trunk:1] Macro("SIP/2111-b6a400b0", "hangupcall|") in new
stack
[Jul
2009 May 18
7
callcenter / dialer / predictive dialer / vicidial program is now open
This is a global message to all to announce our callcenter / dialer /
predictive dialer / vicidial program is now open.
Codecs: G711, GSM, G729, G723
Protocols: SIP
Duration Rate : 30/6 (6/6 with monthly minutes over 100,000)
Channels : 100 to start with , more on demand.
We are predictive dialer friendly , your account will not be shut off.
Contact us to do a test run.
Mike
2009 Apr 02
0
SIP topology hiding
Dear All,
Is anyone having luck with using some code for SIP network topology
hiding + NAT traversal (SBC functionality) with Asterisk ?
I tried OpenSBC but it didn't do NAT from Asterisk to ATA correctly.
It's in plans for OpenSIPS but it's not implemented yet ... checked
their svn.
Martin