Displaying 20 results from an estimated 1000 matches similar to: "VoiceMailMain issue.."
2005 Jul 25
7
Some more VOICEMAILMAIN issue...
Hi everybody,
I have corrected this line in extensions.conf by stripping spaces off and now it executes:
exten => 22999,1,VoiceMailMain(s${CALLERIDNUM})
when it runs, the mail box number is asked and password too. I expected no question were made, because I inserted CALLERIDNUMBER and s in front of box number.
Anybody knows why?
Thank to you all, very kind members of this list!
Ciao
Mauro
2005 Aug 24
3
Issue in calling mobiles....
Hi dear group members,
I have finally an Asterisk box working, capable of receiving and making
calls. I have this issue while calling mobiles from our SIP softphones:
--------------------------
linux*CLI>
-- Executing NoOp("SIP/2000-6850", "3487024125") in new stack
-- Executing Dial("SIP/2000-6850", "ZAP/g1/3487024125") in new stack
-- Called
2006 Jan 17
1
Asterisk under SUSE 9.2/VMWARE 5.5.1
Hi everybody
I'm trying to make Asterisk 1.2.1 run under VMWARE and Suse 9.2.
I use ZTDUMMY module for timing and ZTTEST gets an average precision of 98,4 %.
Is there any way to improve it?
Best regards
Mauro Zanin
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2006 Jan 20
1
Connecting a TE to a NT BRI isdn
Hi everybody,
I'm strugling between two devices: the both TE but one was set up as a TN. I
have no current on that interface. I have tried to find some circuit over
the net to power the connection, both commercial and home made. Can anybody
give some hint?
Ciao
Mauro
2006 Nov 10
1
Need to automatically park an incoming call and then connect to an extension.
Hi everybody,
I have this issue:
I need to automatically park an incoming call, play a welcome prompt and
then connect to some extension but under extension user's command.
I was thinking to use a small database to comunicate between asterisk and
the main application.
Has anybody had this kind of experience?
Best regards
Mauro
2007 Jun 03
1
Loud noise instead of MOH
Hi Everybody,
I'm experiencing this kind of issue.
One ASTERISK 1.2.17 is connected to a Bristuffed HFC single ISDN channel
card. Everything seems to work but sometimes the third party caller when
listening to MOH listens some "SSHHHHH!" instead of MOH, this is not
continuos, MOH plays ok for, say, 20 seconds then the sound and then another
30 seconds of good MOH.
We have some
2007 Mar 24
1
Issue with Hamlet ISDN PCI card(Cologne Chipset)
Hi everybody
I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded
software.
I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD
in a normal Italian EUROISDN installation. The * works fine except for the
ISDN CARD. It is always Channel D down, but if a Call comes in, it works
perfectly for some time, both inbound and outbound. It prompts Channel D UP!
2005 Aug 26
3
Re:TE110P EuroISDN dial out timing out
Try different entry in this parameter.
In Italy mobiles start with 3, while public services with 1 and normal user
numbers with 0.
Using pridialplan=none, every number different from 0 was resulting in
termination code 1, normally used for number never seen on the network.
Ciao
Mauro
2006 Oct 18
1
Re: Is 1.2.12.1 production ready (Mauro Zanin)
Hi Everybody,
as far as I see, I installed 1.2.12.1 on a 1.2.0 box. Everything ran OK, but
VoiceMail application stated that there were no entries in voicemail.conf,
so it didn't work. Installed again 1.2.0 and voil? the VoiceMail app. was
working again. I asked to the group, but it seems I'm the only one with this
issue!
In Italy we say: "Chi lascia la via vecchia per la nuova, sa
2007 Apr 20
1
Why duoble digits must be so fast to activate features?
Hi everybody,
I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm
using Trixbox..).
I must be as fast as a flash to press *2 and do an attended transfer. If I
wait only a tenth of a second nothing happens.
I think it is an issue. I have seen the source code and found nothing bad.
Is this a known issue?
Many thanks
Best regards
Mauro
2005 Aug 05
1
Problem running Astrerisk after defined card in /etc/asterisk/zapatal.conf
I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start:
Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters
Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15'
Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1
== Unregistered
2005 Aug 03
0
Installing a TE100P (Digium) card over Suse 9.2..
Hi everybody,
I managed to install card over Suse 9.2, I substituted Zaptel drivers and compiled them. Now "ztcfg" says I have one card with correctly configured 31 channels, but red led on back of card doesn't flash. Suse 9.2 has detected the card as a Tiger Jet card, since the chip on it is a Tiger 320. The second card configuration is still waiting for configuration, but I think
2007 Sep 13
6
Number -> Fraction
Hi everybody!
I'm new to this list and also to the R program.
I'd like to know if there is a function able to convert results into
Fractional form like my scientific calculator have. For example:
> 1/3
[1] 0.3333333
> function_that_return_a_fraction_from_numbers(0.3333333)
[1] 1/3
Thanks
Mauro
--
Man, he is constantly growing
and when he is bound by a set
pattern of ideas
2017 Sep 18
6
how many hosts could be down in a 12x(4+2) distributed dispersed volume?
Dear All,
I just implemented a (6x(4+2)) DISTRIBUTED DISPERSED gluster (v.3.10) volume based on the following hardware:
- 3 gluster servers (each server with 2 CPU 10 cores, 64GB RAM, 12 hard disk SAS 12Gb/s, 10GbE storage network)
Now, we need to add 3 new servers with the same hardware configuration respecting the current volume topology.
If I'm right, we will obtain a DITRIBUTED
2018 Jan 02
2
"file changed as we read it" message during tar file creation on GlusterFS
Hi All,
any news about this issue?
Can I ignore this kind of error message or I have to do something to correct it?
Thank you in advance and sorry for my insistence.
Regards,
Mauro
> Il giorno 29 dic 2017, alle ore 11:45, Mauro Tridici <mauro.tridici at cmcc.it> ha scritto:
>
>
> Hi Nithya,
>
> thank you very much for your support and sorry for the late.
> Below
2017 Dec 29
2
"file changed as we read it" message during tar file creation on GlusterFS
Hi Mauro,
What version of Gluster are you running and what is your volume
configuration?
IIRC, this was seen because of mismatches in the ctime returned to the
client. I don't think there were issues with the files but I will leave it
to Ravi and Raghavendra to comment.
Regards,
Nithya
On 29 December 2017 at 04:10, Mauro Tridici <mauro.tridici at cmcc.it> wrote:
>
> Hi All,
2018 Jan 02
1
"file changed as we read it" message during tar file creation on GlusterFS
Hi Ravi,
thank you very much for your support and explanation.
If I understand, the ctime xlator feature is not present in the current gluster package but it will be in the future release, right?
Thank you again,
Mauro
> Il giorno 02 gen 2018, alle ore 12:53, Ravishankar N <ravishankar at redhat.com> ha scritto:
>
> I think it is safe to ignore it. The problem exists due to the
2018 Jan 02
0
"file changed as we read it" message during tar file creation on GlusterFS
I think it is safe to ignore it. The problem exists? due to the minor
difference in file time stamps in the backend bricks of the same sub
volume (for a given file) and during the course of tar, the timestamp
can be served from different bricks causing it to complain . The ctime
xlator[1] feature once ready should fix this issue by storing time
stamps as xattrs on the bricks. i.e. all bricks
2003 Jul 02
3
Rsync utilization every 5 seconds
Hi.
I installed the rsync and I need it to run every 5 seconds between 2 machines.
I was planning to use it in crontab, but this allows only every minute.
I couldn?t find how to do it.
Could someone explain me how to run rsync automatically every 5 seconds?
Thanks in advance,
Mauro
2009 Aug 26
4
Multiple user registration ...
Hello there!
We are planning to use Asterisk on our VoIP platform, and we are
spending some brains on a way to provide the following facility: let
some SIP user (extension) registrate with more than one client (ATA,
SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate
calls from any of this devices that are registrated with the same user -
no problems on tests too -,