Displaying 20 results from an estimated 58 matches for "nhadie".
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nadie
2008 Jul 01
3
music on hold realtime
Hi,
Is it possible to use realtime for Music On Hold?
Is it also possible to store the music/audio files on the database, same
way a voicemail can be stored on the database?
Thank You
Regards,
Nhadie
2008 Apr 23
2
prepaid on the trunks
...sion 101 and 102, customer B will have a different trunk 87659043 but will also have the extension 101 and 102.
i want to create a billing system to monitor only the trunks and also to load amounts on those trunks. is this possible? will i be able to use app_prepaid for this?
thank you.
regards,
nhadie
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2009 Jan 28
4
route based from source
...tbound i will make him use of
[sip-trunk-100]
another user, 101300 when that users calls outbound i will make him use
of [sip-trunk-101]
actually the 100 and 101 at the beginning of the username is the
accountcode i used for cdr.
hope my question was clear enough. thanks in advanced
regards,
nhadie
2009 Feb 18
3
US DID
Hi,
Anyone knows a DID provider that can do both outbound and inbound?
Regards
Nhadie
2008 Aug 11
1
Asterisk Realtime Unregister
...nything that would
not make my dialplan to wait for 30 secs.
also i'm not using rtcachefriends, how would i know in the CLI which
user is registered? i tried sip prune but it shows me nothing
sip prune realtime peer all
No peers found to prune.
anyone experienced this?
thank you
regards.
nhadie
2008 Aug 22
4
set callerid with plus sign
...y this is what i do CALLERID(num)=+6523450017
but telco is denying calls, coz they said they are seeing "bs523450017"
instead of +6523450017.
i tried putting it inside double quotes CALLERID(num)="+6523450017"
telco says the same thing.
is this possible? thank you
Regards,
nhadie
2008 Jun 11
2
time on asterisk
Hi,
I'm using gotoiftime on asterisk, but it seems there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob?
Regards,
nhadie
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2008 Oct 22
3
asterisk video
...erisk+phone+xten+eyeBeam
i can see that eyebeam is trying to broadcast a video but the other
eyebeam is not receiving it.
i tested the same setup but this time using ser with rtpproxy and
eyebeam video works fine.
any ideas? where do you think should i start troubleshooting this?
TIA
Regards
nhadie
2007 Aug 23
1
[Serusers] why combine ser with asterisk
...still
likely run into NAT issues that need to be dealt with somehow) and still
let you use Asterisk as an in-between point.
Together, Asterisk and SER make a very powerful combination for
providing a full suite of services to clientele, and each plays well off
the other's strengths.
N.
Nhadie wrote:
> Hi All,
>
> What's the advantage of combining ser with asterisk? I always see
> comments like using ser with asterisk is a very good solution etc. etc.
> the thing i liked with ser is that it does not do codec translation,
> which saves me cpu usage and also bandwid...
2007 Aug 16
2
Outbund Route via Extension
...nd route 1 extension that
starts with 4 goes to outbound route 2. Basically, i'm hosting two(2)
office, extension 3XXX is office 1 and extensions 4XX is office 2, they
both have the same dialling pattern so i need to choose route based on
source. i'm using freepbx for this.
thank you
Nhadie
2008 Nov 25
1
cdr mysql error
...ove the timeout if
there is any? thanks
[Nov 25 13:22:37] ERROR[21026]: cdr_addon_mysql.c:171 mysql_log:
cdr_mysql: Server has gone away. Attempting to reconnect.
[Nov 25 14:20:32] ERROR[21061]: cdr_addon_mysql.c:171 mysql_log:
cdr_mysql: Server has gone away. Attempting to reconnect.
regards,
nhadie
2009 Mar 24
1
Asterisk Originate Command
...7 extension 987654321 at outbound-route
What i'd like to be able to is instead of a local extensions i would
call an outside number then connect it another outside number. e.g.
originate SIP/85431210 at outbound-route extension 987654321 at outboudn-route
is this possible? thanks.
regards,
nhadie
2008 Oct 09
2
retransmitting NAT
...3] -- Called 103100
it stops there until it reached the timeout i set then it will say
unavailable.
is there a way that realtime will know that the phone is not registered
anymore? or what could be causing the retransmitting of NAT? has anyone
encountered the same prob? thank you
regards,
nhadie
2008 Jan 08
2
:POSSIBLE SPAM: conferencing help
Hi All,
kind of need help on the conference module, i'm using freepbx and
enabled conferencing, i created a conference number, 6000. when i dial
to it, my phone says it is connected but i'm hearing nothing, maybe logs
below can help you.
also, when i hang up the phone, the conference did not disconnect me.
how can i end a conference? thank you
-- Executing
2008 Jun 25
1
AS5400 E1 SS7
...k to send traffic to AS5400 connected to an SS7-PRI.? this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200?
TIA
Regards,
Nhadie
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2008 Jul 30
0
RES: GotoIftime
Hello Nhadie,
I had a very similar situation. My solution, even tough might not look very
wise, solved my problem the way I needed.
I repeated the GotoIftime command in the next line in my extensions.conf .
Like this:
GotoIfTime(22:00-23:59|*|30|jul?test,s,1)
GotoIfTime(00:00-02:00|*|31|jul?test,s,1)
Rgs,...
2008 Oct 08
1
registration limit
...mething whereby a user can assign his extension on an IP
phone in the office, and assign the same thing maybe to a softphone on
his laptop or maybe a sip client on a mobile phone. so that whenever he
leaves the office he can still be reach on his extension via the
sotphone. thank you.
regards,
Nhadie
2008 Oct 14
1
asterisk+heartbeat
...ill have the IP
10.10.10.2 eth0
10.10.10.3 secondary eth0
problem is i have to bind asterisk to the secondary IP if dont, i cant
make calls. but if server 2 is inactive, asterisk does not run, as on
the config it is binded on the secondary ip.
anyone uses heartbeat for failover? tia.
regards,
nhadie
2008 Jul 22
1
issue with high latency
....204.205: bytes=32 time=43ms TTL=56
Reply from 202.203.204.205: bytes=32 time=12ms TTL=56
Reply from 202.203.204.205: bytes=32 time=13ms TTL=56
even if the latency is high i still have internet access as i can still
browse and using yahoo messenger.
anyone encountered something similar?
regards,
nhadie
2008 Apr 30
2
Sending caller name out PRI?
I have a PRI connected to a traditional PBX using NI-2 and a typical
config (further below). When I call from a SIP/IAX phone to an extension
on the PBX, only the number makes it through. If I plug that same port on
the PBX to a carrier the PBX presents both name and number.
Hints or pokes to relevant chapters in documentation? My config is
essentially like one found here: