Displaying 20 results from an estimated 40000 matches similar to: "wget from within asterisk?"
2006 Feb 23
9
auto provision of IP501 polycom
Has anyone been able to get the IP501 to discover the FTP server IP
address (via dhcp or dns) and download 100% of the config from a
provisioning server?
We are still having to touch each unit to enter the ftp server address
and password, as well as set many of the options that will not take from
the config file.
Have a sample config file you are willing to share?
What is required in
2005 Jun 13
9
SIP Listen to multiple ports
Hello all
I'm trying to get my asterisk config to listen to multiple ports. This
is since some clients have port 5060 blocked by their ISP.
Does anyone know how to do this in sip.conf or if it is even supported?
Thanks!
2006 Apr 18
6
T1 to cross connect remote PBX and asterisk
Looking for someone with a successful experience similar to this;
I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk,
but over a long distance. We do not need any IP connectivity and the
solution requires G.711u audio so there is no benefit to using IP.
Has anyone here successfully cross connected any PBX PRI interface
expecting NI2 PRI signaling B8ZS/ESF with an
2006 Jan 27
3
OT?: International number parsing
Can anyone shed some light on "rules" that might make the task of
parsing the country code and city codes from a dialed number in the
CDRs?
I know that there is almost never a case where a concatenated country
and city code could overlap with another country code, but what about
city codes and local numbers? Is it possible for a concatenated city
code and local number to match another
2006 Jun 12
3
get value from DB directly
Hi,
I want to know how I can get a value from a table. Say, I have a
table sip_buddies for storing sip user account information. There is
a field called 'accountcode' that I want to get its value in the dial
plan. As I find that there is no direct way to get the value from the
table. Does anyone can tell me how can I get its value in the dial
plan?
Thanks!
2005 Aug 19
4
any ISDN/PRI signaling experts out there?
I have officially engaged in a pissing contest with the local Telco over
PRI calling name delivery.
The telco publishes their calling name delivery over PRI feature as
being bellcore gr-1367-core compliant.
The gr-1367-core spec states that the calling name is to be included as
a facility IE in the setup message, or sent in a subsequent facility IE
message with an indicator in the setup message
2006 Jan 25
20
* point to point t1 solution?
Can anyone point me to a reference or sample config for bypassing a
nailed up (point to point) t1 between two PBXs with asterisk and a pair
of t1 cards?
Right now I have 2 Nortel norstars connected to each other via a leased
line t1. I also have a solid 10mbps low latency microwave link between
the 2 sites.
My goal is to run an asterisk box at each end with a t1 card and
Ethernet card to
2005 Aug 02
3
priority "a" in macro to access voicemail
I have added the following to a macro that is used for all extensions so
a user can access voicemailmain by pressing * during the voicemail
prompt
; check voicemail
exten => a,1,voicemailmain(${macro_exten})
exten => a,2,hangup
The behavior is a little weird, the * key is not recognized during the
portion of the greeting where the extension number is being played back,
after it is
2006 May 01
12
CallerID Name problem
I'm having trouble getting callerid name to show up on my phones (Cisco
7960 and a few softphones)
When I look in the CDR database I see the name but not on any phone when
being called.
I'm running
Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
Any help would be great !
2006 Feb 02
9
Asterisk on laptop connected to POTS line
Anyone know of any equipment that I can use to connect
a laptop running asterisk to a POTS line (RJ11) ?
Regards,
Dovid
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2005 Aug 12
4
voicemail - 99 message limit
Anyone know how to override the 99 message limit in voicemail? (yeah, we
have a public VM that gets that many a day).
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2005 Aug 29
3
How to use * and # as part of number indialcommand
Michel
Send me the same output for a dial string that only sends the *31*
Is this an ISDN line? What type of card/signalling/switchtype are you
using?
It looks as if the PSTN switch accepts the *31* and then hangs up so you
can make the NEXT call with the *31* feature enabled. If so I assume the
*31* feature will be enabled for the next call on the ENTIRE SPAN if it
is an ISDN trunk group.
If
2006 May 17
3
Providers using Embedded Devices
Just curious...
Does anyone know if any companies using Asterisk on embedded hardware (out at the customer premisis), such as the Soekris Net4801, to provide VOIP service?
Doug.
2006 Jan 26
6
* point to point t1 solution? / alternatives
This has been an interesting discussion for me (except for the
sniping). The last post led me, out of curiosity, to this wiki entry:
http://www.voip-info.org/wiki-Asterisk+TDMoE
I was unaware of this feature, and it looks pretty good. I've been
pondering replacing some T1's by leveraging IP capacity but of course
have run up against the QoS issue. My idea was different...
I
2007 Jan 16
2
Polycom IP601 - some hints working, not others?
Are all of the sip phones in the same context?
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Robert Jenkins
> Sent: Tuesday, January 16, 2007 1:44 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] Polycom IP601 - some hints working,
2005 Sep 08
4
Solution for 12 to 16 FXO to asterisk connection
Hi, today a customer asked how to use asterisk with 12 to 16 FXO
ports. I can use a channel bank with 16 FXO ports and connect the
channel bank with a T1 cable to a T1 card in the Asterisk Server.
Asterisk will then send the calls to the Voip provider over the
internet.
However a 16 fxo port channel bank is about USD 1500 + a t1 card USD
500 + a USD 1000 computer = 3 thousand us dollars + my
2006 Jan 27
3
sip qualify=yes interval
In an earlier thread Andrew Kohlsmith enlightened me on the use of
qualify in sip.conf to deal with a peer that is down.
Since then I have been searching for information on how the behavior of
qualify can be tuned.
The wiki is vague on this;
" Syntax:
qualify=xxx|no|yes
where XXX is the number of milliseconds used. If yes the default timeout
is used, 2 seconds.
If you turn on
2006 Jan 30
2
RE: [Asterisk-Announce] Asterisk 1.2.4 and Zaptel 1.2.3
Does anyone know what date this memory leak was introduced and/or how to
check source code for it?
I am running a pre-1.2 CVS head version and would like to know if the
potential problem exists.
> -----Original Message-----
> From: asterisk-announce-bounces@lists.digium.com [mailto:asterisk-
> announce-bounces@lists.digium.com] On Behalf Of The Asterisk
Development
> Team
> Sent:
2007 Aug 07
2
turn off music on hold for a single sip user
Is there a clean way to disable music on hold for a specific user sip
user?
I have seen one example that creates a class called [none] that points
to an empty directory, which creates log errors that are annoying (but
not harmful?)
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2006 May 26
3
hint priority and realtime
Can someone shed some light on why the 'hint' feature was implemented in
the 'priority' field that is purely an integer in the rest of the
dialplan?
There seems to be a conflict with realtime and the hint priority, in
order to put in the hints you would have to change the priority column
in the database from int to char and give up some performance (since int
indexes better and