Re: [asterisk-users] conf files vs astdb
On Tue, May 11, 2010 at 04:48:30PM +0200, Harel Cohen wrote:
Hi all,
Could someone please tell me what is the relative cost in using conf files
oppose to the astdb? Basically I need to match a name to a phone number in
order to have all users registered by name and not by number (which I
understood is not a good practice). I have 2000 users and a complex dial-plan
and server resources become an issue.
I could implement this via a context in my extensions.conf:
exten = number,1,Dial(SIP/name) ; obviously I would need to hard-code
this for every extension
- or I could do it via astdb:
exten = _XXX,1,Dial(SIP/${DB(Names/${EXTEN})})
Which method would consume fewer resources (put aside other pro's con's)?
Is there any better way of implementing this?
Would 'hints' help me out here? If yes, I would appreciate a detailed
explanation how to use it.
How often do you update? With configuration files access is cheaper but
updating is more expensive.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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[asterisk-users] conf files vs astdb
Hi all,
Could someone please tell me what is the relative cost in using conf files
oppose to the astdb? Basically I need to match a name to a phone number in
order to have all users registered by name and not by number (which I
understood is not a good practice). I have 2000 users and a complex dial-plan
and server resources become an issue.
I could implement this via a context in my extensions.conf:
exten = number,1,Dial(SIP/name) ; obviously I would need to hard-code this
for every extension
- or I could do it via astdb:
exten = _XXX,1,Dial(SIP/${DB(Names/${EXTEN})})
Which method would consume fewer resources (put aside other pro's con's)?
Is there any better way of implementing this?
Would 'hints' help me out here? If yes, I would appreciate a detailed
explanation how to use it.
Thanks in advance,
Harel
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[asterisk-users] conf files for dahdi
upgrading from zaptel to dahdi, with a TDM400P: Is /etc/dahdi/system.conf the same as /etc/zaptel.conf? As I read the system.conf.sample, no echo canceller need be specified if there's a hardware ec. Can I just rename zaptel.conf? What about zapata.conf? Is this just renamed /etc/asterisk/chan_dahdi.conf? Or zapata-channels.conf? Or just left alone? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *.conf files not parsing
Has anyone ever seen Asterisk fail to parse files referenced by an #include by a *.conf command? e.g.: #include /etc/asterisk/sip-phones.d/*.conf Where the dir sip-phones.d contains sip extension conf files. This worked fine for nearly a month and then mysteriously stopped working for me last night! Regards, /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *.conf files not parsing
On Feb 11, 2005, at 3:42 PM, Tzafrir Cohen wrote: On Fri, Feb 11, 2005 at 09:35:12AM -0800, Robert Goodyear wrote: Has anyone ever seen Asterisk fail to parse files referenced by an #include by a *.conf command? e.g.: Two options: a. A number of versions have missed the inclusion of that patch. Latest version should work (asterisk 1.0.5-1.7) . b. If you use both xorcom and standard debian testing package, you may be using their package. In any case, what version exactly is your asterisk package? dpkg -l asterisk I deleted all my SRC files, grabbed latest from CVS and it seems to work now. Here's what asterisk -V tells me: Asterisk CVS-HEAD-02/11/05-13:14:58 Thanks! /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conf files
Hi All,
In my extensions.conf I have : -
exten = _6XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _6XXX,2,Playback(remote_unavail)
exten = _6XXX,3,Hangup
;
exten = _7XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _7XXX,2,Playback(remote_unavail)
exten = _7XXX,3,Hangup
;
exten = _81XX,1,Playback(transfer,skip)
exten = _81XX,2,Dial(IAX2/[EMAIL PROTECTED],50,t)
exten = _81XX,3,voicemail2(u${EXTEN})
exten = _81XX,5,Goto(s|6)
exten = _81XX,103,voicemail2(b${EXTEN})
exten = _81XX,104,Goto(5)
;
exten = _80XX,1,Playback(transfer,skip)
exten = _80XX,2,Dial(SIP/${EXTEN},20,t)
exten = _80XX,3,voicemail2(u${EXTEN})
exten = _80XX,5,Goto(s,6)
exten = _80XX,103,voicemail2(b${EXTEN})
exten = _80XX,104,Goto(5)
Which is great for setting up banks of extensions.
Question Is there any way to set a range of SIP, IAX and VOICEMAIL
extensions up in the coresponding .conf files instead of: -
SIP.CONF
[8000] ; SIP Phone
type=friend
insecure=yes
host=dynamic
reinvite=no
canreinvite=no
nat=1
mailbox=8002
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
allow=speex
allow=lpc10
;
[8001] ; SIP Phone
type=friend
insecure=yes
host=dynamic
reinvite=no
canreinvite=no
nat=1
mailbox=8003
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
allow=speex
allow=lpc10
IAX.CONF
[8100]
type=friend
host=dynamic
secret=
disallow=all
allow=gsm
context=default
;
[8101]
type=friend
host=dynamic
secret=
disallow=all
allow=gsm
context=default
;
[8102]
type=friend
host=dynamic
secret=
disallow=all
allow=gsm
context=default
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