Displaying 20 results from an estimated 800 matches similar to: "Multiple Line Appearances / Why use this?"
2005 Sep 07
3
Extensions - Realtime
CVS HEAD/Asterisk 1.2: Is there a way to have the entire
extensions.conffile coming from the realtime?
It appears that RealTime for the extensions.conf file is on a context by
context basis, but you have to create each new context in the
extensions.conf file then add a "switch => Realtime" line (then reload). I
want to be able to add phones without having to edit any files.
2005 Sep 07
3
Hosted PBX (vPBX) and Call/PickUP Groups
Hi,
I'm working with this issue for a while, Now I already solve the
dialplan issues, but I still have a question about the Callgroups,
I read at www.voip-info.org that , there is a 63 limit of callgroups.
And I'm wondering why?? and if the 1.2.0beta version supported more than
63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any
unoficial patch for that ?
Thanks
2005 Sep 07
1
Several SIP clients behind router register with the same IP, messing up call routing, any ideas?
Dear all,
Has anyone seen this before and can suggest a solution?
I've got 3 sip clients behind the router, and they all register with
Asterisk using the same IP address.
Now, wenn all are registered, all the calls get routed to the client that
registered most recently, but not to the correct client.
Also, there seems to be some problems registering all 3 clients
simultaneously.
Could
2005 Sep 08
1
Additional: Several SIP clients behind router registerwiththe same IP, messing up call routing, any ideas?
Hello,
I'm still looking for any ideas on this problem:
I've got 3 sip clients behind the router, and they all register with
Asterisk using the same IP address.
Now, wenn all are registered, all the calls get routed to the client that
registered most recently, but not to the correct client.
Also, there seems to be some problems registering all 3 clients
simultaneously.
I have
2005 Sep 08
2
play each person's voicemail
How do I set each extension to play it's own voicemail prompts? I have vm
working in that it plays the standard "person at extension 1234 is not
available....." and takes the message. I've recorded seperate .gsm files for
each user but can not figure out how to use them.
- Gary
Edison Information Technologies www.EdisonInfo.com
P.O. Box 554
2005 Sep 08
2
Distinctive ringing on Cisco 79xx
Greetings, I am trying to implement distinctive ringing on a Cisco 7960.
I have tried setting alert_info to chirp1 or chirp2 before dialing the
phone, but it has no affect. If you have successfully implemented
distinctive ringing on a 7960, I would really appreciate seeing the snipit
of code that works.
Thanks in advance
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775)
2005 Sep 06
5
Good Polycom Dealer?
Could any of you provide me information on a good
Polycom phone dealers to utilize. One who provides
firmwares ..etc
Thank you!
Kenny
______________________________________________________
Click here to donate to the Hurricane Katrina relief effort.
http://store.yahoo.com/redcross-donate3/
2005 Sep 07
1
asterisk.org blocked - rejecting connections
Address lookup
canonical name asterisk.org.
aliases
addresses 216.27.40.102
Service scan
FTP - 21 Error: TimedOut
SMTP - 25 Error: ConnectionRefused
HTTP - 80 Error: ConnectionRefused
POP3 - 110 Error: TimedOut
NNTP - 119 Error: TimedOut
digium.com is ok though
Address lookup
canonical name digium.com.
aliases
addresses 216.207.245.1
Service scan
FTP - 21 Error: TimedOut
SMTP - 25
2005 Sep 08
1
(OT) Dialplan Standards for Business/Offices
Are there any standards for setting up pbx dialplans for businesses/offices?
What I mean is that, which numbers are reserved for a specific use ex. 0
for operator ? Putting Zero for operator in the dialplan seems to be the
common practice of businesses.
If there is such a standard, * and # are used for what ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2005 Sep 06
1
Some problems (SendDTMF, Wait, Parked Calls)
Hi all! I would like to solve some problems:
I have a sip provider that lets me make pstn calls after listening some
stuff and entering a pin number:
1) How can I make Asterisk enter the pin number? Then wait 1 second and
enter the phone number?
I have in extensions.conf:
exten => 6*,1,Dial,SIP/2002@myprovider,60,tr
I have tried with w (like with ZAP channels) but it does not work, nor
2005 Oct 04
1
Strange Problem
Hi
I am facing a strange problem. I have integrated speex
codec's narrowband mode in my SIP based server. Then I
tried to integrate the wideband mode. But the program
crashes mysteriously. My encode and decode codes for
wide band mode are exact similiar to that of
narrowband, except the mode initialization, where I
put "speex_wb_mode" instead of "speex_nb_mode".
My
2005 Sep 10
1
PRI echo
Hi,
My configuration is pri----*(te405p)---iaxclient.
My * version is 1.0.7 running on tyan dual opteron
board.
I have several problems.
1) inbound echo
For outbound call(iaxclient-->pri), there is almost no
echo. But for inbound(pri-->iaxclient), I can hear
distinct echo. Can Sangoma a104 or digium te406p help
this problem?
2)Today i received te406p. I know T1/E1 jumper. But
how can i
2005 Sep 07
1
Eeven Stranger - Asterisk BT100 Password Issue
Following on from my below email, things have taken another bizarre
twist..
I have continued getting the error when 2092 tries to listen to messages
by dialing 9999.
--Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack
--Playing 'vm-password' (language 'en')
WARNING: app_voicemail.c:4922 vm_authentication: Unable to read
password.
Then I
2005 Sep 15
0
Transfering from a device to a queue crashesAsterisk
Hi David,
I've got probably the same/a similar problem. Do you add the phones to the queue (AgentLogin/AddQueueMember)?
If there are entries like: " Spawn extension (macro-dialout-trunk,s,21) exited non-zero..." in your * log file you might have the same problem like me.
I suspect that something goes wrong with the nested macro calls within the AMP-generated dialplan, so what I
2005 Sep 29
0
Can't make outside call with SIP softphone
Hi,
I am can make local extension to and from SIP X-Lite
softphone, but I can't dial out using X-Lite but local
analog works just fine. Here are my conf files any
idea?
Thanks,
David
my sip.conf
[general]
bindport=5060 ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind
to (0.0.0.0 binds to all)
allow=all
[3000]
2005 Oct 05
3
IPComms Setup
Hey I just setup service with IPComms and they are
telling me to setup such as this:
iax.conf:
[IPCommsNet]
type=user
host=69.15.xxx.xx
context=voicepulse-in ;(changed by me)
nat=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=gsm
When I'm calling once of my numbers it's giving me
this though:
Oct 5 12:11:06 NOTICE[49584]: chan_iax2.c:5476
socket_read: Rejected connect
2005 Sep 29
14
Draggables and overflow div''s revisited
I have two scrollable div''s (overflow:auto), one with
a list of elements (the source) and the other is the
drop target (dest).
I''ve enabled ghosting so that the drag element gets
out of the scrollable box (good).
Interesting, at least on Firefox, the ghosted drag
ends up going ''under'' the destination div when I drag
it. No amount of z-order fidding seems to
2005 May 25
4
Polycom IP501
Hi All -
I noticed that the Polycom IP501's are now shipping. Has anyone
gotten one yet, and if so, what's different about the phone? Any UI
improvements, or is it just better hardware?
Thanks,
Noah
2005 Oct 05
2
Trillian?
Has anybody got Trillian to work?
If so, how?
Dex
--
-----BEGIN GEEK CODE BLOCK-----
Version: 3.12
GCS d--(+)@ s-:+ a- C+++(++++) UL+>++++ P+>++ L+++>++++ E-- W++ N o? K-
w--(---) !O M+ V- PS++(+) PE(-) Y+ PGP(-) t++(---)@ 5 X+(++) R+(++) tv--(+)@
b+(+++) DI+++ D G++ e* h>++ r%>* y?
------END GEEK CODE BLOCK------
http://www.againsttcpa.com - nothing fights like the opposition
2005 Sep 12
2
Callerid fails in any release after beta1 fails
I have 1 x100p. Caller id works fine with the beta1 release. Cvshead
releases fail with a combination of checksum and ss_thread errors?
I'm concerned when beta2 or the 1.2 release comes out it will not work.
I have been through the configs I can't find and changes that need to be
made to get CVSHEAD to work.
Thanks
John Hill