Displaying 20 results from an estimated 300 matches similar to: "Problems Configuring Cisco 12SP+"
2005 Feb 06
1
no caller ID presented from 12SP+
Hi folks,
I have got a Cisco 12SP+ working (thanks Derek!) but I'm having a minor
issue with it. When I use it to call another desk I get no CallerID. The
receiving phone diplsays "asterisk" as the CID. Below is the skinny.conf
stanza.
[2207]
device=SEP00308062B006
version=P002L2J2
context=intern
callerid="Luke's Room" <2207>
mailbox=2207
transfer=1
2005 Sep 13
1
wctdm, issue w/outbound calls
Hi all,
I've been running Asterisk with a TDM400P for about 6months, no problems.
2 in/outgoing analog lines, one analog phone. Recently I was messing with
the XTEN client, got to finagling with things, and not knowing what was
wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was
testing various things, and found everything worked except outgoing calls.
So I checked
2005 Jan 04
1
Displaying incoming e.164 callers number - how?
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
phone number on my phone. In the log is the following - which displayed
'601' on my phone. The caller was +886288097680 - am I getting the wrong
ClID because of my end or the caller end?
2007 Jan 30
3
musiconhold restarts for every extension
Hello!
I've upgraded from 1.2.9 to 1.2.14 recently but experience an
unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was
playing continuous on sequential extensions after a
timeout, it is restarted for every extension in 1.2.14:
;music starts
exten => 902,1,Dial(SIP/phone1@proxy.com|5|m(mymusic))
;music starts again
exten =>
2003 Oct 13
1
AGI solution to Grandstream BT102 call waiting problem
I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification.
PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)
2005 Mar 16
1
cisco 12sp+/30vip IP phone
I was able to get Asterisk working with the demo on FreeBSD 5.3 without crashing, but not the music on hold, so I just have that disabled for now, but I'm ready to get some IP hardware working.
So I picked up a Cisco 12sp+ IP phone (mistake?) and am having difficulty finding any truly helpful instructions / troubleshooting to get this configured to work with asterisk. If I could just get
2005 Aug 09
1
Incoming call #2 sent to VM immediately when already on phone with incoming.
I'm having this problem where if the phone is ringing from
IncomingCall #1, IC#2 will be immediately sent to VM. Is there
somethign wrong with my dial plan? I currently have 4 incoming lines
going into a TDM400 with the group set to g0.
Could it be that the way I've set this up, if any of the phones are
busy, it goes immediately to VM?
exten => s,1,Answer()
exten => s,2,Wait(1)
2006 Mar 03
2
Does an entry in AstDB stay after reboot?
I set up a call forwarding script in extensions.conf which uses the
AstDB but I'm wondering if I reboot the server, will the entry in
AstDB still reside?
What the script does is when a call comes in, it check to see if there
is a null value or a call forward number. If null, it will call the
local office connections. If there is a number, it calls that. Now I
just need to know if I reboot
2004 May 12
1
Musical interruptions
Whilst on a call, I'm getting the following...
-- Started music on hold, class 'default', on SIP/phone3-a7d5
-- Playing 'pbx-transfer' (language 'en')
-- Unable to find extension '#' in context 'default'
-- Playing 'pbx-invalid' (language 'en')
ie - without anyone pushing keys - I hear the music on Hold - as does
the
2004 May 14
3
SoftPhone to SoftPhone with No Voice
Hello
I Installed Asterisk on RedHat 9. I am currently try to configure minimum with
two softphone talking to each other over the LAN. I am using X-Lite softphones
from xten.com site. I defined 3 phones in sip.conf and also specifies in
extensions.conf file. I am able to ring other computer but there is no voice
exchange ( i can't hear any think except ring). Here is the portion of sip.conf
2005 Jun 22
3
indexing tables for dialing
Hello
I would like to know how can I manage to implement a table which translates
an extension number into a phone number. Let see an example:
If I dial an extension like 3021, Asterisk has to Dial an agent (our
employees) located at San Francisco using the following telephone number:
415 541 XXXX. If it does not work we can also use his/her mobile number.
We need to manage more than 180
2005 Nov 21
2
Halflife / Counterstrike (using Steam) does not work.
Hi,
I'm using Gentoo Linux and Wine 0.9.1.
I had no problem to install Steam, but Halflife and it's Mods like CS doesn't
work. After starting the game through Steam, the background image appears,
but there are no menu entrys and the App didn't reponse anymore. If I
minimize CS and maximize it again, the background didn't get repainted. When
having virtual desktop disabled
2005 Jul 16
2
beginners question about extension context
Hi, all
I have couple of SIP phones and they are in [from-sip] context.
I also have an IAX2 phone. I have put this one in [iax-user] context.
I want to make calls between SIP and IAX2 phones. If I put them all in same
context all is fine, however when they are in different contexts they will
not call each other and I will get message (in * CLI) that particular
extension does not exist in a
2005 Mar 07
1
Custom Development
Hey guys,
I'm looking for a programming or Development Team/Company to do some custom
coding for Asterisk. What we need is not exactly simple. In fact, I'm not
sure the extent of the coding as far as technical terms go at all.
Currently we have a "call center" with 4 phones. There will be a total of 8
people using the phones. Obviously, no more than 4 people will use
2003 Jun 26
0
Kphone not working with Asterisk?
I'm trying to get two linux machines with kphone-3.11 two communicate with
each other over asterisk. I have them configured correctly on asterisk to use
sip channels, but when I call from one phone to the other I don't any voice
communication between the phones. According to the phones I'm connected, but
according to asterisk, I get the following message:
-- Executing
2005 Aug 02
0
strange asterisk issue
I have the following asterisk configuration (sip.conf) :
[General]
externip=82.79.81.3
localnet=192.168.10.0
localmask=255.255.255.0
[Phone1]
type=friend
host=dynamic
nat=yes
qualify=yes
context=sip
callerid="Phone1" <1>
disallow=all
allow=gsm
[Phone2]
type=friend
host=dynamic
qualify=yes
context=sip
callerid="Phone2" <2>
disallow=all
allow=gsm
[Phone3]
2004 Jul 17
1
Using a group variable for a group ofextension to dial
That could be it. What I want to do is set a group of callers and have
the event cause the phone to ring them in order. I will tie it to my
IVR portion and thus I can make sure peole in sales get calls based on
our hierarchy in the office. So if I am reading your example right the
syntax is....
Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf)
Is that a valid way to cause
2004 Oct 06
1
Hello - Simple SIP configuration
I'm new in hire so Hello to everyone!
I'm beginner user of Asterisk CVS-HEAD-10/01/04-14:31:34 . I just installed
it on my Mandrake Linux 10.0 kernel 2.6.3-4mdk with sample configuration
(used make samples).
I would like to make phone connections between X-Lite (SIP) installed on
computers in LAN. How to make this? I was reading manual, and tried to make
changes in sip.conf but this all
2006 Nov 27
2
SIP group management
Hi
can i set up a group of SIP users and forward a call to it?
I am looking for a group, not for a queue.
I won't listen any musinc on hold, and i won't that someone has to pay
if nobody of the user's in the group accept the call.
Can i do that?
Thanks to all