Displaying 20 results from an estimated 300 matches similar to: "911, channel full"
2011 Apr 02
1
Problem getting TDM400P clone card to go off-hook and dial
I am having problems getting a Nicherons TDM400P wildcard clone to dial out.
Everything appears to be configured correctly, but although I see call
progress, it never seems to actually pick up the phone.
(The following is a test of 911 emergency, where I substitute 811 [repair
service] as the actual number dialed.)
*CLI>
-- Executing [911 at from-internal:1]
2010 Aug 30
2
help with dialplan
Todd
How do you have the context in the phones sip configs set?
Bryant
From: "Todd Reese" treese65 at gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk
2005 Mar 17
3
Undocumented "exten" syntax?
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten => s,1,SetVar(SET_EMERG_FLAG=0)
exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten => s,n,SetGlobalVar(EMERGENCY=1)
exten => s,n,SetVar(SET_EMERG_FLAG=1)
exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} =
2006 Jan 10
1
busydetect
Hi,
I'm struggling to get busydetect to work.
I'm using asterisk 1.2.1 and a digium TDM04B (4 port FXO) card.
I've set busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf
and i've modified zondata.c with a busy setting of 620+480, 300/200 which is
the busysignal received from Korea Telecom.
Asterisk isn't detecting the busy signal and doesn't hangup.
2005 Jun 15
0
Asterisk slow transferring calls
Hi,
Running Asterisk CVS-Head latest on a Dual P3 800 1Gb Ram.
For some odd reason now that I have the asterisk box almost to the stage
I want it, I hit a problem.
I have a te405p in the system, Zap/g1 is connected to the telco as an
ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an Ericcson BP250
phone system.
When calls come in on g1 they go straight through instantaneously to the
2006 Feb 19
2
Line Dropouts on E405P
Hi,
We have a Ericsson BP250 Phone system setup witht he following configuration
Telco <-> Asterisk E405P <-> BP250
The system seem to work perfectly on 1.0.9 for a very long time but there is some functionality we wanted to take advantage of in the 1.2 version branch so we upgraded.
Currently running
Asterisk 1.2.4
Zaptel 1.2.3 (noticed 1.2.4 is out will upgrade next
2007 Aug 27
3
voip provider settings problem, please help
hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk
1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my
problem is the following one:
when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown
in the asterisk CLI and caller and callee can hear each other when
2007 Aug 28
1
calls being forwarded to neighbor?? please help, thx :)
hi ppl :D
my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2,
i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a
grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have
a machine (machine 1), which functions as my router and machine 2 and sip device are behind it,
grandstream box
2005 Mar 22
0
RE: Asterisk-Users Digest, Vol 8, Issue 150
The update worked like a charm! Hold music is as cheesy as ever!
Thanks much, this list is a life saver!
Dan
------------------------------
Message: 2
Date: Fri, 18 Mar 2005 09:16:59 -0600
From: Eric Wieling <eric@fnords.org>
Subject: Re: [Asterisk-Users] Redhat 9 Music on hold
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
2010 Jan 28
2
911, location
Hi there,
I am running a PBX under asterisk 1.6. I have few FXO analogue lines
connecting to PSTN. These lines are in a hunt group. I trying to make
my extensions to dial 91, but this is a bit scary, I mean if somebody
make an emergency call after hours and without completing call is not
able to tell his/her location. How can I make 911 call center to know
the exact location of my extension. I
2011 Feb 04
1
SoftHangup on asterisk 1.8.2.3
I am trying to use SoftHangup in my dialplan, but it's either not
working or I'm not using it correctly.
when i'm on the console, i see:
pbx1*CLI> core show channels
Channel Location State Application(Data)
SIP/vgw1-000000a2 2156181505 at inbound:1 Up AppDial((Outgoing Line))
SIP/143-0000009f s at macro-SaferSIPDial Up Dial(SIP/99302156181505 at vgw1,,
2 active
2005 Sep 23
0
Problem with outbound calls
Hi everybody,
I have some problems making calls from a sip user (HT286) to the pstn trough
Digium Wildcard TE110P, i allways have an error : SIP 403
INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd
From: "test" <sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79
To: <sip:0170708959@192.168.1.4;user=phone>
2009 Oct 15
4
PSTN to SIP line ratio
Hi,
I am planning to deploy an Asterisk PBX for 100-200 users. I am not
sure about PSTN incoming/outgoing line ratio for SIP users. I mean if
you recall dial up internet the common line ratio is 1:10 (one line
for 10 users on access server or an E1 for 300 users). Can somebody
tell me what is the good ratio for incoming and outgoing analogue/
digital PSTN lines.
Regards
Smir
2010 Jan 31
0
asterisk-users Digest, Vol 66, Issue 75
Hi Shahnawaz
Have you considered how you are going to address location issue for Mobile
users calling 911. You should think of SS7 MAP/TCAP to atleast know their
Cell ID
Regards
Sam
> Thanks very much everybody who contributed their thoughts. I would try
> to get some DID's so that each physical location can be identified by
> 911 call Center.
>
> Regards
>
> Shahnawaz
2010 Mar 30
2
Priority based softhangup
Hi,
Is it possible to softhangup a channel based on priority. I mean I
want to put some calls in higher priority lets say 100. If all
channels are busy and somebody wants to dial an extension with
priority higher than 100. How can softhangup drop a line which has
priority less than 100? I will appreciate your valuable help.
Thanks
Smir
2010 May 18
0
Location with PRI / Analog lines
Hi there,
I am stuck with the location issues. It would be easy if you have DID
for each extension so that outgoing caller id would be DID of the
respective extension and also physical address. Now if you are not
able to get DID's for some reason. I am thinking of some situations
and appreciate your thoughts.
1. If I have a PRI and map physical number (original numbers in hunt
group not the
2014 Mar 26
2
[Bug 911] New: The ICMPv6 type is param-problem or parameter-problem ?
https://bugzilla.netfilter.org/show_bug.cgi?id=911
Summary: The ICMPv6 type is param-problem or parameter-problem
?
Product: nftables
Version: unspecified
Platform: x86_64
OS/Version: Debian GNU/Linux
Status: NEW
Severity: normal
Priority: P5
Component: nft
AssignedTo: pablo at
2008 Jun 15
0
[Bug 911] Wishlist: Warn when unable to write .Xauthority
https://bugzilla.mindrot.org/show_bug.cgi?id=911
Damien Miller <djm at mindrot.org> changed:
What |Removed |Added
----------------------------------------------------------------------------
Status|NEW |RESOLVED
Resolution| |WORKSFORME
--- Comment #1 from Damien Miller <djm at
2003 Dec 14
0
CAMA MF signaling for a 911 Trunk
Hi all,
I'm trying to get a handoff between me and a carrier going using Asterisk.
I need to handoff using CAMA signaling. On a Cisco, you can see the
configuration types that I'm referring to on this site as an example:
http://www.cisco.com/en/US/products/hw/routers/ps221/prod_configuration_guide09186a008019b16e.html#35393
The example of the interaction is the following:
CLEC end
2003 Dec 19
1
911 settings.
I would like to know if anyone has come up with a script for 911 dialing
rules that put correct information on our locations. We have our office
in 3 different building one being our production & shipping dock. It is
almost 2 blocks away. We are connected with Ethernet Wireless between
the buildings and have Sip phones setup in the other 2 locations. All
the phones are working just fine.