Displaying 20 results from an estimated 11000 matches similar to: "international calls and NOANSWER"
2004 Dec 03
2
DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL)
Just throwing this out here, hopefully someone can tell me why.
*CLI> show version
Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running
Linux
Zap/g1 is pri_cpe to Bell Canada
5551234 is a normal POTS line I have busied out (handset offhook)
exten => 1234,1,Dial(Zap/g1/5551234,,g)
exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
2004 Nov 30
0
No voice when I dial out
I can dial from 601 to a public number.
The public number rings. I pickup and hear nothing, while on 601 it keeps ringing.
(BTW, is it right to say "ringing" on the active phone?)
The *CLI> doesn't show me anything useful:
Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack
Executing SetGlobalVar("SIP/601-8238",
2004 Dec 01
0
extension and PSTN connection
I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B
(603, 604). I have two lines on the TDM22B.
I cannot figure out some of the problems:
1. 601 dials via ZAP/3-1 to local phone number at PSTN:
ringing
pickup on PSTN (empty)
still ringing in the phone set 601
2. call from PSTN back:
601 picks up ... everything works !!!
No caller id shows up
3. For testing I have only one
2005 Feb 12
0
Outbound calls on a busy Zap/1: BUSY vs. CHANUNAVAIL
Hi,
I'm using the macro below in extensions.conf for most of my outbound
calls. One issue with my current configuration is that when I make an
outbound call it doesn't properly detect that my PSTN line (Zap/1) is
busy with another call and then overflow to my outbound IAX
connections. I think the root cause is that DIALSTATUS gets reported
as BUSY. The debug output is below. My desired
2005 Jun 22
1
missing cdr records
I am experiencing a very wired problem.
Some of my cdr are lost.
I use logging cdr to csv, mysql and odbc. But some of them are lost. They miss in csv mysql and odbc, so i'm pretty sure it is related to asterisk functioning.
I am running asterisk 1.0.7; this is simple configuration file:
extensions.conf
[general]
static=yes
writeprotect=no
[macro-gw-voipjet]
exten =>
2006 Mar 06
0
No ring when doing blind transfer.
Hi,
I have an odd problem when doing a blind transfer. The transfer is
intiated and the transferred caller hears nothing until the timeout. I
have tried setting the 'r' and the 'm' variables in the dial command.
Nothing happens when I use the 'r' variable when I use the 'm' variable
I briefly hear music on hold and then it stops until the timeout for no
answer
2006 Mar 30
0
BUG: FOP reports incorrect (duplicate) IP address until restarted
Hi,
This problem may be related to a configuration problem but I believe it
is a bug in the FOP since restarting the FOP server clears the problem.
Here is the scenario: Using AgentCallBackLogin and have four agents
logged in a call is made to one of the agents directly from an internal
phone. Okay so far. Call is hung up and the same extension is used to
call another agent okay again, no
2005 Sep 14
0
Dial Application Return Codes - Help needed
Hi.
I'm dialling two numbers - one that's unobtainable, one that's busy.
${DIALSTATUS} is coming back ANSWER each time right before the channels hang
up.
Am using the following dialplan macro to dial out.
[macro-advdial]
exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
2005 Sep 15
3
${DIALSTATUS} problems
Hi.
I'm dialling two numbers - one that's unobtainable, one that's busy.
${DIALSTATUS} is coming back ANSWER each time right before the channels hang
up.
Am using the following dialplan macro to dial out.
[macro-advdial]
exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
2008 Apr 29
0
PRI CallerID - leading zero added
Hello List!
We have problems setting the right caller id on outgoing calls. The
Asterisk Pbx is located
in Bucarest(Romania), our Telco provider is rcs-rds.ro. We have the
local telefon number
40787 00-99, associated to our PRI E1 Line. Where 00-99 are the DID
numbers available.
The telco is aspecting a 3 digit long Callerid from us, for example
like "710", for the extension 10.
2005 Jun 09
2
Sixtel is still alive?
Whoa, talk about flying under the radar. I got a few DIDs *months* ago
from Sixtel (or iax.cc). Initial respone was great, but then it seemed
that the only tech-support person had fled the country. No responses,
bad responses, poor call-quality. I had a few $s left in the balance
and kind-a just forgot about them. Not worth 10 minutes of my time to
get $10 of my money back, quite frankly.
2006 Dec 07
1
AMI - Originate Action and Busy, NoAnswer calls - CDR
Gang,
I'm wondering if anyone has run into this problem and found a solution.
When I use the manager interface to generate a call, I don't get very
much information in my CDR records when the dial status is BUSY, FAILED,
NOANSWER, etc. I am putting the dialed number into the CDR Userfield in
my dialplan, but the field doesn't populate the CDR record unless the
Originate action is
2007 May 22
0
Dialplan Problem - Outgoing
Hi,
I have some really disturbing problems with Asterisk 1.4.1 and my dialplan for
outgoing calls. First of all i switched some weeks ago from * 1.2 (bristuffed
version ) to this version and in my opinion a lot more troubles arose....
For outgoing calls I use a Digium B410P with chan_misdn (before a Junghanns
QuadBRI with zap).
1) So first thing is, that a user reports to me (highly
2008 Jan 26
1
CHANUNAVAIL
I've got a setup where we have 100 DID's. Our default dialplan has one
line that calls a macro:
exten => _22XX,1,Macro(STDEXT,${EXTEN})
The macro is pretty basic:
[macro-STDEXT]
exten => s,1,NoOp
exten => s,2,Dial(SIP/${ARG1},15,Tt)
exten => s,3,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${ARG1}|u)
exten => s-NOANSWER,n,Hangup
exten =>
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
2014 Oct 23
1
logger.conf
with the below defined in logger.conf on 11.6 cert 6
I am not getting any log message other than notice and warning in any files
when doing module reload logger - queue log is the only one that says it
restarts
*CLI> module reload logger
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Queue Logger restarted
built fresh box with make samples - added 2 stations, dialing from
2009 Feb 12
4
Asterisk Queue and URL Calling
Dear All
I want to integrate sugarcrm and asterisk , so when customer call the call
center the agent or extension which answers the call , before pickup the
phone and talk to customer , view his/her information if it is available.
I do this as described below :
1-Setup login username for sugarcrm for each extension
2-Extension Users will login to the sugarcrm.
3-Develop php script to handle new
2006 Jan 20
1
instant fallback to zap in case of sip/iax/xyz-failure
i would like to carry some oversea pstn-destinations via sip to providers
like stanaphone, however, in case of a network-failure or if the provider
is not available, i want to fallback to the zap-channels so the call is
carried out to the pstn directly.
the usual approach would be to check the dialstatus(e.g.NOANSWER).
however, asterisk tries >60seconds to reach that peer(even when the ip
2008 Mar 19
3
How to configure Voice mail for multi users.
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,.......and these user can call to each
other. Now i want to configure separate voice mail box for separate
user.
my extensions.conf ..... settings below..
[voicemail]
exten => _X.,1,Dial(SIP/${EXTEN})
exten =>
2008 Mar 19
0
How configure Voice mail for multi users.
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,.......and these user can call to each
other. Now i want to configure separate voice mail box for separate
user.
my extensions.conf ..... settings below..
[voicemail]
exten => _X.,1,Dial(SIP/${EXTEN})
exten =>