Displaying 20 results from an estimated 10000 matches similar to: "Fax detect"
2004 Nov 29
1
Outbound E&M?
I've got a new setup (different building) where Asterisk is sitting
between the PBX and phone company on a E&M T1 line.
Mitel PBX <-> Asterisk <-> Phone company
Inbound works. Asterisk gets the in-band digits from the phone company
and hands the call off to the Mitel just fine.
Outbound is weird. Asterisk seems to expect that the mitel will send
routing information
2003 Dec 16
1
DISA - Zap/DTMF Problem
Hi guys,
I am trying to use DISA. The scenario is - I call my home number (where
X100P seats) from mobile phone, enter the password, enter international
number and get connected via voiptel. It works perfectly when I call
extension setup with DISA from X-PRO SIP phone, but when I dial into
Zap, It seems that it does not detect DTMF tones. Here is a log and
config files
Please help
2003 Oct 06
1
chan_zap.c - echo cancelation getting in the way of dialing????
It seems consistant after dialing dozens of times that the call that
doesn't go through is the one the gets the log message "No
echocancellation requested" (chan_zap.c) and the "Scheduleing timer"
(channel.c) in the middle of receiving the DTMF tones.
I'm now using the T400P card last week very simular problems the the T100P
(although I think I was actually loosing
2005 Jun 01
1
rxfax problems - cont.
Well, my faxes passes through asterisk successfully, however I still have
some problems about fax reception by rxfax.
The softfax answers, and negotiates transmission, however then as some stage
of communiation something is wrong.
But I have nothing more but this log:
Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on
Zap/10-1
Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242
2003 Dec 17
5
ALL incoming Zap channel calls are getting picked up as FAX calls!
All,
I upgraded my asterisk setup from CVS on or about 12/15. Suddenly, *all*
of my incoming calls are coming up as FAXes. I had to disable my fax
extension because every call to my POTS line was getting redirected to my
FAX machine. After removing the FAX extension, if I call my POTS line from
my cell phone, I get the following:
*CLI> -- Starting simple switch on 'Zap/1-1'
2004 Jan 15
4
People detected as fax machines
A caller to me was this afternoon detected as a fax machine:
Jan 15 15:31:17 NOTICE[41997]: File chan_zap.c, Line 3564 (zt_read): Fax
detected, but no fax extension
... and then redirected to voicemail. An extract from extensions.conf is
attached below. Is there any way to stop * even considering an incoming
call on a line as a fax call?
Iain
bell]
include => mailboxes
include
2003 May 22
2
new DTMF tones
I just loaded from CVS this afternoon and in the debug output I see...
DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: m on Zap/16-1
DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: u on Zap/16-1
I knew about DTMF 0-9, A-D, *, and #, but I didn't know about m and u :-).
2004 Oct 05
1
difference between dtmf digit 8 and 9
Hello,
this is an example extensions.conf.
[default]
exten => 500,1,Answer
exten => 8,1,SetGlobalVar(firstdigit=8)
exten => 8,2,Goto(process,s,1)
exten => 9,1,SetGlobalVar(firstdigit=9)
exten => 9,2,Goto(process,s,1)
I call extension 500 and send dtmf digit 9. This is printed to the
CLI:
-- Executing Answer("Zap/20-1", "") in new stack
-- Accepting
2006 Feb 19
2
spandsp 0.0.2pre25
Hello,
Is anyone successfully using spandsp 0.0.2pre25 with either asterisk 1.0.x or
1.2.4? I've built a Gentoo ebuild for this version of spandsp and app_rtxfax,
and it builds, but I'm not having any luck getting it working. 99% of my test
faxes fail. Reverting to 0.0.2pre20 yields a much higher success rate.
I've bumped the console debugging level in logger.conf to include debug
2005 Mar 11
1
Incomplete incoming fax using spandsp 0.0.2pre10
Hi,
I have successfully compiled spandsp 0.0.2pre10 with * 1.05 which can accept
inbound fax calls. However, all fax received are incomplete (the first 10%
of an A4 page is fine, the remaining is either missing or garbled). I
suspect this is due to 'training error' (see below) which, according to
Steve Underwood's postings, cannot be resolved further. I wonder if it would
help to
2003 Nov 18
4
Help with Warnings
I'm trying to clean up some notices/warnings that are repeatedly logged
in *.Any Help would be appreciated as I'm not sure of the cause
/solution.
Here are the errors:
Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321
(zt_call): cidspill already exists??
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
/* Don't send audio while on hook, until the call
2003 Nov 25
2
zt_rec: Unknown error 500
I have a number of Zap/ extensions defined in a queue with ringall
strategy. When this queue is called sometimes Asterisk seems to think
that one of these channels is busy, while it is NOT. The following is
shown on the console:
--Called 44
-- Called 36
-- Called 41
-- Called 35
-- Called 38
-- Zap/44-1 is ringing
-- Zap/36-1 is ringing
-- Zap/41-1 is ringing
2004 Jan 08
3
Asterisk hanging?
Hi,
I compiled and am running the latest CVS but strange things are now happening..
it looks like asterisk is randomly declaring my calls to be fax calls,
complaining and then sending the calls into a black hole... if I hangup the
calls below (soft hangup) asterisk locks up and I have to kill the process.
NOTICE[21526]: File chan_zap.c, Line 3520 (zt_read): Fax detected, but no fax
2005 Jun 28
2
Trying to get *8 call pickup to work
I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When
I call from a zap channel or a SIP phone to another SIP phone, then dial
*8 from a third SIP phone, I get 503 Service Unavailable on the
third phone and I get this at the Asterisk console:
Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible...
Jun 28 09:01:24 NOTICE[16774]:
2003 Aug 25
1
chan_zap.c zt_rec: Unknown error 500
Hi all,
I'm using asterisk CVS-08/14/03-22 on a box with a digium T1 connected to a
channel bank and a digium E1 connected to the PSTN.
I get occasional warnings from asterisk:
WARNING[37909]: File chan_zap.c, Line 3197 (zt_read): zt_rec: Unknown error
500
This happens mosttimes in a loop like this:
[netland_helpdesk]
exten =>
2006 Feb 16
1
Problem making outbound calls on TE210P using NFAS
Hello,
I'm running Asterisk@home 2.5
asterisk 1.2.4
zapatel 1.2.2
libpri 1.2.2
on a Dell Poweredge 2850 (1 CPU) with a TE210P
I have 2 t1 circuits using NFAS with dchan on 24 and no backup dchan. I am able to receive inbound
calls on all channels and can only make outbound calls on channels 25-48.
Attempting to make an outbound call on channels 1-23 results in congestion.
2006 Feb 16
1
Update to the latest zaptel driver - Congestion gone, but scary write errors replaced it
Hi,
Yesterday I updated asterisk to the latest zaptel driver and today
my congestion problems are gone... (see
http://bugs.digium.com/view.php?id=6509), only to be replaced by:
Feb 17 10:02:37 DEBUG[19225] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 26
Feb 17 10:03:08 DEBUG[19274] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 216
Feb
2004 Nov 28
4
PRI Dialing failure?
So I reached the point where my PRI is accepting incoming calls, but I
cannot dialout. I must be doing something stupid, but I can't figure it
out. The Asterisk box is sitting between the Mitel and the phone company,
and has PRI lines to each. Asterisk was built from CVS r1-0
Log for a call from mitel heading outbound:
-------------------------
-- Accepting call from '' to
2006 Feb 08
3
PRI to PRI not passing callerid
I must be doing something stupid, but I can't figure it out.
I have three PRI lines connected to Asterisk, one from the phone
company, and two more connected to PBXs. Asterisk uses the incoming DID
information to decide which PBX to route the call to. Should be simple.
Asterisk is clearly getting the caller id info from the phone company:
-- Accepting call from '512345xxxx'
2006 May 01
6
Problems with zaptel and TE210P
Hello,
I'm just starting out with asterisk and I'm playing around with the
system. Currently I have a Digium TE210P connected to a PRI on the
Asterisk server. I have a SIP soft phone on my laptop for testing that
is working fine. When I try to place a call from my soft phone I get
this from Asterisk:
May 1 09:11:41 NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to
create