Displaying 20 results from an estimated 300 matches similar to: "error receiving a fax ... but with a fax that was received without problems"
2015 Jun 25
2
Receiving faxes with spandsp question
Hello!
I?m trying to add fax functionality to my asterisk installation. Right now I?m focusing on receiving faxes. This is not explained in a book, but I assume that I can use same context, add ?fax? extension and if someone calls to send fax - it will autodetect. Right?
Per book, I made following setup additions:
1. In sip.conf [general] I added:
;FAX stuff
faxdetect=yes
t38pt_udptl=yes
2.
2014 Aug 11
2
Sending and receiving fax with Digium FFA
Hello.
I've been trying to setup Free Fax for Asterisk on a Debian machine with
Asterisk 1.8. I have managed to register and installed the Digium
modules. Sending and receiving through it have resulted in failure. The
output of fax show capabilities is:
Registered FAX Technology Modules:
Type : DIGIUM
Description : Digium FAX Driver
Capabilities : SEND
2010 Jul 12
1
Fax for Asterisk, capable of receiving from website but not from fax machine !!
Hi Guys,
i am using the latest version of asterisk 1.4 (1.4.33.1), dahdi (2.3.0.1)
and FFA (Applications: 1.4_1.2.0, Digium FAX Driver: 1.4_1.2.0). the issue
i'm having is that i'm able to receive faxes from a website (that offer this
service) but not able to receive from a regular fax machine (that is working
perfect).
[fax-rx]
exten => receive,1,NoOp(**** FAX RECEIVE ****) exten
2011 Feb 21
1
Dialplan execution stops on app call even with TryExec (Am I missing something simple?)
We're having an issue where we call ReceiveFax in a context that
includes a hangup extension and half the time dialplan execution doesn't
continue after the fax is received successfully. Am I missing something
simple here? Below is a sample call where this happened:
The last log line for this channel/call is:
[Feb 21 09:10:53] VERBOSE[13730] res_fax_digium.c: -- Channel
2011 Jun 29
1
dialplan execution stops after ReceiveFax
Hello,
I have a noticed strange behavior in Asterisk 1.6.18.2 with ReceiveFax
Digium FAX Driver: 1.6.2.0_1.3.0 (optimized for i686_32).
I use a context [capi-in] for icoming ISDN calls:
======
[capi-in]
; Faxe fuer Ruben
exten => 12345,1,Macro(faxin,ruben.roegels at jumping-frog.org,${EXTEN})
======
My macro for the fax receiving looks like that:
======
[macro-faxin]
; Faxe
; ARG1 =
2013 Jun 19
6
Mailing a fax with mutt does not succeed
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the Linux
shel it works, but when trying with Asterisk's System command it doesn't.
Successful Linux command:
echo | mutt -s "New fax" earohuanca at gmail.com -a /tmp/faxes/201306191111.tif
Unsuccessful Asterisk Command:
same => n,System(mutt -s "New fax" elder.arohuanca at
2010 Apr 09
2
res fax help
I have res_fax setup and working for the most part. However, I'm seeing
some fax machines drop the connection on me -
Apr 9 17:33:11] NOTICE[30809]: res_fax.c:906 generic_fax_exec: Channel
'DAHDI/1-1' did not return a frame; probably hung up.
-- Channel 0/1, span 1 got hangup, cause 102
-- Channel 'DAHDI/1-1' FAX session '20' is complete, result:
2018 May 21
5
Looking for better fax handling
I am having troubles with sending faxes. I hope someone can help me
work out a better method.
Basically we have a special address that our users can send to. It
winds up on our Asterisk server which runs a Python script that parses
the message for attachments and the phone number from the recipient
address. The attachments are converted to TIFF and stored in a folder
with various information
2014 May 26
2
dahdi "hungup" after each ring
Hi,
I guess something's wrong with my chan_dahdi configuration, ... but I can't
seem to get it.
When I test incoming calls on a DAHDI-channel (incoming from pstn),
asterisk seems to interpret it as a caller hangup after each ring.
Any ideas.
OUTPUT:
-- Starting simple switch on 'DAHDI/5-1'
-- Executing [s at from-pstn:1] *Verbose*("*DAHDI/5-1*",
2013 Aug 21
2
[PATCH 1/3] Rationalise whitespace to 4 space indentation with no trailing spaces
RHSrvAny.c was using a mixture of 4 space indentation, and tabs with a width of
4. This commit rationalises the whitespace to use only 4 space indentation, and
removes trailing whitespace.
---
RHSrvAny/RHSrvAny.c | 537 ++++++++++++++++++++++++++--------------------------
RHSrvAny/RHSrvAny.h | 1 -
RHSrvAny/resource.h | 2 +-
3 files changed, 269 insertions(+), 271 deletions(-)
diff --git
2013 May 01
1
Call "stuck" in queue
Asterisk 11.1.0
One queue with strategy=leastrecent. (Full queues.conf below.)
Occasionally (several times today), a caller will get "stuck" in the
queue - there are operators available to take the call, but the caller
stays in the queue for a long time. Any idea what might cause this, or
where I can start looking to debug it? I'm going to start digging
through the queue log
2014 May 21
1
issue installing voicemail imap support: imap_tk module missing
Hi,
I'm trying to install voicemail-imap support but there seems to be a
missing module:
imap_tk
checking for mandatory modules: IMAP_TK... fail
configure: ***
configure: *** The IMAP_TK installation appears to be missing or broken.
configure: *** Either correct the installation, or run configure
configure: *** including --without-imap.
My configuration
Ubuntu 14.04 LTS
Asterisk
2017 Apr 19
2
Voicemail asking for login
On 2017-04-18 08:17 PM, Pete Mundy wrote:
>> On 19/04/2017, at 7:58 am, D'Arcy Cain <darcy at VybeNetworks.com
>> <mailto:darcy at VybeNetworks.com>> wrote:
>>
>> <snip>
>> Everything looks the same as another one that works except for two
>> things. The one that works doesn't have the "Probation passed" lines.
>> I am
2015 Jun 12
0
Voice mail and caller ID
Try this for CHAN_SIP:
same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer
same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the mailbox
same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we have a
mailbox defined log into it
If you are using PJSIP it's more complex
same => n,Set(EndPoint=${CHANNEL(endpoint)}) ; Get the peer
same =>
2013 Jun 08
0
H.323 Trunk between Asterisk 11 and Avaya
Hello,
I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have
done this before between Asterisk 1.6 and Avaya but had some issues placing
external calls from the Asterisk to the Public network which is connected
to Avaya. I'm trying to create that trunk on Asterisk 11 because the 1.6 is
outdated and has no support.
On the Asterisk side I have Aastra 6731i SIP phones
2013 Aug 29
5
[PATCH 1/6] Rationalise whitespace to 4 space indentation with no trailing spaces
RHSrvAny.c was using a mixture of 4 space indentation, and tabs with a width of
4. This commit rationalises the whitespace to use only 4 space indentation, and
removes trailing whitespace.
---
RHSrvAny/RHSrvAny.c | 537 ++++++++++++++++++++++++++--------------------------
RHSrvAny/RHSrvAny.h | 1 -
RHSrvAny/resource.h | 2 +-
3 files changed, 269 insertions(+), 271 deletions(-)
diff --git
2017 Apr 18
4
Voicemail asking for login
On 2017-04-18 02:42 AM, Pete Mundy wrote:
> Try this:
>
> asterisk -r
> core set verbose 10
> [get user to trigger fault]
> [examine console output, and post to list if still unclear]
>
> If you don't solve it yourself, then we'll be able to help further once
> we've seen the output.
I can't see much more than at my previous debug level but here it is
2011 Jun 19
3
Problem with ReceiveFAX app from FFA
Hi all,
I am running to the following problem, when using the below dialplan to
receive fax, everything works perfect till this line
exten => receive,n,ReceiveFAX(${FAXFILE}):
and then the following line cannot be executed, it's like asterisk can't go
back to dialplan and continue, the good news is when i check what is
received in my fax folder i find that the file is a valid one (not
2005 Nov 23
0
Some code change suggestions of thenwin32-service package
> -----Original Message-----
> From: win32utils-devel-bounces at rubyforge.org
> [mailto:win32utils-devel-bounces at rubyforge.org] On Behalf Of
> Park Heesob
> Sent: Wednesday, November 23, 2005 6:16 AM
> To: Development and ideas for win32utils projects
> Subject: [Win32utils-devel] Some code change suggestions of
> thenwin32-service package
>
>
> Hi,
>
2015 Jun 12
2
Voice mail and caller ID
I have this in my sip.conf:
exten => *98,1,Verbose(0,CALLERID number is "${CALLERID(num)}")
same => n,VoicemailMain(${CALLERID(num)}@LocalSets,s)
same => n,Hangup
However, my extensions are set up so that they always show the external
number, not the extension:
[foobar2](client-phone)
secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx
callerid=Candace <5555551212>