have posted a while ago on issues of receiving faxes by an Asterisk host
using an x100p fxo interface attached to BT pstn
the asterisk installation is the cvs download as of 23/09/04
is anyone able to confirm that the rxfax / txfax application that seems to
be 'bundled' in thecvs download is the
-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, September 26, 2004 3:22 PM
Subject: Re: [Asterisk-Users] spandsp
Graham Turner wrote:
have posted a while ago on issues of receiving faxes by an Asterisk host
using an x100p fxo interface attached to BT pstn
the asterisk installation
i am experiencing errors with the rxfax application when receiving faxes
from a 'brother' fax device.
the rxfax application picks up the incoming fax but the subseqeuent
'negotiation' process seems to fail with the messages logged to the asterisk
console as below;
fast carrier up
coarse
apologies as i forget to mention to the receiving device connected to PSTN
is x100p fxo i/f
- Original Message -
From: Graham Turner [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 24, 2004 1:12 PM
Subject: latest cvs / spandsp
i am experiencing errors with the rxfax
-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, September 21, 2004 10:41 AM
Subject: RE: [Asterisk-Users] uk caller id
Graham Turner [EMAIL PROTECTED] lazily top-posted:
i have installed asterisk / zaptel from cvs distribution as of 17/09/04
so i assume this does it
If you have a TDM
I am attempting installation of spandsp on to an Asterisk installation on
Linux RH9
the distribution i am using is that are URL http://ftp2.tootai.net - the
README for which i have followed verbatim -
my only issue on this was the target for the port.h / tif_dir.h / tiffiop.h
files in the
, 2004 3:54 PM
Subject: Re: [Asterisk-Users] spandsp / compilation errors
Graham Turner a écrit :
I am attempting installation of spandsp on to an Asterisk installation
on
Linux RH9
the distribution i am using is that are URL http://ftp2.tootai.net - the
README for which i have followed
Add '/usr/local/lib' to /etc/ld.so.conf if not already, and run
'ldconfig' as root. Then start asterisk.
On Mon, 2004-09-20 at 09:45, Graham Turner wrote:
Daniel, thanks for mail back - this has got me much further through
spandsp
installation process
i have progressed through your
Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, September 19, 2004 1:48 PM
Subject: RE: [Asterisk-Users] uk caller id
Graham Turner [EMAIL PROTECTED] wrote:
dear all, i am looking to enable CALLERID on an Asterisk system
comprising a X101P FXO interface
dear all, i am looking to enable CALLERID on an Asterisk system comprising a
X101P FXO interface connecting to BT PSTN in the uk
seems this is supported by the interface but there seems to be varying
information on how to enable it in zapata.conf
1. usecallerid=uk
2. ukcallerid=yes
being two
it seems that the use of renivite in sip peer configuration is very much
dependent on sip endpoint
have read of what seems defnite no no when using the cisco ata 186
it seems eminently preferable from a networking / performance view for the
media data transfer to be between the two endpoints and
looking to move an asterisk pbx server to a different vlan and as such
looking to check the impact of this change on the asterisk application
obviously we have the linux interface reconfiguration to complete
are there any application level settings that need to be changed to reflect
the changed
was wondering if someone could give any indication of the messages that are
appearing on the console of an Asterisk PBX
WARNING[1116941120]: chan_sip.c:532 retrans_pkt: Maximum retries exceeded on
call [EMAIL PROTECTED] for seqno 103 (non-critical request)
192.168.90.1 is a 7940 ip phone
was wondering if anyone could give us a run through
an explanation of the wiki and other examples of connecting to iptel's sip
express router using asterisk pbx so i can understand better the call
processing ..
given the example i work from on john todd's
www.loligo.com site ;
exten =
was wondering if anyone could give us a run through an explanation of the
wiki and other examples of connecting to iptel's sip express router using
asterisk pbx so i can understand better the call processing ..
given the example i work from on john todd's www.loligo.com site ;
exten =
dear all, was hoping someone could give me instruction on the syntax of
extension pattern matching for letters
the proposed 'dial plan' is one where any letter in the dialled digits
causes the pbx to assume we are dilaling a sip url and as such forward to
the appropraite sip service provider
was
have managed to establish voicemail functionality using voicemail /
voicemailmain applications
the documentation on these applications from digium.com suggests that
voicemail greetings are customizable (as one would be expect), but am not
able to find any supporting documentation
can anyone
to configure the voicemail URI on the 7940 so that it
calls extension 8500.
One nice thing about the Cisco phone is that they will keep track of WMI
separately for each configured line.
-brian
Graham Turner wrote:
can anyone give me a reference to the retrieval of voicemail from the
Asterisk
hopefully a quick and not too daft a question but just wanted to check to
see if the extension numbers as defined in extensions.conf and the sections
of sip.conf needed to have a numbering plan that was exclusive of the
numbers that are allocated to the mailbox numbers that are established for
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