similar to: Re: Fax support and 'f' DTMF tone extension & Asterisk mangling faxes

Displaying 20 results from an estimated 1300 matches similar to: "Re: Fax support and 'f' DTMF tone extension & Asterisk mangling faxes"

2005 May 16
4
[Bug 1041] Allow the admin to specify PAM service name
http://bugzilla.mindrot.org/show_bug.cgi?id=1041 Summary: Allow the admin to specify PAM service name Product: Portable OpenSSH Version: -current Platform: All OS/Version: Linux Status: NEW Severity: enhancement Priority: P2 Component: PAM support AssignedTo: bitbucket at mindrot.org
2015 Feb 25
2
Proxying of non "plain" SASL mechnisms.
Hi, I understand from earlier discussions that the reason dovecot doesn't support proxying of other SASL mechanisms than those which supply the plaintext password is that in general it would be possible to proxy any SASL mechanism since it might protect against man-in-the-middle attacks (which would prevent proxying). However, that has led to choice between letting users use PLAIN (or
2013 Dec 04
1
Testing failover and recovery
Hello, I've found GlusterFS to be an interesting project. Not so much experience of it (although from similar usecases with DRBD+NFS setups) so I setup some testcase to try out failover and recovery. For this I have a setup with two glusterfs servers (each is a VM) and one client (also a VM). I'm using GlusterFS 3.4 btw. The servers manages a gluster volume created as: gluster volume
2012 Jun 28
1
Rebalance failures
I am messing around with gluster management and I've added a couple bricks and did a rebalance, first fix-layout and then migrate data. When I do this I seem to get a lot of failures: gluster> volume rebalance MAIL status Node Rebalanced-files size scanned failures status --------- -----------
2004 Apr 28
1
Softfax/spandsp compilation
Only to signal that if you want to compile app_rxfax and app_txfax with last cvsses of asterisk you have to modify the patched versions of the apps directory Makefile to define the symbol _GNU_SOURCE example, the line to compile app_rxfax from: gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c to gcc -D_GNU_SOURCE -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c
2016 Feb 17
0
CEBA-2016:0178 CentOS 7 cyrus-sasl BugFix Update
CentOS Errata and Bugfix Advisory 2016:0178 Upstream details at : https://rhn.redhat.com/errata/RHBA-2016-0178.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: 41bcfe83e915dfe6408766d8c5d7d172fffab42e55c39f44ee7ded90ef9bbdfd cyrus-sasl-2.1.26-20.el7_2.i686.rpm
2012 Feb 25
2
Finding name of variable supplied as function argument
Greetings All. I want to do the following simple thing. I have defined a function med3x3() such that, given vectors X,Y, med3x3(X,Y) returns a 3x3 table where: Row 1: X > median(X) Row 2: X = median(X) Row 3: X < median(X) Col 1: Y < median(Y) Col 2: Y = median(Y) Col 3: Y > median(Y) (with intersections of these conditions for the individual cells). I can easily define fixed
2004 Jul 15
3
SIP to H323 call timeout
Hi all, I have the following setup: UAs ------------SER ------------------------ ASTERISK ---------------------GNUGK --------------- GWs SER is configured to route call requests from UAs to Asterisk. Asterisk is configured to receive the call on SIP channel and dial out to GNUGK over H323 channel. The problem I'm facing is that asterisk sends out the call request to GNUGK and times out
2004 Jun 01
9
Hyperthreading?
Are they any issues still with hyperthreading processors, I've read and been told by a few people to make sure its disabled in bios if I want to use * on a hyperthreading machine. Kind Regards, Chris Bond
2004 Jul 13
1
caller id problem on incominc call to x100p
hi, when i call asterisk (on x100p) i got this : CLI> -- Starting simple switch on 'Zap/7-1' Jul 13 15:03:34 ERROR[311316]: callerid.c:192 callerid_feed: fsk_serie made mylen < 0 (-9) Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4735 ss_thread: CallerID feed failed: Success Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4777 ss_thread: CallerID returned with error on channel
2005 Jun 10
1
Call inband progress indication and zaphfc
Hello all, I've a little clue with zaphfc used to connect to a BRI linethat probably can be a configuration issue.... (really I hope so) Here, telcos (expecially mobile operators) use to substitute the dialtone with some vocal indication without answer the line. (Indications like "The customer is not reachable" or "wait because the customer is on the phone" ecc..) For
2015 Mar 17
0
Proxying of non "plain" SASL mechnisms.
On 25 Feb 2015, at 20:59, Peter Mogensen <apm at one.com> wrote: > So, why not just extend the support for proxy authentication forwarding > to any single-handskake SASL-IR mechanism, which doesn't use > channel-binding? (which includes PLAIN, but also GS2-KRB5, and possibly > others). Yeah, I guess it would work for several of the auth mechanisms. It's a lot of work
2003 Nov 28
1
Problem with SIP-Phones and * audio-files
Hi All, I am a newbie to asterisk, and here is my first problem, where I do not know any further. I have to grandstream BT100 connected to asterisk. Working fine, for calling to each other, and to call via a IAX-Link to the outside. If I try to call the initial demo from the samples.extensions.conf I have nothing to hear. The CLI fine reports: -- Executing
2008 Nov 14
1
Problems running autoconf on Solaris 10 Update 4
Sorry for the double post, I've been doing some more looking into this, and it seems like this is a separate problem of more general interest than my original AD issue. My understanding is that the repository does not maintain a configure script in the source directory. Packaged releases of the samba code include one, but if you're trying to build from the repo, you need to do it
2003 Oct 28
0
Unable to find a path from G729A to ALAW, Unable to find a path from GSM to G729A
I have installed G729 but I cannot make a outgoing call with it. SIP/dennis-2c23 is making progress passing it to SIP/1010-8b60 NOTICE[311316]: File channel.c, Line 1476 (ast_set_read_format): Unable to find a path from G729A to ALAW NOTICE[311316]: File channel.c, Line 1446 (ast_set_write_format): Unable to find a path from GSM to G729A WARNING[311316]: File codec_gsm.c, Line 136
2013 Oct 10
2
utils.c: fwrite() returned error: Broken pipe how to solve it ???
Dear all, I want to make call through socket i have set code given below: #!/usr/bin/perl -w use IO::Socket::INET; sub asterisk_command () { # my $command=$_[0]; my $ami=IO::Socket::INET->new(PeerAddr=>'127.0.0.1',PeerPort=>5038,Proto=>'tcp') or die "failed to connect to AMI!"; print $ami "Action: Login\r\nUsername:
2004 Apr 23
3
Problem With zaphfc
I've this error How i can find the problem? Apr 23 12:24:43 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:47 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:48 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:24:53 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:02 WARNING[131081]: PRI:
2002 Jan 14
3
smbpasswd file NFS sharable?
If I want to make two servers use the same encrypted passwords can I share the smbpasswd file via NFS? Will updates/locking work correctly? Is there an alternative? Can LDAP/PAM use encrypted passwords? -- Gary Algier, WB2FWZ gaa@@ulticom.com +1 856 787 2758 Ulticom Inc., 1020 Briggs Rd, Mt. Laurel, NJ 08054 Fax:+1 856 866 2033 A self-addressed envelope
2004 Oct 05
2
Dialing a # in phone number?
Hi, I have not been successful in working out how to dial a # within a phone number. EG: exten => _12345,1,Dial(Zap/1/0868563823#,5,t) or exten => _08XXXXXXXX,1,Dial(Zap/1/${EXTEN}#) I'm trying to append a # character so that I can use a cellsocket (mobile phone to pots adapter) connected to an x100p. I think that asterisk is simply ignoring the # character. The docs on
2004 Jul 19
0
Setup for Go2call ? Or any SIP provider using phonejack or linejack g729 g723
Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723