similar to: Callerid and trunk

Displaying 20 results from an estimated 2000 matches similar to: "Callerid and trunk"

2007 Mar 21
4
FWD outgoing problem
I have configured iax.conf and extensions.conf as instructed on FWD website (http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76) and I can successfully receive calls and make test calls to 612, 613, etc. The problem is that I can not make a call to another FWD user. Here is what asterisk says: -- Executing [393xxxxxx@default:1] Set("Zap/1-1",
2006 May 29
4
registration at Voipbuster times out
Hi, I am new here on this list, and have a problem of which I hope that somebody here can help me with it. I have a Voipbuster account, with which I would like to make phone calls via my Asterisk PBX. If I let X-Lite register directly at voipbuster.com, everything is OK, but if I let Asterisk register there, it says "registration for XXXXXX@sip.voipbuster.com timed out, trying again",
2006 Feb 07
1
asterisk to FWD
Hello all, Here is my problem, I try to place a call to FWD (free world dialup) trough my asterisk PBX. my config is as follow: extensions.conf ---------------- [internal] exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD) exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD
2006 Jan 03
1
IAX2 channels denoted as '(None)'
I have some stuck channels that I think I'm going to have to bounce Asterisk to get rid of, but am curious to know what they are and how they've managed to accumulate. The show up with a channel identifier of '(None)' as in the output below, and do not show up in the soft hangup list, and so can't be cleared by that method. Here is the output from iax2 show channels:
2004 Oct 07
1
spandsp RxFAX problems.
Hello, Anyone else experiencing problems with the latest spandsp (pre3) and last libtiff beta? I'm getting 8 bytes long file, with the TIFF header only during such connection: -- Accepting call from 'XXXXXXX' to 'YYYYYY' on channel 0/2, span 1 -- Executing SetVar("Zap/2-1", "FAXFILE=/tmp/foch.tif") in new stack -- Executing
2017 Aug 19
4
xp: unknown user name or bad password
Hi All, Fedora Core 26 # rpm -qa \*samba\* samba-common-4.6.7-0.fc26.noarch samba-common-libs-4.6.7-0.fc26.x86_64 samba-4.6.7-0.fc26.x86_64 samba-client-libs-4.6.7-0.fc26.x86_64 samba-winbind-modules-4.6.7-0.fc26.x86_64 samba-libs-4.6.7-0.fc26.x86_64 samba-client-4.6.7-0.fc26.x86_64 samba-common-tools-4.6.7-0.fc26.x86_64 samba-winbind-4.6.7-0.fc26.x86_64 I am replacing a CentOS 5 Samba Server
2017 Aug 19
2
xp: unknown user name or bad password
On 08/19/2017 12:31 AM, Rowland Penny via samba wrote: > On Fri, 18 Aug 2017 19:53:26 -0700 > toddandmargo via samba <samba at lists.samba.org> wrote: > >> Hi All, >> >> Fedora Core 26 >> >> # rpm -qa \*samba\* >> samba-common-4.6.7-0.fc26.noarch >> samba-common-libs-4.6.7-0.fc26.x86_64 >> samba-4.6.7-0.fc26.x86_64 >>
2010 Mar 08
1
nss_winbind.so delivers first group only on Solaris 10
Hello, I'm trying to integrate some of our Solaris 10 10/09 hosts into Microsoft AD running on 2003/2008 R2 servers. After some compile trouble I finally managed to get the whole thing running including winbind in nsswitch.conf for users and groups and PAM for authentication. The problem is that winbind only reports the primary group of an AD user. 'wbinfo -r aduser' only reports
2008 Dec 15
2
cmusieve, vacation and error at file_dotlock_create
Hello again, managesieve is working. Now I tried a vacation script with the result that the vacation response is sent but I got he following error: deliver(test1 at xxxxxx.de): 2008-12-15 14:34:28 Error: file_dotlock_create(/home/vmail/%d/%n/Maildir/.dovecot.lda-dupes) failed: No such file or directory /home/vmail/xxxxxx.de/test1/Maildir/ is the correct dir, there are the other files like
2005 Feb 17
2
Warning in logs: gid of user does not exist
Hello. I've setup a FreeBSD 4.x machine with Samba 2.x and and OpenLDAP backend. I recently upgraded to Samba 3.0.10 and put "passdb backend=ldapsam_compat" in smb.conf. Now, everythings seems to work all right, but I see a lot of this entries in the logs: Feb 17 19:42:27 xxxxx smbd[44170]: get_alias_user_groups: gid of user yyyyyyy doesn't exist. Check your /etc/passwd
2009 Jan 28
1
asterisk-users Digest, Vol 54, Issue 94
> Date: Wed, 28 Jan 2009 13:11:19 -0600 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1? > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005> >
2003 Jul 30
2
building packages using S4 methods
I have been building a package around a sequence of S4 classes which I have coded in separate *.R files in the "./R" subdirectory of the package. The package builds without error, but when I load it in R I get: Error in reconcilePropertiesAndPrototype(name, slots, prototype, superClasses) : Class "xxxx" extends an undefined class ("yyyyyy" I guess R is trying to
2009 Apr 22
1
Queue() Ignore Hangup Request
I saw a few posts of this problem before I could not figure out the reason I am getting it. I am running RHEL 5, Asterisk 1.4.21.2, zaptel 1.4.11 and libpri 1.4.4 Basically, if I dial into a queue and hang up the phone, Asterisk did not detect the hangup request and Asterisk will only hang up when the timer expires. There is no such problem if I do not use Queue(). Any thoughts? Here is my
2010 May 12
1
problems with unicall
Hello, i'm using asterisk 1.4.9 in fedora 7, i was compiled its with this package: libpri-1.4.2 asterisk-1.4.9 spandsp-0.0.4 unicall-0.0.5pre1 libmfcr2-0.0.3 libsupertone-0.0.2 libunicall-0.0.3 zaptel-1.4.4 i'm using a E1 pci card with R2 but they not work, when I start the asterisk its generate this log: [May 12 08:53:24] WARNING[30814] channel.c: No channel type
2007 Jun 29
1
Asterisk 1.4 Warnnings
Dear Users ! I have recently installed asterisk 1.4 i got a warning message whenever i use reload or extensions reload. [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-dundi-e164-local'
2019 Oct 11
3
clarification on gosub, macros and AEL
I'm trying to clarify my understand of gosub, macros and AEL. My understanding is that macros using the Macro() application, which is defined in extensions.conf by: [macro-foo] ... and called in extensions.conf with exten => _9NXXNXXXXXX.,n,Macro(fastbusy) is deprecated in favour of Gosub(). True so far? But then there are "macro"s defined in extensions.ael: macro foo() {
2009 May 03
2
Asterisk not starting up due to database problems
When I try and start asterisk I get the following, however I have commented out the data the connections in res_mysql.conf and res_pgsql.conf. I am not sure therefore why I am getting these errors. Do I have to change something else to turn this off? Thanks Asterisk 1.4.21.2~dfsg-3, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk
2010 Aug 03
4
Dial() M parameter in 1.6.2.11-rc2
Hi, I can't figure out what syntax to use with the Dial() "M" parameter for the AEL parser to interpret properly. Creating an AEL macro named "macro-screen()" partly works as a hack, but it must not turn into a gosub properly, so I get warnings about the "return;". Dial(...,tgM(&screen)) with the ael macro named "screen" does not work
2006 May 29
2
Problem with IAX2 dialin with portunity
Hi, I'm using http://www.portunity.net/ I configured now asterisk with the following setup: iax.conf: register => XXXXXXX:YYYYYYY@iax.iaxport.de [portunity-out] type=friend host=iax.iaxport.de username=XXXXXXX secret=YYYYYY context=incoming-portunity notransfer=yes [guest] type=user context=default ;callerid="Guest IAX User" And in extensions.conf: [default] ;exten =>
2005 Mar 17
1
Different codecs for different numbers via same IAX provider; how? Configs included.
Hi, I have been trying to work this out and haven't been able to. I have some incoming numbers that come in over IAX, from the same server, and wish to use different codecs for different calls. This doesn't seem to work for incoming either. I cant seem to get any combination of allow/disallow to work.. Ideally the following would work: [general] register => XXXXXX disallow=all