similar to: include sip configuration from another file in sip.conf

Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register =>

>From my .bash_history on the schema master DC, effectively: # sed -e 's/${DOMAINDN}/dc=MYDOMAIN,dc=MY,dc=TLD/g' \ -e 's/${NETBIOSNAME}/MYDOMAIN/g' \ -e 's/${NISDOMAIN}/MYDOMAIN/g' \ /usr/local/samba/share/setup/ypServ30.ldif > ypServ30-JMH.ldif # service samba4 stop # ldbmodify -H /usr/local/samba/private/sam.ldb

I *think* I may have encountered a bug, or a feature, in the idmap/winbind area. I have recently added rfc2307 attributes to my AD, and am in the process of switching over. This means that I still have (unintentionally) some files/directories/etc. around with old UIDs e.g. 3000007, rather than my rfc2307 specified UIDs. What I am seeing is that the SID2XID mapping is initially correct for a

On 13/06/15 00:01, Jonathan Hunter wrote: > On 12 June 2015 at 08:55, Jonathan Hunter <jmhunter1 at gmail.com> wrote: >> Thanks buhorojo. The sssd list came up trumps here. When changing ID >> mappings, the sssd database must be manually removed (rm >> /var/lib/sss/db/*). I now have sssd working again :) >> >> I shall keep an eye on the mappings during the

Thanks buhorojo. The sssd list came up trumps here. When changing ID mappings, the sssd database must be manually removed (rm /var/lib/sss/db/*). I now have sssd working again :) I shall keep an eye on the mappings during the day today.. On 12 June 2015 at 07:36, buhorojo <buhorojo.lcb at gmail.com> wrote: > On 12/06/15 01:34, Jonathan Hunter wrote: >> >> On 11 June 2015 at

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 ==

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

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

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

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

Hello list, I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to sipgate.co.uk as a sip agent/client (with "register =>" statement in sip.conf). If I restrict the number of ports used in rtp.conf (to 10000-10005 for example) - will that affect the sip sessions to sipgate.co.uk as well - or only those sessions where Asterisk acts as a sip proxy/server? Many thanks,

Hello everybody, after upgrading dovecot on a debian wheezy installation from the standard package version (dovecot 2.1.7) to dovecot 2.2.13-11 from wheezy-backports, i noticed some errors in my logs... Apr 28 22:00:13 lmtp(4879, xxxxxxxxx at unipd.it): Info: copy from <lmtp DATA>: box=INBOX, uid=error, msgid=<20150428200011.47D801F32 at mydoom.unipd.it>, size=1523 They are not

Hello, I've problems with following - ----- --- --- PSTN | --- isdn --- | A | ----- iax2 ------ | B | ----- --- --- On [B], there is unconditional call forwarding set back via [A] (dialparties.agi is used) to PSTN. So, call from PSTN is routed via [A] to [B] and than back again into PSTN.

Checkout http://www.voip-info.org/wiki-Asterisk+variables I believe that should have the answer for you. furthermore assuming that your number is always going to be 12 digits. exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number. Hope this helps. Umar On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote: > Hi, > > this

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",

Am 18.09.19 um 16:17 schrieb Rowland penny: > On 18/09/2019 03:41, Simeon Peter via samba wrote: >> I would remove any uidNumber & gidNumber attributes from the >> following users (if set): >>> administrator >>> guest >>> krbtgt >> Administrator has a uidNumber since long time and owns some files. >> Are there disadvantages if I leave his

Hi again, Am Dienstag, 2. Februar 2016, 12:09:59 CET schrieb Rowland penny: > On 02/02/16 11:26, Markus Dellermann wrote: > > Am Dienstag, 2. Februar 2016, 09:51:03 CET schrieb Rowland penny: > >> On 01/02/16 22:24, Markus Dellermann wrote: [....] > Ok, there are two schools of thought here, you can give Administrator a > uidNumber attribute, but this, as far as Unix is

Hello everybody I'm trying to enable the Digium res_fax app at my *11.7 Server. a fax show stats comes up with FAX Statistics: --------------- Current Sessions : 0 Reserved Sessions : 0 Transmit Attempts : 0 Receive Attempts : 1 Completed FAXes : 1 Failed FAXes : 1 Digium G.711 Licensed Channels : 1 Max Concurrent : 0 Success : 0 Switched to

Hello, I am using Winbind and Samba to tie in Linux VMs with a Windows DC for authentication. The UIDs are automatically assigned when I create a new user in Active Directory. I need to manually change the UIDs of my users to a UID that is used on another network so they can access a specific share. How can I change the UID successfully and still be able to authenticate with the DC. Also, how

On 25.02.2019 10:20, Rowland Penny via samba wrote: > On Mon, 25 Feb 2019 09:24:24 +0100 > Viktor Trojanovic via samba <samba at lists.samba.org> wrote: > > > >>>> I'm confused.. how is the choice of the idmap backend related to an >>>> AD DC use case? >>> Only in the case of wanting the same ID everywhere. >> In my understanding, the