search for: hartmann

...phone? If this is not an available feature, any ideas on the difficulty in making this feature? Example: Extensions.conf exten?=>?700,1,AgentCallbackLogin(${CALLERIDNUM}|?AnnounceCAllQue-TechSu pport?); ....... exten => s,6,Queue(queue1) Agents.conf agent => 2204,1234,Ron Hartmann VoiceFile... AnnounceCAllQue-TechSupport contains Allison saying "This call is from the Technical Support Call Queue Please press # to accept the call" Ron Hartmann dials 700, enters the agentid and password and extension he is working at. A Call comes into the office... the ca...

.../de_DE/ de_DE contails all the .sln16 and .gsm files, owned by asterisk:asterisk, from the source above. Since the prompting of numbers and the date works well, but not the voicemail prompt, there is something fishy I can not fathom. For your help I'd like to thank in advance, Oliver -- O. Hartmann Ich widerspreche der Nutzung oder ?bermittlung meiner Daten f?r Werbezwecke oder f?r die Markt- oder Meinungsforschung (? 28 Abs. 4 BDSG). -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 313 bytes Desc: OpenPGP...

I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. Asterisk is behind a NAT router, the physical setup is very much a trivial one. The Asterisk PBX is supposed to act as the telephone gateway for several VoIP/SIP phones. I'm using throughout pjsip as configuration, I have no experience with chan_sip since I started recently using Asterisk for several SoHo and lab's

...th0 with rrdtool. Or does anybody know of such a script , which is available for download ? I assume the bps in "rate 5728bps 34pps" is Byte per Second. Is that right ? How can i set all the counters back to zero ? I did not found anything in the manpage of tc. regards Joerg -- Jörg Hartmann Tel: +49 391 40 00 125 J.Hartmann@megalearn.de _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

Is for these UFS bugs in FreeBSD since 6.1 a fix uderway? See: http://projects.info-pull.com/mokb/ MOKB-08-11-2006,CVE-2006-5824, MOKB-03-11-2006,CVE-2006-5679 Regards, Oliver

Is for these UFS bugs in FreeBSD since 6.1 a fix uderway? See: http://projects.info-pull.com/mokb/ MOKB-08-11-2006,CVE-2006-5824, MOKB-03-11-2006,CVE-2006-5679 Regards, Oliver

...s the call through to an endpoint and then it takes another 10 seconds until the endpoint starts ringing (it is in fact a group of phones ringing alltogether). I can not see anything unusual with the underlying OS or some critical debug messages from asterisk itself. Any ideas? Kind regards, O. Hartmann [...] ==>> START [Nov 15 13:21:06] == Setting global variable 'SIPDOMAIN' to '192.168.2.1' [Nov 15 13:21:15] == Using SIP RTP Audio TOS bits 184 [Nov 15 13:21:15] -- Executing [511 at internalsip_o2:1] NoOp("PJSIP/501-00000008", "") in new stack [N...

Hi, I have a problem where I can join an xpsp2 machine to a domain but, no matter what %COMPUTERNAME% i use, it says "system error: a duplicate name exists on the network" after the reboot when upon successfully joining. If I try to log in as a valid user, i get the "the system could not log you on because domain 'DOMAIN' is not available". I'd just like to

...g the 192.168.254.1:5060, and since the VoIP phones are all in 192.168.254.0/24, this results obviously in an error. This is surprising me :-( How to deal with this without adding more network complexity like routing (by putting the phones into a subnet or other network)? Kind regards, oh -- O. Hartmann -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 313 bytes Desc: OpenPGP digital signature URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170729/bb0b3798/attachment.pgp>

...ether # Ethernet support #device tun # Packet tunnel. device pty # Pseudo-ttys (telnet etc) #device gif # IPv6 and IPv4 tunneling #device faith # IPv6-to-IPv4 relaying (translation) device bpf # Berkeley packet filter ------------------ Thanks a lot for your help, Oliver -- MfG O. Hartmann ohartman@mail.physik.uni-mainz.de ------------------------------------------------------------------ Systemadministration des Institutes fuer Physik der Atmosphaere (IPA) ------------------------------------------------------------------ Johannes Gutenberg Universitaet Mainz Becherweg 21 55099 Mai...

Good Day all, I have a Fractional PRI connected to my Asterisk Box via a T100P card. When I initiate a call out to phone number 123-8888 the call sounds great no echo what so ever. If the person at 123-8888 hangs up and calls me right back (same handset on both sides) same trunk line The call always has echo on it. The Asterisk sip extension hears them selves echoing. The remote party

Anyone know if it is possible to control how aggressively the "Aggressive" mode behaves. Meaning, is it possible to dial back the aggressive mode to have a happy medium between Regular and the Aggressive defaults. I have a situation where Normal echo cancellation is not quite enough, however when I turn on aggressive mode We are attacking it to hard and I am unhappy with the walkie

...change of the source code path (old: /usr/src/usr.bin/groff, doing a make clean depend obj all install, new: /usr/src/contrib/groff, no make would work correctly). Can anyone help? How to repair groff? In the past I did not need setup the environment manually. Thanks in advance, Oliver -- MfG O. Hartmann ohartman@mail.physik.uni-mainz.de ------------------------------------------------------------------ Systemadministration des Institutes fuer Physik der Atmosphaere (IPA) ------------------------------------------------------------------ Johannes Gutenberg Universitaet Mainz Becherweg 21 55099 Mai...

Hi all, I am not 100% sure if this is a -dev or a -user topic, but I am leaning towards the former. Feel free to cuss at me and tell me to ask -user, instead. I used to run a patchset that allowed full logs of everything taking place via OpenSSH. This also allowed me to replay any session, live or after the fact. I am fully aware of the security implications of logging everything, especially

...ut),RemoveQueueMember(${ARG1},SIP/${MACRO_EXTEN:4}) exten => s,n,Set(DEVICE_STATE(Custom:q${MACRO_EXTEN:0:4}_a${CALLBACKNUM})=NOT_INUSE) exten => s,n,UserEvent(Agentlogoff,Agent: ${CALLBACKNUM}) exten => s,n,Playback(agent-loggedoff) exten => s,n,Hangup() Alec _____ From: Ron Hartmann [mailto:ron_hartmann at hotmail.com] Sent: Saturday, 4 May 2013 3:08 a.m. To: Alec Davis Subject: RE: BLF and asterisk Queue Alec, I was able to get this working and my staff loves it. Thanks a million!!!!!!!! Now off to find a way to show an agents status via blf :-) ~ron _____ F...

Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker

...eds to be answered no matter what scheme is ultimately chosen, is this: Under what circumstances should the query string be appended to the path? -- Gabriel On Mon, Jun 29, 2015 at 4:03 AM Richard W.M. Jones <rjones@redhat.com> wrote: > On Thu, Jun 25, 2015 at 06:44:50PM +0000, Gabriel Hartmann wrote: > > I have written a patch (please see attached) which fixes both of these > bugs: > > > > https://bugzilla.redhat.com/show_bug.cgi?id=1092583 > > https://bugzilla.redhat.com/show_bug.cgi?id=1232477 > > > > By default, when saving a URI using xmlSaveUri...

I have written a patch (please see attached) which fixes both of these bugs: https://bugzilla.redhat.com/show_bug.cgi?id=1092583 https://bugzilla.redhat.com/show_bug.cgi?id=1232477 By default, when saving a URI using xmlSaveUri it escapes everything in the URI. QEMU doesn't want anything escaped, so now I unescape everything after the URI is generated. Unfortunately there's no flag to

I have been getting quite a bit of PRI Resets using my Quad PRI Digium card. Prior to the resets I am getting similar notices to the following chan_zap.c:7482 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 Telco claims the PRI's are fine on their end and that it is my unit. Is this timing? (google somewhat leads to this) I am running 1.08 asterisk zaptel

Hi All, Here's the latest patch. I think this should address the problem. The query string is now only appended to the end of a URI in the HTTP and HTTPS cases. The add-uri test now passes, and 'make check' still passes. -- Gabriel