similar to: granstream, vlan, tftp

hello, my situation is h323gw - gatekeeper - asterisk - SIP client my problem is, that I can't make call from h323gw, when this GW is Micronet (sp5004). A ----------- CUT ----------- -- Executing Wait("H323/ip$62.152.225.18:52434/20702", "1") in new stack == Spawn extension (postel, 169, 1) exited non-zero on 'H323/ip$62.152.225.18:52434/20702' -----------

Hi All, I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as well as any configuration in asterisk thats needed, on this phone? Thank you, James -------------- next part -------------- An HTML attachment was scrubbed...

I recently posted a question RE the Sipura 941 and using different ring tones, Thanks to hads I managed to use SET(_ALERT_INFO=Classic-1) to achieve this but trying this on the GXP 2000's didn't seem to do the trick?? Has anyone one had any luck on this topic? Also haven't been able to find any info on an auto-answer for the GXP 2000, again, I have succeeded in doing so with the

To all: (newbie) I have setup a BT 100 phone and mostly everthing is working pretty good except for the message button. I have place value in the appropiate field in the web configuration but nothing seems to work. When I press the button the speakerphone led goes on but the phone does nothing else (no dialtone, no sip request to *). Does anyone have this buttton working? I would like to

We are having many successes with Asterisk and starting to get the hang of it. But, I am still having problems getting my Budgetone BT100 (firmware 1.0.4.50) to work fully. I can receive calls, but cannot make them. We have the latest version of Asterisk, Fedora Core 3, Digium TDM400P with one FXO and one FXS card configured and working well. We have a PSTN line going into the Digium card,

Hello, can anyone help me please? I have installed libxml2, libogg, libvorbis, and then i have cvsed and installed libshout (autogen.sh && make && make install). Then I cvsed ices and tried to run autogen.sh. It just quit with: checking for shout-config... /usr/local/bin/shout-config checking libshout settings Using libshout config in /usr/local/bin/shout-config checking

hello, I'm quite new to dovecot. My instalation is working good Debian Etch - exim: 4.63-17 - dovecot: 1.0.rc15 I'm using virtual user setup, with LDA delivery method. There is no database backend at this time. Is there any way to configure exim to reject non-existing user /asking dovecot lda/ at SMTP time. The standard exim configuration in ACL section do not work: --- cut ---

hi all, I don't know how to setup asterix to work as PBX. If I want just basic configuration with 2 SIP phones (snom, ata), what all I have to write in the configuration files, or respectively in the configuration of ata and snom ? If there is any good documention available, send me URL too. All (ata, snom) are behind firewall (nat) and astrix is on the public IP, but I can move for

hi, I known, that this is not * related, but a lot of members of this ML uses GS BT phones. I have patched the atftp serber to recognize the TFTP OPTION, whis these phone send during boot. Patch includes - another locations for configs, firmware and ring tones - different FW versions for phones - custom ring tones for the phones You can find patch, source/unpatched/ and DEB for

Hi, by XFS do you mean filesystem quotas, yes ? regards Peter > On 14 Mar 2019, at 15:19, Edgaras Luko?evi?ius via dovecot <dovecot at dovecot.org> wrote: > > I was fighting domain quota for a long time, too. And I was never really successful at it, because group/domain quotas have multiple problems, and domain quota recalc is just one of them :) > > After a long trial

I was fighting domain quota for a long time, too. And I was never really successful at it, because group/domain quotas have multiple problems, and domain quota recalc is just one of them :) After a long trial and error I ended up implementing XFS project quotas. On 14/03/2019 15:14, Peter Hudec via dovecot wrote: > Hi all, > > I?m a little bit confused about the quota plugin in

Hello, hmmmm sweet problem. All I can recommend you is have you tried another player? I've heard that WinAmp3 has certain problems with .ogg streaming. No other idea :(. Martin On Fri May 16 2003 11:14, photos-ifree wrote: > Hy, > I've just installed the latest release of icecast2 and ices from CVS. > I have a linux Slackware 9.0 with the default packages. > After install

On 4/29/2016 10:14 PM, Jeremy Allison wrote: > On Tue, Apr 26, 2016 at 10:18:26PM +0100, Andrew Watkins wrote: >> Hello, >> >> I have noticed that smbclient returns an error code when a windows >> server has disabled NetBIOS, but should it do this, if you know >> NetBIOS has been disabled. I think there should be a flag to >> indicate that you don't want

Hello, I have noticed that smbclient returns an error code when a windows server has disabled NetBIOS, but should it do this, if you know NetBIOS has been disabled. I think there should be a flag to indicate that you don't want to check port 139 (NetBIOS). For example the following command which tries to connect to a Windows server with NetBIOS disabled returns an ERROR! $ smbclient -L

I think I have something misconfigured regarding voicemails. They work great, I have this setup: Sip.conf [ext1] Context=phones Mailbox=201 Voicemail.conf [home] 201,password,name,email@mail Voicemail delivery and all works great but when I check sip extension ext1 (analog phone using a Granstream ATA 286), the stutter tone signaling message waiting does not work. Anything wrong with

Good evening all. I'm having this strange behavior when dialing two or more simultaneus calls via IAX to other * boxes. Sound starts to have more latency, wich increments until it's almost impossible to talk (6 or more seconds), I try this calling with two grandstreams, one grandstream one tdm410p, one tdm410p and sjphone, sjphone and one grandstream, the result are similar.

Anyone in need of some of these? Garrett Smith Sales Executive garrett.smith@b2llc.com B2 Technologies 454 Sonwil Drive Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales Specializing in New and Used equipment from vendors including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, Granstream, Snom, Mediatrix,

Hi All We have a private network setup (no nat) with three types of phones connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco 7940 IP phones. When we ring polycom to grandstream or grandstream to polycom then both phones can send and receive voice fine and all is well. When we dial any combination of Cisco and either Polycom, or Granstream the Cisco, no voice is being sent

Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten => 0,1,Playback(pls-wait-connect-call) same=> n,SIPAddHeader(Alert-Info:;info=ring3) same=> n,Dial(SIP/310&SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com

Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All