similar to: BOTH UAs behind same FW/NAT

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' -----------

hello, my questns are about few * functionality. 1) how can I make call tranfer. Not call parking. If I'm talking with some one a I want to tramnfer call to the another extension, to the other person. 2) how can I make call confernece. Not Meetme If I'm talking with some one and I want to join another person to our talk . I haven't found this in any manual :( hudecof -- mail:

hello, my situations is as follows. In our comapny we are planing to have *. I'm testing it now. If we will buy G729 codec for * ... UA(SIP) <-> FW <-> (SIP)*(H323) <-> (H323)GATEKEEPER(H323) <-> (H323)AS5300 <-> world the following equipment speeks G729: *, GK ,AS5300. All call from UA to another endpoing go through *, because of mixed SIP and H323

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

On Feb 18, 2015 7:43 PM, "Tris Hoar" <trishoar at bgfl.org> wrote: > > On 18/02/2015 07:17, aditya hilman wrote: >> >> Hi folks, >> >> I've already configured split DNS for internal-view and external-view. Also >> already configured the master - slave dns. >> But i've problem with external-view zone transfer. >> Based on the

Hi, I upgraded our servers from Wheezy to Jessie. I use samba in classic mode, with openldap backend. After the upgrade, on the PDC (srv3) everything seems to be ok, it authetnicates, the netlogon share is accessible on it, but on the BDC (srv7), what is the file server, the authentication doesn't work, shares are inaccessible. I compared and syncronized the configuration files to as similar

Hi folks, I've already configured split DNS for internal-view and external-view. Also already configured the master - slave dns. But i've problem with external-view zone transfer. Based on the logs, the master notify to slave using the public ip, which is not accessible by master to transfering the zone over public ip. Is it possible to transfer zone over local ip for external-view ?

Hi all, I've just been setting up such a phone with * and I'm facing some problems. At this time there is no customized extensions & others, so I use extension 8500 (voicemail) in default config. What I get when this extension is called from the SNOM is pasted below. The phone ignores *'s invites and * sends a "BYE" to it. Please note the IP in the "Contact"

Hi all, -------- I have a Asterisk at a public Network (official IP address). In the local network I have isntalled a Snom 200 IP phone and in my home network (behind NAT) a Snom 100 device. I can dial the Snom200 device from my home location without any problems but the Snom200 can not dial me. It always gets a "we do not rely". I tried to forward the SIP Port (5060) UDP via UPnP

I'm trying to get a Snom100 configured with H.323. Right now, the phone is not even connecting to the Asterisk server, so there's obviously a problem with the snom config. Does anybody have a sample working configuration with the snom phone, using H.323? I've checked the archives, but everybody seems to be using SIP with the Snom phone, not H.323. -Tilghman

On 18/02/2015 07:17, aditya hilman wrote: > Hi folks, > > I've already configured split DNS for internal-view and external-view. Also > already configured the master - slave dns. > But i've problem with external-view zone transfer. > Based on the logs, the master notify to slave using the public ip, which is > not accessible by master to transfering the zone over

On Wed, Feb 18, 2015 at 7:47 PM, aditya hilman <aditya.hilman at gmail.com> wrote: > > On Feb 18, 2015 7:43 PM, "Tris Hoar" <trishoar at bgfl.org> wrote: > > > > On 18/02/2015 07:17, aditya hilman wrote: > >> > >> Hi folks, > >> > >> I've already configured split DNS for internal-view and external-view. > Also >

OK I did some researches and tests with it, and finally: I registered my messenger to the asterisk and called if from the snom. The audio from the snom to the messenger was PERFECT. By the time of the call This message was running on the asterisk console: WARNING[16400]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames My conclusion is that the snom100 utilizes MSGSM

Hi, there were some postings a few weeks ago telling that the GSM codec problem with snom100 will be fixed. But it still seems to be very quality. Will be any change in this subject? THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/2fa9d206/attachment.htm

Hello freebsd-stable, i know this is may be the wrong place to ask these question, i just cvsup my 5.1 box today, and during the make installworld it's stop with these following error messages: PASS: Test notdef detected no regression. Running test modifiers PASS: Test modifiers detected no regression. Running test funny_targets PASS: Test funny_targets detected no regression.

Hi all, ------ I tried to use * over satellite, but all my effort did not succeed. The Asterisk is behind the VSAT and is resposibel for alle the SIP clients in a field location. The clients are notebooks and PDA's running SJPhoen for Windows and PocketPC. Unfortunately I could not find any Linux Client wich worked satisfying. SJ LAbs promised a Linux Version at the end of August but they

sorry, this problem is fixed by myself. we must need set 'canreinvite=no' each user. --- I'm try to discconect a call with SIP. when caller make a call, 'show channels' result is following. mack*CLI> show channels Channel (Context Extension Pri ) State Appl. Data SIP/mack-1bfc (default 1 ) Ringing AppDial (Outgoing

Well? I?m not convinced KEA can?t be simple DHCP, though I understand that one would never know that from reading the KEA documentation. The following is a complete KEA configuration suitable to do a single subnet. In fact, you could probably get away with slightly less. I?ve replaced the unique addresses from my environment with placeholders (e.g. <network/mask>), but otherwise, this is a

Hi, For some time I've fiddled with Debian and Ubuntu LTS. There's one really nice feature for local networks: apt-cacher, a package proxy for APT. My company is in the remote South French countryside, and more often than not, schools and public libraries only have some very limited Internet access with relatively low bandwidth, which can make the updating process very tedious. A