similar to: SIP Security

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

Hi all, We're in the process of finally moving from our aging Samba3-based infrastructure across to Samba4. Rather than trying to migrate, we're just making a clean break so that we can do it properly from day one, as there were likely some mistakes that were made years ago we want to leave behind. The machines under test are virtual machines running Ubuntu 14.04 LTS AMD64. I've

Hi, the smbldap tools are a great thing to have, thanks a lot! I just wonder if it would be possible to set a password via the command line, like: smbldap-passwd.pl testuser1 not24get I have to create accounts for several hundreds people every year and give them an immutable password. So long I do that with smbpasswd, ldappasswd and a tiny script around these tools. I tried smbldap-passwd.pl

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

Helo you, I try to build a mail server based on Centos 7, postfix and dovecot 2. My backend is a Samba4 ad-dc. I tried a lot and I don't know what else I could try.I'm new to this mailing list so please forgive me if I don't give right information or anything Samba4 ad-dc is up incl. dns. Win10 Client joined domain and authentication works. Postfix is up and checks against ldap

I have a problem that is now becoming very annoying. Namely I have a Centos 7 member server running Sernet Samba 4.7.4 for which everything seems to work except gdm or ftp logins. On the linux client it seems winbindd is set up correctly. For example (the data shown below has been sanitized): > getent passwd testuser2:*:3001108:3000513::/home/testuser1:/bin/bash

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Hello, I am doing a samba pdc with ldap. When I try to login to the domain, or access some shares I get this in my ldap logs: Mar 7 16:46:16 localhost slapd[3588]: conn=25 op=4 SRCH base="ou=People,dc=test,dc=org" scope=1 filter="(&(objectClass=posixAccount)(uid=DOMAIN\5CTD))" Mar 7 16:46:16 localhost slapd[3588]: conn=25 op=4 SEARCH RESULT tag=101 err=0 text= My

Hello I have an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a drop in call. It does not have a certain time, it is random. The audio is flowing normally and the call is dropped. Has anyone ever experienced this? My settings changed below: allowoverlap = no udpbindaddr = 0.0.0.0 tcpenable = no tcpbindaddr = 0.0.0.0 transport = udp, ws, wss srvlookup = yes directmedia = no

I am facing a very strange problem with samba running as PDC and using LDAP. The new users that I create using smbldap-tools aren't able to log in, but they can browse shares on the server. I had the problem with samba 2.2 and hoped it would be solved with 3.0. I spent two days testing without success even with samba 3.0.2pre1. In the log I can't see anything wrong : it says "sam

" Hello, I have been trying to get my Grandstream GXP2000 phones to stop showing ringing state on monitored extensions. But no matter where I put notifyringing=no asterisk still sends the ringing state to the phones. Is this a bug I should report or is there another way around it. Here is how i have my subscriptions setup: extensions.conf [demo] exten => 6100,hint,SIP/100 exten =>

Hello. Thanks for the reply. Yes. In the traffic analyzed, the BYE is sent by the originator of the call, but there is no "human" hangup, but the asterisk one. BYE is sent, received and confirmed. I don't know how I could investigate the reason for this BYE. Em 21/09/2020 17:12, Dovid Bender escreveu: > Is there anything in the Asterisk logs? Which side sends the BYE? Were

Hello list, I'm stuck since 2 days and I have no clue how to troubleshoot and solve that problem. Any help really really appreciated. Scenario: ========= I am using Samba 4.1.12/sernet on DC1 (172.19.100.1) and DC2 (172.19.100.2) with default [netlogon] and [sysvol] share only. I installed an additional samba4 server with fileserving role which is called MEMBERSRV1 (172.19.100.3), which is

Hello We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF lamps on our Polycom phones stop working. After a lot of googling and a lot of testing, I have been unable to find a solution. I did try to change the call-limit value from 4 to 1, and this actually made BLF work (noone suggested this, and what documantation I can find states that this option is deprecated). This

Ishfaq Malik <ish at pack-net.co.uk> schrieb: Hi Ishfaq > Look into Busy Lamp Field/Presence > > Here's a starting point: > > http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html Thanks a lot, but it does not seems to work... Here my configuration: sip.conf: [general] allowsubscribe=yes subscribecontext = default

Am 23.06.2020 16:22, schrieb Marek Greško: > It seems your problems lie in something other. Most probably it is not > mtu problem. All my suspections are contradicted. If it is true you > have inter vlan voice quality problems, it is definitely something > different. Formerly I assumed you were trying only LTE vs LAN using > internet. I'm not sure what you mean with the last

On 01/02/16 02:52, Stuart Longland wrote: > Hi all, > > We're in the process of finally moving from our aging Samba3-based > infrastructure across to Samba4. > > Rather than trying to migrate, we're just making a clean break so that > we can do it properly from day one, as there were likely some mistakes > that were made years ago we want to leave behind. > >

Hello everyone, I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my queue interfaces, despite the fact it is free at that time, can you give help? 1. I see many sip channels from that extension: [root at mysweetpbx]# asterisk -rx "*sip show channels*" |grep 648 Peer User/ANR Call ID Seq (Tx/Rx) Format Hold

Dear Help, Initially, I thought that I had solved this problem, but it turns out that I haven't. I currently have Samba set up as a PDC with an eDirectory/LDAP backend. There are also a few Samba BDCs in play as well. If a user enters the correct password, there are no issues and everything authenticates fine. If I turn off the PDC and force a user to authenticate against a BDC with the