similar to: Asterisk media problem behind NAT

Displaying 20 results from an estimated 1000 matches similar to: "Asterisk media problem behind NAT"

2004 Jan 19
1
Problems with source routing
Hello I have the following problem: LAN<--->LINUX_ROUTER<--> 2 internet gateways gateway1: adsl gateway2: ppp connection I want the following Machines from LAN going to Internet tcp port 80 :-> gateway1 Machines from LAN goint to Internet tcp port 22 :-> gateway2 Everything else: -> gateway1 How can I acomplish this? I am using kernel 2.4.24 Can I combine dead gateway
2012 Feb 06
4
[Bug 1977] New: ProxyCommand seems to no execute shell commands
https://bugzilla.mindrot.org/show_bug.cgi?id=1977 Bug #: 1977 Summary: ProxyCommand seems to no execute shell commands Classification: Unclassified Product: Portable OpenSSH Version: 5.9p1 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: ssh
2009 Oct 20
2
all our circuits are busy now
I am not sure why I am getting this message, I have an outbound route that goes to asterisk gateway1 then asterisk gateway2 When all lines on asterisk gateway1 are full, I get the message " all our circuits are busy now" then few second later, the phone rings, going to the second route! And the call can be established, how can I get rid of this message?? thanks --------------
2004 Aug 24
0
multiple gateways causes problem
Hi all, My machine has got two network cards and both of them are connected to two different ISPs. I have added two ISPs to my default gateway with same metric. Let a.b.c.d be the IP of 1st network card and e.f.g.h be the ip of the second network card i.e., Card1 - IP=a.b.c.d, Gateway=gateway1.isp1.net Card2 - IP=e.f.g.h, Gateway=gateway2.isp2.net route add -net 0/0 gw
2006 Oct 06
12
Two outbound internet links, using one network interface
Hi, I am trying to categorize the network traffic and to send it out across two different providers. For this I mark the packets in the firewall (in the PREROUTING chain of table mangle), and then use another routing table for the marked packets, which has a different gateway from the main routing table. Basicaly I am following the cookbook example in this page:
2005 Oct 05
0
Unwieldy outbound macro
I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP
2020 Nov 08
4
Can't join domain (LDAP error)
Hi, I'm trying to set up an AD DC in an iocage jail on FreeBSD (to avoid the issues of having the DC a file server) but I'm running into some trouble. I've setup Kerberos and can kinit OK: root at samba-addc:/ # kinit administrator administrator at BEGER.COM.AU's Password: root at samba-addc:/ # klist Credentials cache: FILE:/tmp/krb5cc_0 Principal: administrator at
2004 Jun 01
1
Stuck SIP channels? -> SIP show channels
Hello all I've discovered that SIP channels sometimes get stuck in *. I've read some posts from Fri 29 Aug 2003 which mentions this issue, but there doesn't seem to be any final answers I don't know if this is related to the 0001604 bug? Below is a list from one of the incidents: I know the (d) means that it is scheduled for destruction but the 10.1.1.45 channel hasn't
2007 Feb 27
1
Help understanding SIP SHOW CHANNELS
I have a high volume asterisk 1.40 installation and I ran a SIP SHOW CHANNELS. (see partial output below). My questions are: 1. "wc-l" of the output shows 4000 lines. Does this mean 2000 active calls? (2 channels per call) 2. The latter part of the output shows "unkn" for Form column. Why does it not know the codec? Could it be UDPTL? Or are these calls messed up? 3.
2007 Sep 06
1
Dead SIP channels
I am using a2billing as calling card platform with asterisk 1.2.17. After running for several days, if I issue 'sip show channels' command, I got a lot of dead sip channels although 'show channels' command only show 5 channels. What cause these dead channels? How can I clean out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean?
2008 Feb 13
3
urgent-channels
Hi All I am using asterisk 1.2.4 Please see the results when I execute Sip show channels X X X X x 192.168.8.106(None) 04cddc1f5a0 00101/00000 unkn No 215.96.142.83 (None) caac0846-cf 00101/00000 unkn No 192.168.8.106(None) 94910146-46 00101/00000 unkn No 192.168.8.106(None) 793ed1eb0f2 00101/00000 unkn No 85.219.172.253 (None)
2020 Nov 08
2
Can't join domain (LDAP error)
> On 8 Nov 2020, at 20:24, Rowland penny via samba <samba at lists.samba.org> wrote: >> ldbsearch does not work either: >> root at samba-addc:/ # samba-ldbsearch -H ldap://gateway2.beger.com.au -U beger/darius '(objectclass=person)' >> Failed to connect to ldap URL 'ldap://gateway2.beger.com.au' - LDAP client internal error: NT_STATUS_INVALID_PARAMETER
2020 Aug 30
0
Unable to do offline backup (start transaction stalls)
Hi, I'm trying to do an offline backup but it gets stuck here: root at gateway2:~ # samba-tool domain backup offline --targetdir=/var/db/samba-backup running backup on dirs: /var/db/samba4/private /var/db/samba4 /usr/local/etc Starting transaction on /var/db/samba4/private/secrets load: 0.06 cmd: tdbbackup 94560 [lockf] 316.69r 0.00u 0.00s 0% 2896k An online backup completes OK: root at
2020 Nov 08
0
Can't join domain (LDAP error)
On Sun, 2020-11-08 at 16:06 +1030, O'Connor, Daniel via samba wrote: > Hi, > I'm trying to set up an AD DC in an iocage jail on FreeBSD (to avoid the issues of having the DC a file server) but I'm running into some trouble. > > I've setup Kerberos and can kinit OK: > root at samba-addc:/ # kinit administrator > administrator at BEGER.COM.AU's Password: >
2020 Nov 08
0
Can't join domain (LDAP error)
On 08/11/2020 05:36, O'Connor, Daniel via samba wrote: > Hi, > I'm trying to set up an AD DC in an iocage jail on FreeBSD (to avoid the issues of having the DC a file server) but I'm running into some trouble. > > I've setup Kerberos and can kinit OK: > root at samba-addc:/ # kinit administrator > administrator at BEGER.COM.AU's Password: > root at
2020 Nov 08
0
Can't join domain (LDAP error)
On 08/11/2020 11:52, O'Connor, Daniel wrote: > >> On 8 Nov 2020, at 20:24, Rowland penny via samba <samba at lists.samba.org> wrote: >>> ldbsearch does not work either: >>> root at samba-addc:/ # samba-ldbsearch -H ldap://gateway2.beger.com.au -U beger/darius '(objectclass=person)' >>> Failed to connect to ldap URL
2006 Jun 22
3
Showing Current Calls
Can someone recommend the best way to view current calls in progress on the Asterisk console? Neither the 'show channels' or 'sip show channels' commands are easy to read. hestia*CLI> show channels Channel Location State Application(Data) SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2) SIP/2944079-e7f2
2007 Dec 07
2
7960 Won't Register Yet Multiple Attempts?
Hi List, I've got a 7960 that's behind NAT (nat_enabled: 1 and nat_received_processing: 1) and for whatever reason doesn't seem to register, or at least hold a registration. If both the phone and the router (netgear) are rebooted, the phone will register, take a few incoming/outgoing calls no problems, then a few hours later, it drops the registration and never re-registers. If the
2020 Aug 30
2
ID mapping with SFU not setting shell
Hi, I'm trying to use SFU to set user IDs, shells, etc. but I can't work out the right magic - it always seems to use default template shell and home directory (but the UID seems correct) The global section of my smb4.conf looks like so: # Global parameters [global] log level = all:2 netbios name = GATEWAY2 realm = BEGER.COM.AU server role = active
2003 Oct 23
0
WAS: Call pickup (*8) on SIP devices. Bug #116
I've attached two SIP debugs in reference to bug #116. They are from today's CVS build. 1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the call. After which, SIP(2) rings for about 30 seconds then stops. 2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before the call is answered. SIP(1&3) are Cisco 7960's and SIP(2) is a Polycom