search for: unrecheable

Hi, I'm experiencing a strange behavior on one of my asterisk servers. When I make a sip connection (sip.conf) between the 2 boxes using the primary interface it works fine. When however use an aliased interface (ethX:Y) it fails. from sip show peers: ast-rem-vpn 192.168.3.253 A 5060 UNREACHABLE when I use the ethX interface directly, it works fine. Did anyone

i have this configuration , An Asterisk server connected to my private LAN 192.168.10.0/24 when i do port forwarding for port 5060 so that i make a call from Internet into Asterisk wireshark show the message "destintion port unrechable" i configured sip.conf for "nat=yes" and "qualify=yes" and "externip="my public IP" did i forget some other ports

>Date: Wed, 23 Sep 1998 21:40:38 -0500 >From: "Daryl L. Biberdorf" <darylb@superserve.com> >To: samba@samba.anu.edu.au >Subject: Pages printed >Message-ID: <3.0.5.32.19980923214038.00816310@pop.a001.sprintmail.com> > >We have the need to track printer usage (pages used). >Right now, my Samba printer share defines the print command >as "print

On Mon, Jun 12, 2017 at 09:51:53AM +0200, Reindl Harald via samba wrote: > Am 12.06.2017 um 09:41 schrieb Mike Brown via samba: >> On Mon, Jun 12, 2017 at 09:28:20AM +0200, Reindl Harald via samba wrote: >>> Am 12.06.2017 um 09:03 schrieb Mike Brown via samba: >>>> On Mon, Jun 12, 2017 at 01:53:10PM +1200, Andrew Bartlett via samba wrote: >>>>> On Sun,

Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia

Hello tinc users, i have the follow configuration: 1 client/server called master, it is always reachable from internet (with dyndns) 5 clients, that connects to the master and the other clients (all behind a router (NAT)) master-hosts-file: Address = ... Port = ... Subnet = ... Compression = 0 ---- key ----- client-hosts-files: Compression = 0 Subnet = ... ----- key ----- tinc.conf Name = ....

On Mon, Jun 12, 2017 at 09:28:20AM +0200, Reindl Harald via samba wrote: > Am 12.06.2017 um 09:03 schrieb Mike Brown via samba: >> On Mon, Jun 12, 2017 at 01:53:10PM +1200, Andrew Bartlett via samba wrote: >>> On Sun, 2017-06-11 at 20:08 -0500, Mike Brown via samba wrote: >>>> Yes, I know, XP-SP3 is very old.????It works for what I need it for.????I >>>>

Am 12.06.2017 um 09:41 schrieb Mike Brown via samba: > On Mon, Jun 12, 2017 at 09:28:20AM +0200, Reindl Harald via samba wrote: >> Am 12.06.2017 um 09:03 schrieb Mike Brown via samba: >>> On Mon, Jun 12, 2017 at 01:53:10PM +1200, Andrew Bartlett via samba wrote: >>>> On Sun, 2017-06-11 at 20:08 -0500, Mike Brown via samba wrote: >>>>> Yes, I know,

Am 12.06.2017 um 10:00 schrieb Mike Brown via samba: > On Mon, Jun 12, 2017 at 09:51:53AM +0200, Reindl Harald via samba wrote: >>>>> Damn firewall. By default, Samba isn't allowed to connect. Found it by >>>>> using wireshark to look at the packets and that gave me the clue >>>> >>>> no need for wireshark - normally one does simply

Hi! I recently upgraded Asterisk from version 1.6.2 to 1.8.5 Now about every 10 minutes all SIP TRUNKS becomes UNRECHABLE for a few seconds or minutes after become LAGGED and later become OK. I have no idea of the cause of this problem. With the version 1.6.2 all runs perfectly. I can't say more because I have no idea where the problem is. Any suggestions? thanks -------------- next

...I route all LAN throu 195.5.56.40 and CISCO throu 195.5.56.41? And another thing... When I have connected both inet lines, and after that disconnect one of modems, my default route gateway is gone, and when I ping any host (excluding the endpoint of PPP-line 195.5.56.2), ping says ''Network unrecheable''. Why my PPP lines are not independent? _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

Hello, Did someone have a solution for a line fax detection for outgoing call For exemple I call number 0123456789 - if it is a fax then redirect to extension A - if it is a line then redirect to exention B whats ia want its somthing like AMD application that i use for the answering machine . http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD search in the wiki give this application :