similar to: Using frequent keepalives to eliminate need forNAT port forwarding?

I have an asterisk system behind NAT, and need to connect to public PSTN originators via SIP or IAX2, but don't have the option of forwarding any ports (4569, 5060, etc) to the asterisk system. However, the NAT system does properly establish transient UDP forwarding on the basis of outgoing connections, so is it possible to configure asterisk to send frequent keepalive UDP packets (say every

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

Hi, I think I encountered a bug in the qualify timers for IAX on asterisk 1.8 but I'd like to check if I'm not messing up in my config somewhere before reporting a bug. In my IAX peer configuration I have this: [remote-host] type=friend host=172.16.6.45 username=remote-host secret=test notransfer=yes qualify=16000 qualifyfreqnotok=30000 disallow=all allow=alaw allow=ulaw allow=ilbc

This has come up before on the list and archives but I don't seem to find a solution for this. On just a few nodes we have this situation where we see the IP disappear from the CLI iax2 show peers list but the status shows OK: 3012/3012 (Unspecified) (D) 255.255.255.255 0 OK (89 ms) How can the status be OK a few milliseconds ago and have no IP ?? The strange thing is

Ich habe eine dringende Frage!!!!!!!! ch ben?tze das Rplus mit Windows 98 . Wo kann ich die Vector heap von Rgui.exe erh?hen. Bei Eigenschaften kann ich das nicht machen. Vielen Dank f?r Eure Bem?hungen Nadine Brauchli, FORNAT *************************************************************************** FORNAT Forschungsstelle f?r angewandte ?kologie und Naturschutz Asylstr. 46/48 Postfach 8708

Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12

Hi folks, I have a couple of observations about NetBIOS keepalive handling in Samba, in case anyone is interested. I came across a few oddities when investigating why some of our Samba servers were logging: lib/util_sock.c:write_socket_data(540) write_socket_data: write failure. Error = Broken pipe type errors on a very frequent and regular basis. We have a number of Solaris 2.6 machines,

Need SIP KEEPALIVES in Asterisk, but QUALIFY won't presently work for you (due to it's channel disabling behavior)? Someone posted on the list that they would like to split "keepalives" and "qualify" into different features. Sounds like a good plan, but until that is done you can turn "qualify=" into a keepalive mechanism, without disabling your channels.

    Not these particular two servers. On 02/03/20 12:16, Doug Lytle wrote: >>>> I am trying to troubleshoot two Asterisk servers that have an IAX2 >>>> trunk between them. > Carlos, > > Had caller-id ever worked between these two systems? > > Doug > -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161

Hi helpful list! I have this user that has two different accounts om the mail server. One is a system account, the other is a virtual account (for what it is worth...). This user would like to have all his email to the virtual mail box (maildir format) automatically moved (or delivered) to the system mailbox (also maildir fornat). Both are on the same machine and use the same postfix/dovecot

I am trying to tweak my dial plan and I am running into a problem. Sometimes my VoIP out bound calls do not complete on overseas calls(busy or just a hang-up). Is there a way in the dial plan to automatically dial out of my PRI when something like this happens. Either by time limit by a failure event? Any point in the right direction would be great Thanks, CLI output (cleansed to protect the

Please help with this strange issue. When "sip show peers" returns status "Unknown" the CDR does not include the accountcode even though the call is correctly processed. I'm using A2 Billing and it uses the accountcode to determine the authentication. Asterisk version 1.4.21.2 I'm calling from a Quintum device. I'm very puzzeled. Name/username Host

Just a shot in the dark... but have you tried Answer() before Playback()? Josh McAllister -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Engleward Sent: Monday, May 01, 2006 11:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] /var/spool/asterisk/outgoing/ prematurely hangingup I have

I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse & repeat. Using the IAX2 debugging, I'm seeing this a lot: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00018ms SCall: 04050 DCall: 00000

I have attached a patch which adds null packet keepalive functionality to the client. This patch is made against the current CVS tree as of 3/14/01. Please consider this patch for inclusion in the OpenSSH main tree. This patch is based upon and includes code from the Chris Lightfoot (chris at ex-parrot.com) patch posted 2/23. The original patch from Chris is at:

On Tue, Nov 01, 2016 at 05:09:34PM -0700, Jeremy Allison via samba wrote: > On Tue, Nov 01, 2016 at 05:16:47PM -0500, Ed Siefker via samba wrote: > > My expectations are probably wrong, but I had to manually kill a process > > to unlock a file when I think it should have killed itself. > > > > I have deadtime set to 15 and keepalive at default (300). I opened a file on

http://bugzilla.mindrot.org/show_bug.cgi?id=443 Summary: Ability to set KeepAlive time Product: Portable OpenSSH Version: -current Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: ssh AssignedTo: openssh-unix-dev at mindrot.org ReportedBy: danfuzz at

Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where the phone did not send rtp keepalives when on mute (resulting in disconnect from tech support hold and concalls) A side effect seems to be that Asterisk pops the following warning on the console... Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short Grandstream say they are not sure what it is but

http://bugzilla.mindrot.org/show_bug.cgi?id=443 djm at mindrot.org changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |RESOLVED Resolution| |WONTFIX ------- Additional Comments From djm at mindrot.org 2003-01-03 14:58

Hi, I've got a problem with rtp keepalives. I'm using basically the same config on 2 hosts, but one of them sends rtp comfort noise when it's on hold, the other doesn't. The only difference I can think of now is that one of the machines is multihomed, but that might be unrelated. rtpkeepalive is set to 2 and I can confirm is by doing `sip show settings`. I've tried all