similar to: IP-ID in RTP/UDP/IP packets

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

I have tried to set up app_rxfax to receive faxes over IP. I realise there are mixed stories about how reliable this is at the best of times, but at this point all I'm after is some guidance in interpreting the log below. What does "RTP: Received packet with bad UDP checksum" suggest? Here is the full log: -- Executing SetVar("SIP/0892130888-b27c",

HI zytor, I hope you can help me. I have above network chipset on a Mitsubishi im-2000 mobo. I don't see an actual chip on the mobo w/ this ID so I suspect it embedded in the mobo BIOS or in one of the 3 Intel chips. I am attempting to use LTSP to boot a workstation w/ above NIC chipset. It has imbedded boot to net software (Intel Landesk service agt v.99b) using PXE. I can't get it

While working on SPTPS UDP relaying I realized that there is one issue I didn't account for, which is that the sending node only knows the PMTU to the first relay node. It doesn't know the PMTU of the entire relay path beyond the first hop, because the relay nodes don't provide their own PMTU information over the metaprotocol. Now, in the legacy protocol this is not really an issue,

hi, guys. I come from china, a student of BUPT. I have developed a softphone using osip&ilbc. Now, i want to add speex to my softphone. I heard speex have been a daft of rfc, but I just found draft of the format of rtp in ietf's website. Is speex dead? Who can tell me that? Thanks in advanced. yunyong

Thank you very much for your immediate attention! I've exported my wav files to raw files before encoding and the click is now gone. That sure beats discarding the last packet, which was the workaround "solution" I was going to implement... Best regards, David Resnick -----Original Message----- From: Jean-Marc Valin [mailto:Jean-Marc.Valin@USherbrooke.ca] Sent: Thursday, July 07,

Hi all, The problem is on the volume of the voice sent by the SPA-841. I think the echo cancel algorithm sets a limit to the microphone when detects sounds or noise from the earphone. This problem generates an oscillation on the voice volume sent by the phone and even turns it off completely for very little lapses of time making the communication very uncomfortable. I manage three different

Heya, Thanks Rich for the hints on IRC about journal bindings for libguestfs. The below script is what I ended up with. At least I learnt how to iterate over the systemd journal entries. $ sudo python find-ip.py 192.162.122.118 Obviously, this only searches for the _first_ occurance of the IP address and doesn't take into account the guest may have acquired a new IP on a later

Hello, I'm experiencing strange behavior on ssh-enabled file transfers (scp and rsync): When connecting to the Internet through an RTC PPP link to a certain provider, file transfers from a remote host (debian potato) to my local host (debian woody) get immediately stuck when issued from the local host, while they work very well when issued from the remote host. On the other hand, no

SIGHUP (-kHUP) should reload that config for you and SIGURS2 (-kUSR2) will drop currently known subnets (etc) to syslog. This will not work on Windows. Both of these are in the tinc manual (http://tinc-vpn.org/documentation/tinc.pdf). From: tinc [mailto:tinc-bounces at tinc-vpn.org] On Behalf Of Bright Zhao Sent: Saturday, May 6, 2017 9:33 AM To: tinc at tinc-vpn.org Subject: Show the subnets

Hello All I installed shorewall 3.0.8 on Centos 4.3 with openvz.org kernel it work well i have in this Host 3 virtual servers (VPS) i can access from a VPS to the internet , and with NAt rule (Via Shorewall) i can access from Internet to the 3 VPS. i want that all the 3 VPS can communicate between them. i can''t do a tcp connection from a VPS to an other , in my shorewall log in the

Surely you need to patch *all* functions, not just the initial? The point is with the current solution no matter which version of the function another function is linked to, it will hit a sled of JMPs and eventually end up at the newest. If you only patched the first, that sled wouldn't work. So you'd have to patch all instances. That still shouldn't be too hard. Cheers, James

Hi people, As I said before: I did the IDCT to run on the FPGA. My friends from university did the Reconstruction routines running on the FPGA. I'm helping with the LoopFilter, and it is almost there. (all VHDL) I did a small profiling of the libTheora running on a Altera Stratix II device: The processor used was the NIOS II with 8Kb of data and instruction cache, branch prediction and

Pablo Fernandes Yahoo wrote: > Hi Andy and All, > > Thank you so much for your explain. I really apreciated it. I was thiking > about this UDP question and bandwidth shaping... i saw the same udp > connections pass through the shaping of the cable modem and CBQ in a linux > box. As I said I''ve not used CBQ - The LARTC section does say it is hard to configure and may

https://bugzilla.netfilter.org/cgi-bin/bugzilla/show_bug.cgi?id=34 Summary: Redirecting udp packets to closed port gives bad icmp error Product: netfilter/iptables Version: linux-2.4.x Platform: i386 OS/Version: RedHat Linux Status: NEW Severity: normal Priority: P2 Component: ip_tables

https://bugzilla.netfilter.org/cgi-bin/bugzilla/show_bug.cgi?id=34 laforge@netfilter.org changed: What |Removed |Added ---------------------------------------------------------------------------- BugsThisDependsOn| |2 Status|NEW |ASSIGNED ------- Additional Comments From laforge@netfilter.org

We are tracking the following situation: SIP client connects to our Asterisk server, and then connects to another SIP user. Re-invite is OFF, so Asterisk is in the middle of the whole conversation. When one SIP client sends DTMF tones, the SIP client uses RFC2833 to send the tones to the server. (This is correct). The server then sends RFC2833 tones out to the other SIP client. The problem is,

For the life of me can't figure out why the Asterisk server generates an enormous quantity of outgoing packets on port 8004/udp. They seem to have no effect whether they are blocked by the firewall or not. We're running SIP. Everything appears to be OK (except a large number of ChanSpy write buffer overflow messages, which I also don't understand). How can I discover whether these

I have a CentOS release 5.8 that has snmp traps being sent to it. I've been trying to forward the snmp traps to another system. I've tried forwarding with snmpd/snmptrapd, iptables, and some forwarding programs. I can see snmp traps getting delivered to the system with tcpdump and wireshark, but no matter what app I run, the traps do not appear to be reaching the application or port

On Wed, May 13, 2015 at 11:28:23PM +0200, Sven-Haegar Koch wrote: > I have a couple of tinc hosts in the same network, some using the latest > tinc 1.1 git and some using 1.0.24. > > It seems like traffic between 1.1 and 1.0 nodes is always transfered > using TCP (and an intermediate node, if not directly connected), never > with UDP. [...] > # tinc log 5 | grep 2.3.4.5