I need to ask few question about Asterisk
1) Does ACL mean by Access Control list here ?If yes than how could i use it?
>ip show user 6001
* Name : 6001
Secret : <Set>
MD5Secret : <Not set>
Context : DLPN_Admin
Language :
AMA flags : Unknown
Transfer mode: open
MaxCallBR : 384 kbps
CallingPres : Presentation Allowed, Not Screened
Call limit : 2147483647
Callgroup : 1
Pickupgroup : 1
Callerid : "test" <6001>
ACL : No
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Sess-Min-SE : 90 secs
RTP Engine : asterisk
Codec Order : (ulaw:20,gsm:20)
Auto-Framing: No
2) What is mean by "Require Call Token" in Asterisk Digium GIU on Create new User Panel
3) Is There any command from where i can get users VOICE MAIL password ?
4) What AMI or CLI command set call recording on or off for user ? and if i want that file to be stored on client computer not on server memory what could i do ?
Question 1:
Yes, ACL does stand for Access Control List. You can use the settings "contextpermit/contactdeny" to control what addresses a UA can register from; "permit/deny" to control what addresses a UA can establish calls from (INVITE request); and "directmediapermit/directmediadeny" to control what addresses a UA can use to set up direct media between UAs. Note that all of this is in the sample sip.conf, delivered with Asterisk.
Question 2:
Call Token refers to the IAX setting "requirecalltoken". Older Asterisk clients (1.2 before 1.2.35) don't support call tokens. Note that call tokens were added to address a security vulnerability (AST-2009-006). From the AST notification:
"A lot of time was spent trying to come up with a way to resolve this issue in a way that was completely backwards compatible. However, the final resolution ended up requiring a modification to the IAX2 protocol. This modification is referred to as call token validation. Call token validation is used as a handshake before call numbers are assigned to IAX2 connections.
Call token validation by itself does not resolve the issue. However, it does allow an IAX2 server to validate that the source of the messages has not been spoofed. In addition to call token validation, Asterisk now also has the ability to limit the amount of call numbers assigned to a given remote IP address.
The combination of call token validation and call number allocation limits is used to mitigate this denial of service issue."
Question 3:
No. That doesn't mean you couldn't use AGI to call out to a script with the user's voicemail extension, do the parsing yourself, and put the result in a channel variable.
Question 4:
AMI commands are documented at Asterisk AMI Actions. I'm going to assume that by "set recording" you mean start a Monitor application on some particular channel (and not change CDRs, CELs, etc.) In that case, you'd use the Monitor AMI action to start the recording, and StopMonitor AMI action to stop the recording. Once the file is created, you can move it off the server yourself using AGI or some other externally spawned mechanism.
Related
We're using DPDK (version 20.08 on ubuntu 20.04, c++ application) to receive UDP packets with a high throughput (>2 Mpps). We use a Mellanox ConnectX-5 NIC (and a Mellanox ConnectX-3 in an older system, would be great if the solution worked there aswell).
Contrary, since we only need to send a few configuration messages, we send messages through the default network stack. This way, we can use lots of readily available tools to send configuration messages; however, since all the received data is consumed by DPDK, these tools do not get back any messages.
The most prominent issue arises with ARP negotiation: the host tries to resolve addresses, the clients also do respond properly, however, these responses are all consumed by DPDK such that the host cannot resolve the addresses and refuses to send the actual UDP packets.
Our idea would be to filter out the high throughput packets on our application and somehow "forward" everything else (e.g. ARP responses) to the default network stack. Does DPDK have a built-in solution for that? I unfortunatelly coulnd't find anything in the examples.
I've recently heard about the packet function which allows to inject packets into SOCK_DGRAM sockets which may be a possible solution. I also couldn't find a sample implementation for our use-case, though. Any help is greatly appreciated.
Theoretically, if the NIC in question supports the embedded switch feature, it should be possible to intercept the packets of interest in the hardware and redirect them to a virtual function (VF) associated with the physical function (PF), with the PF itself receiving everything else.
The user configures SR-IOV feature on the NIC / host as well as virtualisation support;
For a given NIC PF, the user adds a VF and binds it to the corresponding Linux driver;
The DPDK application is run with the PF ethdev and a representor ethdev for the VF;
To handle the packets in question, the application adds the corresponding flow rules.
The PF (ethdev 0) and the VF representor (ethdev 1) have to be explicitly specified by the corresponding EAL argument in the application: -a [pci:dbdf],representor=vf0.
As for the flow rules, there should be a pair of such.
The first rule's components are as follows:
Attribute transfer (demands that matching packets be handled in the embedded switch);
Pattern item REPRESENTED_PORT with port_id = 0 (instructs the NIC to intercept packets coming to the embedded switch from the network port represented by the PF ethdev);
Pattern items matching on network headers (these provide narrower match criteria);
Action REPRESENTED_PORT with port_id = 1 (redirects packets to the VF).
In the second rule, item REPRESENTED_PORT has port_id = 1, and action REPRESENTED_PORT has port_id = 0 (that is, this rule is inverse). Everything else should remain the same.
It is important to note that some drivers do not support item REPRESENTED_PORT at the moment. Instead, they expect that the rules be added via the corresponding ethdevs. This way, for the provided example: the first rule goes to ethdev 0, the second one goes to ethdev 1.
As per the OP update, the adapter in question might indeed support the embedded switch feature. However, as noted above, item REPRESENTED_PORT might not be supported. The rules should be inserted via specific ethdevs. Also, one more attribute, ingress, might need to be specified.
In order to check whether this scheme works, one should be able to deploy a VF (as described above) and run testpmd with the aforementioned EAL argument. In the command line of the application, the two flow rules can be tested as follows:
flow create 0 ingress transfer pattern eth type is 0x0806 / end actions represented_port ethdev_port_id 1 / end
flow create 1 ingress transfer pattern eth type is 0x0806 / end actions represented_port ethdev_port_id 0 / end
Once done, that should pass ARP packets to the VF (thus, to the network interface) in question. The rest of packets should be seen by testpmd in active forwarding mode (start command).
NOTE: it is recommended to switch to the most recent DPDK release.
For the current use case, the best option is to make use of DPDK TAP PMD (which is part of LINUX DPDK). You can use Software or Hardware to filter the specific packets then sent it desired TAP interface.
A simple example to demonstrate the same would be making use DPDK skeleton example.
build the DPDK example via cd [root folder]/example/skeleton; make static
pass the desired Physical DPDK PMD NIC using DPDK eal options ./build/basicfwd -l 1 -w [pcie id of DPDK NIC] --vdev=net_tap0;iface=dpdkTap
In second terminal execute ifconfig dpdkTap 0.0.0.0 promisc up
Use tpcudmp to capture Ingress and Egress packets using tcpdump -eni dpdkTap -Q in and tcpdump -enu dpdkTap -Q out respectively.
Note: you can configure ip address, setup TC on dpdkTap. Also you can run your custom socket programs too. You do not need to invest time on TLDP, ANS, VPP as per your requirement you just need an mechanism to inject and receive packet from Kernel network stack.
I've been using a sim900 module to replicate many of the functions found in a basic cellphones for an embedded project. I've been successful with most functions with the exception of entering options during a voice call. I am actually looking for a generic solution (e.g. GSM 07.07 etc.) although the GSM/GPRS Module I'm using is the sim900.
Scenario: I initiate a call using ATD<number>; ,then automated voice asks me to dial "1" for an option. How do I send the "1"?
I've search high and low for an answer. I've been through the AT command manual over and over again. Please help.
Very good start in using the official GSM specification, although I want to note that 07.07 has been superseded by 27.007 a very long time ago, and you should use that document (or 27.005 if relevant).
I initially expected there to be two ways of achieving this, dial string modifiers or DTMF command, but I looking up the dial string in the 27.007 specification I do not find the p (pause) modifier1 I was expecting, and nearest thing, the W (wait) modifier is ignored and only included for compatibility.
Sending 0 through 9, A through D, * and # during a call is done using DTMF, although in a GSM network this is signalled separately out of band rather than sending in-band analogue tones. There is a specific command for sending DTMF tones called AT+VTS (with a horrible syntax). So that command is the answer to you question. Invoke it either from online command mode or from another serial connection.
1 The reason I was expecting a p modifier to exist is that I am able to enter one in phone book entries, e.g. `"12345678p123" which will dial 12345678, wait for the connection to be established and then send 123 as DTMF tones. But this is then obviously something (only) the user interface voice call handler manages and not the AT command handler.
I'm developing a product that need to integrate with RADIUS server as an authentication method.
When configuring the RADIUS server (IP Address, Port, Shared Secret) I would like to do a "test" in order to check that the configuration is valid - The server is available and it is indeed a RADIUS server, Shared secret is OK.
I did some research on how to do it,
My options are:
Send Access-Request message with fictional user name and password to the RADIUS server
Send Status-Server message to the RADIUS server
RFC 5997 introduces the use of Status-Server Packets in the RADIUS protocol.
This packet extension enabling clients to query the status of a RADIUS server.
The Status-Server is marked as experimental and as Informational RFC rather than as a Standards-Track RFC
My questions are:
Which are the most common \ in use RADIUS server vendors ? MS NPS, FreeRADIUS, Other?
Are these vendors supporting Status-Server request - Do they implementing this packet type ?
If i will use Access-Request, I will receive "Access-Reject" with a failure message in "Reply-Message" attribute. Can i understand the reason for the refusal from that text message? Is there any list of error codes\messages that are part of the Standard ?
Thanks a lot,
Yossi Zrahia
Ad 1) Exact (or even estimate) numbers are hard to come by, but you should expect to encounter FreeRADIUS, Microsoft NPS, Radiator and maybe Cisco ACS/ISE.
Ad 2) FreeRADIUS, Radiator support it. Microsoft NPS and Cisco ACS/ISE do not. If your "test" is used once (upon configuring) I would use option 1 with the Access-Request. If you wish to periodically check the availability and configuration of a RADIUS server, I would suggest implementing both options and allow for configuration of the check as part of the RADIUS configuration:
IP: 1.2.3.4
Port: 1812
Shared Secret: U7tr453cur3
Servercheck: [x] Status-Server
[ ] Access-Request
Ad 3) From RFC2865, section 5.18 (Reply-Message):
"[...] This Attribute indicates text which MAY be displayed to the user. [...] When used in an Access-Reject, it is the failure message. It MAY indicate a dialog message to prompt the user before another Access-Request attempt. [...] The Text field is one or more octets, and its contents are implementation dependent. It is intended to be human readable, and MUST NOT affect operation of the protocol. It is recommended that the message contain UTF-8 encoded 10646 [7] characters."
There apparently are no standard messages specified; however if IP, Port or Shared Secret are configured incorrectly you should not get a response at all, because RFC 2865 specifies:
"A request from a client for which the RADIUS server does not have a shared secret MUST be silently discarded."
I need to strip the rport parameter from the Via field of SIP messages being generated by Yate (for compatibility with a broken peer). Can I use the scripting capabilities of Yate to do this? How do I intercept and modify outgoing SIP messages?
I discovered, by reading the code, that Yate already supports this feature. In accfile.conf, in the section defining the sip server to register to, place the following line:
xsip_flags=1
This prevents the rport parameter from being placed in SIP messages. This may break routing if you are behind a NAT, so beware.
An example config would therefore be:
[sip_service]
enabled=yes
protocol=sip
description=sip_service
username=user
domain=somewhere.com
authname=auth
password=secret
server=somewhere.com
xsip_flags=1
I have added International Domain Name support to an XMPP client as specified in RFC 6122. In the RFC it states:
Although XMPP applications do not communicate the output of the
ToASCII operation (called an "ACE label") over the wire, it MUST be
possible to apply that operation without failing to each
internationalized label.
However, with the domain I have available for testing (running Prosody 0.9.4; working on getting feedback from someone else about how Ejabberd handles this), sending a Unicode name in the "to" field of an XMPP stanza causes them to immediately return an XMPP error stanza and terminate the stream. If I apply the toASCII operation before sending the stanza, the connection succeedes, and I can begin authentication with the server.
So sending:
<somestanza to="éxample.net"/>
Would cause an error, while:
<somestanza to="xn--xample-9ua.net"/>
works fine.
Is it correct to send the ASCII representation (ACE label) of the domain like this? If so, what does the spec mean when it says that "XMPP applications do not communicate the output of the ToASCII operation ... over the wire"? If not, how can I ensure compatibility with misbehaving servers?