Re: [asterisk-users] Asterisk 1.8 and dual stack support
Jaap Winius, 21.03.2013 17:47: support IPv6. However, it seems that I can't get it to support both IPv4 and IPv6 at the same time. For example, if in sip.conf I set the bindaddr variable to '::' it will only listen on IPv6 and none of my IPv4-only friends and peers will be able to connect to it. On the other hand, if I set it to '0.0.0.0' then it will not listen on IPv6. This is well explained here: http://serverfault.com/a/39561 In short: In Linux, binding to :: means bind to both ipv6 and ipv4. Setting /proc/sys/net/ipv6/bindv6only to 1 changes this behaviour, and Debian has this by default (since squeeze, AFAIK). So this is a system issue, not an Asterisk. At least unless one considers Asterisk's shortcoming of not being able to use more than one socket an issue. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF inband with telephone-event in SDP
Hello everyone! We use Asterisk for various services like voicemail. Our SIP clients usually use rtp events (rfc2833) for DTMF, which works just fine and independent from the codec (g711 vs. g726 etc.). Now we noticed there are some SIP clients that announce telephone-event in their SDP, but send their DTMF inband. The problem with that is, that Asterisk obviously does not try to detect inband DTMF after seeing the telephone-event payload type in the SDP. So we are in a kind of dilemma: - dtmfmode=auto (and dtmfmode=rfc2833) will work for most, but not for the described ones. - dtmfmode=inband would also work for most, but of course not for the ones using g726 et al. Is there any Asterisk setting to force inband DTMF detection (with non-compressing codecs only, of course)? I browsed the code without result. Does anybody have a hint how to handle this? Or if the SIP clients behaviour is even RFC compliant? Regards and TIA, Jakob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP address of remote SIP host
Hi, is it possible to get the SIP IP address of the remote (calling) party, in the dialplan or (preferrably) in an AGI script? (This sounded like a rather basic question to me, but I could not find an answer...) TIA regards Jakob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding to 0.0.0.0 a security risk?
Raj Mathur (राज माथुर), 2012-02-08 03:27: Packets not going out on the same interface as the one they were received on is a general IP issue, not just for connectionless Right, this was a inaccuracy. It should say Asterisk does not reply with the IP address with which packets were received. Asterisk (as most applications) does not care about network interfaces, it just handles IP addresses. protocols. The same behaviour can be seen with TCP too. Unless you mangle with iptables or something, all information about the received A tcp connection is defined by the tuple (source hostport, destination hostport), so if you write to a tcp socket, the kernel knows which source address it has to use (and also which destination address, so the application doesn't need to know that at all). As there's no such relation in udp, the application has to provide the destination address. The kernel then decides which source address to use, as long as the application did not bind() to a specific address. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding to 0.0.0.0 a security risk?
Steve Edwards, 2012-02-06 01:43: Unfortunately, (IIRC) Asterisk does not reply to the same interface packets are received from which limits the usefulness of multiple interfaces. Right, that's what I also observed. We had to take special measures to handle this. The problem lies in the nature of connectionless protocols as UDP. We also use freeradius, which does it right by itself (but still needs a compile time switch --with-udpfromto for it). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding asterisk to two static IPs
On 12.10.2011 23:27, ge...@riseup.net wrote: If you put 0.0.0.0, it will bind to all addresses. In a HA Cluster, on the active node, if you have a box address of 192.168.1.101 and a floating address of 192.168.1.102, then if you use bindaddr=0.0.0.0 ... Any idea how to solve this? Yes: Use bindaddr=192.168.1.102. That's how we solved it on our Asterisk boxes. Another solution would be to use tcp, but not all SIP clients support that (and I don't know how good Asterisk does). Personally, I think this is a shortcoming in Asterisk. Every application with udp server functionality should handle this correctly. E.g. FreeRADIUS has a compile time option for this (--with-udpfromto, unfortunately off by default, for whatever reasons). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding asterisk to two static IPs
On 13.10.2011 00:27, ge...@riseup.net wrote: If I use the floating internal ip, I can't reach my provider anymore. Thought this was clear. After reading your original message, this is clear, yes. Sorry for being sloppy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
