similar to: virt-manager connect remote KVM host and graphic control guest problem

Hi George, Thank you for the response. I'm a little unclear on what you mean by a transport. We're using chan_sip, not pjsip. Do you mean a device in sip.conf, using bindaddr to set the address to bind for that device? We've only used bindaddr in the [general] section before, but if it will work in a device that could be the answer. On Fri, 23 Oct 2020 at 00:13, George Joseph

I'm using Asterisk on a open server (no firewall or NAT) and trying to communicate with a Grandstream BudgeTone 102 SIP phone which is behind NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS about a week ago. My problem is that I'm only getting half-duplex communication -- I can hear voice from the Asterisk server but the server does not understand any voice from

Hello, We have an Asterisk server with two public IP addresses, let's say 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call dialled from Asterisk to an external destination. The external destination sees the SIP packet as coming from 1.1.1.1 and the media address in the SDP is 1.1.1.1, which is great. However if we receive a call in to 2.2.2.2 then the call

thank you very much. this is exactly whats needed for debug example output for your info [Dec 12 15:39:19] DEBUG[2182][C-00000000]: pjproject: <?>:         icess0x7f5d44081e88 .Added new remote candidate from the request: 2.2.2.2:57536 [Dec 12 15:39:19] DEBUG[2182][C-00000000]: pjproject: <?>:         icess0x7f5d44081e88 .New triggered check added: 1 [Dec 12 15:39:19]

OK, thank you George. On Sat, 24 Oct 2020 at 03:16, George Joseph <gjoseph at digium.com> wrote: > > > On Thu, Oct 22, 2020 at 4:13 PM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hi George, >> >> Thank you for the response. I'm a little unclear on what you mean by a >> transport. We're using chan_sip, not pjsip.

Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio On Thu, Oct 29, 2020 at 20:44 David Cunningham <dcunningham at voisonics.com> wrote: > Hello, > > Does anyone know a way with chan_sip to tell Asterisk to use a specific IP > address for its end of the communication for a specific

Asterisk is on public IP (as described in the first email) i have 10 years experience in voip, 4 years webrtc in production. i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism but i confess. i dont understand WHY Asterisk SOMETIMES switches destination IP in RTP. this is not only about ICE. its about RTP engine too which is Asterisk specific and Asterisk DEBUG is

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN

Hello, On my systems i use shorewall 3.2.6. Now all systems where replace by new ones with new ip''s. So i tried with DNAT to map the old ip''s to the new one as long as DNS is updated. But i didn''t get it work. I see in tcpdump that a connect from client-ip to new-server-ip is done while connection the old on. But i get no response. Did i configure something in the

Hello, I have been reading through the documentation and trying to set up a very small VPN as a test for a larger rollout that I would like to complete in the future but cannot get this working. The configuration seems like it should be relatively simple, so I'm most likely missing something basic but I just cannot see what I'm doing wrong. At the moment I am trying to get this working

I am trying to get a tunnel from a cisco 1760 with IOS 12.2.15.t13 to a freebsd 4.9 install with racoon. I have package version freebsd-20040408a and internal version 20001216 in my log file. I posted the full racoon and cisco log below my configs. Racoon keeps saying: 2004-07-26 16:24:03: DEBUG: isakmp.c:2295:isakmp_printpacket(): begin. 2004-07-26 16:24:03: DEBUG:

Hi. I'm using nologin with own reason [1]. That works fine. For example pop3 client gets nice message like "-ERR [AUTH] Account is locked. Please contact support." Unfortunately maillog lacks information details about why user was not allowed to log in. pop3-login: Disconnected (auth failed, 1 attempts in 2 secs): user=<testuser>, method=LOGIN, rip=1.1.1.1, lip=2.2.2.2,

hi After compiling and installing pwlib and openh323 , the asterisk, give the folloing error. please tell me where the problem is ? Best Mani *CLI> -- Executing Dial("SIP/2.2.2.2-086f5ac0", "H323/652#150388590962@1.1.1.1|60") in new stack Mar 28 14:17:23 WARNING[11985]: channel.c:2576 ast_request: No translator path exists for channel type H323 (native 4) to 256 Mar 28

https://bugzilla.netfilter.org/show_bug.cgi?id=887 Summary: iptables.xslt wrong "match" -m handling Product: iptables Version: 1.4.x Platform: All OS/Version: Debian GNU/Linux Status: NEW Severity: major Priority: P5 Component: unknown AssignedTo: netfilter-buglog at lists.netfilter.org

Is there a way to log SNI hostname used in TLS session? Info is there in SSL_CTX_set_tlsext_servername_callback, dovecot copies it to ssl_io->host. Unfortunately I don't see it expanded to any variables ( http://wiki.dovecot.org/Variables ). Please consider this to be a feature request. The goal is to be able to see which hostname client used like: May 30 08:21:19 xxx dovecot:

Sorry if this is the wrong place to post, I'm not a developer. OpenSSH seems to be capable of binding to any given list of IP addresses, on any given list of ports. What it can't seem to do, however, is be given any combination of these. Lets say I want to bind to 1.1.1.1:22 as well as 2.2.2.2:80. (I choose 80 knowing it is the http port, just for that purpose, because some people are

Hello, I'm trying to get Dovecot to use GSSAPI for authentication. I have an IPA server on CentOS 7 with a bunch of my servers attached to the IPA domain, including the server running Dovecot. I've followed official documentation from Red Hat and read numerous wiki articles on how to configure Dovecot to get it to use GSSAPI correctly. I don't think I've done anything

Hi list! I'm trying to get a DID routed to me and the provider seems to have an unusual setup. Or maybe not? From looking at their SIP header they are using "BroadWorks". The problem: they're sending their SIP invite from port 36252. My Asterisk 10.7.1 is answering to that port 36252 but their BroadWorks thingie is not listening on that port, but instead on port 5060. So

Hi all, I am currently deploying tinc as an alternative to OpenVPN. My setup includes a lot of nodes and some of them are sitting together behind the same router on the same network segment. (E.g. connected to the same switch.) I noticed, that those nodes do never talk directly to each other via their private ip-addresses, but instead use the NATed address they got from the router.

Hi! I'm trying to migrate a mailbox from an old IMAP server (not Dovecot) to a new Dovecot server with 'doveadm backup'. The command fails because of an unexpected character in a mailbox name: doveadm -o imapc_user=ag at example.org \ -o imapc_password=XXX \ -o imapc_host=old-mailserver.webflow.de \ -o imapc_port=993 \ -o imapc_ssl=imaps \