similar to: Secure audio cannot be provided

Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

Hi, I have installed Asterisk 1.4 with mISDN with the install-asterisk.tar.gz script from beronet.com. On my system I have two cards, one a AVM Frit!Card Pci 2.0 and one HFC-S chip. I know both to work well with mISDN on my system from a previous installation. Now however, the AVM card works well at first glance, i.e. it "registers" incoming calls and works through the asterisk

Hi, I'm trying a VPN via ssh between Linux (Suse 10.3, 5.0) and OpenBSD (4.2, 4.7). But it doesn't work... On Linux: sudo ssh -v -f -w 0:1 $OPENBSD true sudo /sbin/ifconfig tun0 10.1.1.1 10.1.1.2 sudo /sbin/route add -net 192.168.17.0 netmask 255.255.255.0 dev tun0 On OBSD: sudo ifconfig tun1 10.1.1.2 10.1.1.1 netmask 255.255.255.0 sudo /sbin/route add 10.0.3.0/24 10.1.1.1 And add a

Hi, I have done my best and tired of searching the net about the problem. If anybody could help would be a great favour. Description of Problem ------------------------ I am trying to install two Netpci cards(Traverse Technology Netjet ISDN-s) on Trixbox 2 and aim is to use in Asterisk as dailin and dialout. I compliled the driver as directed in the manufacture manual. After installation dmesg

Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context |

Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context | secret |

Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context |

Hello. I am trying to get multi paths running, i.e I want to specify 2 paths from a particular node to another. To test this, I am setting up the following scenario: I create 2 tap devices tap0 = 10.1.1.1/32 netmask 255.255.255.0 tap1 = 10.2.2.2/32 netmask 255.255.255.0 I then issue the route command: ip route add 10.3.3.3/32 nexthop via 10.1.1.2 nexthop via 10.2.2.3 The route table

Hello, I was testing with sdp and something came up worth asking: While calling from a webrtc client to another (chrome, sip.js) Asterisk receives the following sdp and rejects it with 488 Not Acceptable. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine,

I have * box on machine with external ip address and internal one I'm tring to register to it from to machines - one from innternet ( everything is ok - in sip.conf nat=yes)\ and the other one is in the internal network (in sip.conf - nat=no ) and it say 403 Forbidden? Any Ideas ? here are the logs and configs From the external SIP Client whic registers.

Can anyone work out why my callerid doesn't work on my isdn4Linux with asterisk (or without asterisk for that matter)... This used to work fine, and I am quite confident that the telco is sending callerid information (because they always do on all ISDN lines standard, only extra cost on POTS lines). This is the information from dmesg, whether asterisk is running or not: isdn_net: Incoming

I have a lan with shorewall running as firewall and two local machines, where 10.1.1.2 and 10.1.1.15 are two internal mail servers and where 124.124.124.124 and 123.123.123.123 are the external IPs for the mail servers. The two mail servers need to communicate with each other via smtp (for sending mail from domains hosted on one to the other) but its giving issues. Specificaly when one server

Dear list, I am having a problem with openvpn. I have the following arrangement, running two instances of openvpn on "home fw". I want to protect my WLAN in back of the home fw and that works fine. I can see "Peer connection initiated with 192.168.1.3:5000" in daemon.log on homefw. Nothing gets initiated with officefw, nor can I ping the other end of the tunnel at officefw.

As per my problem yesterday with the Cisco 7960 and getting it talking to Asterisk on a different subnet, I gave up trying and just put the Asterisk box back on the internal subnet. However, I made two changes: - the external IP address is set on an ethernet alias eth0:0 - the main Linux router will change outgoing requests from 10.1.1.2 to the external IP (rather than the default behaviour of

https://bugs.freedesktop.org/show_bug.cgi?id=49295 Bug #: 49295 Summary: artifacts in prboom on NV86 Classification: Unclassified Product: Mesa Version: git Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Drivers/DRI/nouveau

Help or suggestions please: When using smbclient to handle printing from a AIX system to a windows print queue somethings work and others do not. I have been able to print without issues to the windows print queue. But I cannot get a listing of the print jobs in a print queue. This is important so the UNIX users can take action on thier print jobs. I'm not sure which why to turn at this

Recently I wrote about this problem, but it still exist and I can't dial my Xlite SIP Phone So here is the Notice Jul 5 17:14:07 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'Damian Minkov <sip:damian@10.1.1.2>' failed for '10.1.1.11' The * box(10.1.1.2) and the PC(10.1.1.11) on which is the XLite are in the same network Here is part from sip

Hello everybody, I installed eucalyptus on ubuntu10, basic configuration . Everything is ok untill I start the instances, the disk is being created but it terminates soon. Environment: NFS , KVM, ubuntu10 x64, Euca on 2 machines (wc sc cc clc on one machine, and nc on another). libvirtd (libvirt) 0.7.5 QEMU PC emulator version 0.12.3 (qemu-kvm-0.12.3), Copyright (c) 2003-2008 Fabrice Bellard.

I have multihome network line. I want to use multi gateway, but i have some trouble +-----------------------+ | 30.1.1.1 (eth0) | <----> 10.1.1.1 ( eth1) < ---> internet | | | 10.1.1.2 (eth1) | | | <----> 20.1.1.1 (eth2) < ---> internet | 20.1.1.2