similar to: Uverse, Asterisk and SIP

AHOI! I've big trouble by setting up CentOS 6.3 for my new vDSL. As I've found out, the German Telekom is using VLAN7 for her internet-(data) connections. => http://workaround.org/blog/vdsl O.K. what I've done: The NIC where's my DSL-modem is connectet is eth0. cat /etc/sysconfig/network-scripts/ifcfg-eth0 # device for vDSL-modem DEVICE=eth0 HWADDR=00:30:1B:14:08:67

I was given the challenge recently of creating a LAN-LAN bridge between two buildings several hundred metres from each other, using only existing Cat 3 wiring and without having to resort to an expensive and finicky 5 Ghz wireless link. I was able to create a 90 megabit link for about $3,000 Cdn with new PC's, CentOS 4.1, and the newly avaliable Black Box VDSL Ethernet Extender, which

Hello, CentOS 8 dropped the rp-pppoe package. My attempts with NetworkManager-ppp already failed because it seems that it is not possible to use a VLAN as a device. Please, how does it work under C8 to set up a pppoe connection (German Telekom, VDSL (VLAN 7 required), ZTE modem). Thanks. Joe

Hi there, we are using tinc in switched mode for over a year now, currently with 18 clients which are connected 24 hours a day and many which aren't connected the whole day, also. If i'm reading the changes from 1.0.9 to 1.0.10 and 1.0.11 correctly, tinc should work now, although "TCPOnly = yes" isn't set in the config files of clients which are behind a NAT firewall, e.g. a

Hi all, I have an issue with asterisk 13 and pjsip. I guess it is somehow Firewall related, but I'm unsure. A registration to Sipgate is established successfully: <Registration/ServerURI..............................> <Auth..........> <Status.......> ==========================================================================================

I have to provision several dozen * users to a seperate building on our campus in the same subnet. Ordinarily, I'd just run a gigabit cat6 cable to another switch if it doesn't violate the 100 metre rule, but this building is several hundred metres away from my backbone. My only option for cabling to the remote building is copper. My plan is to provision them with a Linux bridge with 4

Hi, Thank you so much for your reply, I've a dedicated server in OVH, where I have done speed test for the server : *bkf at xxxxx ~> speedtest-cli Retrieving speedtest.net <http://speedtest.net> configuration...Retrieving speedtest.net <http://speedtest.net> server list...Testing from OVH SAS (x.x.x.x)...Selecting best server based on latency...Hosted by fdcservers.net

HI! I'm currently reviewing my (functional) setup with one physical ethernet interface eth1 of the host system being connected to a VDSL modem (not router!). A fully virtualized guest acts as router with this source mode: <source dev='eth1' mode='bridge'/> Reading the docs [1] I'm thinking whether mode='private' would be a better choice because the network

Hi all, First of all, thank you for supporting us. I've installed Libretime (https://github.com/LibreTime/libretime) which user Icecast2 on Ubuntu Server 16.04. I've also developed a mobile application to listen to stream myhost:8000/mount. My problem is when the number of listeners increases up 500 (On live streaming with butt), the application blockes, and I hardly get info from

In SIP setting on the e71 I set the public user name as 1995 at 10.10.11.180. There is a sip.conf context [1995] On the asterisk CLI I get: Registration from '<sip:%201995 at 10.10.11.180:5060>' failed for '10.10.11.98' - No matching peer found So I changed the sip.conf context to [%201995] Then: [2009-11-19 20:44:28] WARNING[14371]: chan_sip.c:11797 check_auth:

ekiga registration fails. I've set nat = yes ( also blank ) and i've set externip. Anybody have a sip.conf that works? Here's the sip debug: Reliably Transmitting (NAT) to 86.64.162.35:5060: REGISTER sip:ekiga.net SIP/2.0 Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport Max-Forwards: 70 From: <sip:test at ekiga.net>;tag=as64618445 To: <sip:test at

Hello; To be able to send and receive faxes through asterisk and to be able to have trunk with google voice and to be able to have integration with those that support Jaber .. What are the channels and libs that I have to be sure that they are existed? Regards Bilal

Hey, all. I'm in the middle of a rollout, and just learned that the SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued. The heir apparent is the SoundPoint IP 450 -- for a low, low, low $130 more/handset. AND it doesn't look as nice. Ouch. Does anyone have any recommendations -- Polycom or otherwise -- for a good-quality, mid-range, two-line SIP phone (with good

Has anyone had good results with an on-line database that returns a LATA based on NPA NXX? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100518/d77d09b3/attachment.htm

Im looking for wifi sip phones that support auto provisioning and work flawlessly with atserisk. Can anyone suggest me some models?

Just finished with the instructions from digium website/ net on how to compile FFA: After restart, modules did not get loaded so tried to load manually: [Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module: Error loadin ile: No such file or directory [Dec 18 14:31:26] WARNING[11002]: loader.c:653 load_resource: Module 'res_fax.so Verified the files exist: astbh00*CLI>

Does Asterisk know when it hits a voicemailbox ? When calling to a cell-phone or GSM, after some rings and no pickup you arrive at a voicemailbox. If Asterisk does not know it's a voicemailbox that has answered the call, the voicemailbox will contain 60minutes of 'silence'. This is very expensive 'silence'. How to avoid this ? Jonas -------------- next part --------------

I'm currently receiving over 200 SIP REGISTER requests per second from a machine apparently in Italy, host97-239-149-62.serverdedicati.aruba.it. This has continued for several days, and abuse at staff.aruba.it are unresponsive. I've had a couple of similar incidents recently, the others originating from uk2.net. I have an ADSL connection and responding to these REGISTERS was consuming all

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 ==

I have a situation where we have an asterisk box that is extending several Mitel PBX extensions to some cordless SIP phones (Cisco WIP310). Everything works great, except when the cordless phone walks out of range of one access point and into range of another (cisco 1100 series APs). I've been able to get virtually seamless roaming between access points to work in the past with data but