search for: 3478

Does anyone know if the program stunbdc, which prints one's IP address, is available in CentOS-5? It comes with the libnice package in Fedora-11. ------------------------------ [tim at rose ~]$ stunbdc -4 stun.ekiga.net Server address: 75.101.138.128 port 3478 STUN transaction started (timeout 600ms). STUN unknown: 0 mandatory attribute(s)! Received 88-bytes STUN message No XOR-MAPPED-ADDRESS: 1 Mapped address found! Mapped address: 79.53.131.211 port 44939 ------------------------------ Or is there any other program as simple as this ... ? -- Timot...

I tried to migrate from a server courier to dovecot with imapsync but i get this errore: Couldn't append msg #15 (Subject:[Registrazione]) to folder INBOX: Error sending command '16 APPEND INBOX (\Answered \Seen) "" {3478} ': 16 BAD Invalid internal date. thk you in advance Workino

...failed - 603 declined.....so I can make a call. In Asterisk I debug the channel and I get this log: voip*CLI> debug channel 1 No such channel 1 Debugging on new channels is enabled -- Executing Dial("SIP/user1-08148450", "SIP|user2") in new stack Oct 9 12:52:41 WARNING[3478]: app_dial.c:1024 dial_exec_full: Dial argument takes format (technology/[device:]number1) == Spawn extension (sintys, 1112, 1) exited non-zero on 'SIP/user1-08148450' Oct 9 12:52:41 WARNING[3453]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x81508e8', 10 re...

...bin/echo Not to execute. # Path to stund STUND=/usr/sbin/stund # Set the required args for STUND STUNDPRIMARYHOSTNAME=208.x.x.x # The hostname where another stund server is running on port and alternate # port. STUNDALTERNATEHOSTNAME=127.0.0.1 # The primary response port to user STUNDPRIMARYPORT=3478 # The alternate port to use STUNDALTERNATEPORT=3479 STUNDARGS="-h ${STUNDPRIMARYHOSTNAME} \ -p ${STUNDPRIMARYPORT} \ -a ${STUNDALTERNATEHOSTNAME} \ -o ${STUNDALTERNATEPORT}" Any ideas? Any suggestion for another STUN server? -- Paul

...isk server, it's > *possible* you > might need a true TURN server as well, instead of just an ICE > server. > > Matthew Fredrickson > > > Matthew > > when I set the following in rtp.conf : > > turnaddr=192.158.29.39:3478?transport=udp > <http://192.158.29.39:3478?transport=udp> > turnusername=28224511:1379330808 > turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA > > > then Asterisk 12 gets really slow and sometimes unresponsive. > Calls result in 480 request timeout (possibly...

...as i know, these dont apply because i'm copying the data on the same machine. See the error log below. Is there anywhere else i can check for errors? Also, what's the characters mean in the logfile, such as: >f..tpog.... RAID_One/data/file 8.39M 100% 4.12MB/s 0:00:01 (xfer#3478, to-check=1224/8633) Any help much appreciated. Andrew Logfile: 2008/04/28 20:48:46 [3149] rsync: writefd_unbuffered failed to write 4 bytes [sender]: Broken pipe (32) 2008/04/28 20:48:46 [3149] rsync: connection unexpectedly closed (735103 bytes received so far) [sender] 2008/04/28 20:48:46 [31...

...if the browser you're talking to is > not on the same network as the Asterisk server, it's *possible* you > might need a true TURN server as well, instead of just an ICE server. > > Matthew Fredrickson > Matthew when I set the following in rtp.conf : turnaddr=192.158.29.39:3478?transport=udp turnusername=28224511:1379330808 turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA then Asterisk 12 gets really slow and sometimes unresponsive. Calls result in 480 request timeout (possibly due to the freeze of Asterisk). So this is also no solution. Can not even test if it brings me som...

I start with: v1<-c(1,3,5,7) v2<-c(2,4,6,8) And I want to end up with: v3<-c(12,34,56,78) How do I get there? Thanks, David S. Herzberg, Ph.D. Vice President, Research and Development Western Psychological Services 12031 Wilshire Blvd. Los Angeles, CA 90025-1251 Phone: (310)478-2061 x144 FAX: (310)478-7838 email: davidh@wpspublish.com [[alternative HTML version deleted]]

...no sound either way. Here's what I did on the X-Lite at home in the Topology section: IP address : Discover global address STUN server : Discover server Port used on local computer : Manually specify range 8000-8019 Here are the ports that I forwarded from my NAT router at home: UDP 5060 UDP 3478 (STUN; needed?) UDP 8000 to 8019 Is there something else I should do, either on my home setup or at work on the NAT router or Asterisk? Thank you!

...thing from the softphone. We have used 2 different machines to test this on. We are watching the traffic and noticed that there doesn't appear to be any rtp traffic going back to asterisk (this is where we think the problem might be). The firewalls on both sides have ports 5060, 10000-20000 and 3478 (STUN) open. Out conf files are: ------------------ [sip.conf] *[general] context=incoming ; Default context for incoming calls bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds t...

What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary?

...xport/images/centos51_x86-64,nfsvers=3,tcp,rsize=32768,wsize=32768,sec=sys ip=dhcp BOOT_IMAGE=vmlinuz-2.6.18-custom-2.6.18-53.el5 6. But the /proc/mounts shows that the "sec=" parameter is still set to NULL. /proc/mounts: rootfs / rootfs rw 0 0 /dev/root / nfs rw,vers=3,rsize=3278,wsize=3478,hard,nolock,proto=tcp,timeo=11,retrans=2,sec=null,addr=192.168.16.5 0 0 ............ Kernel versions: PXE server --> uname -a Linux qatest1 2.6.18-53.1.13.el5xen #1 SMP Tue Feb 12 13:33:07 EST 2008 x86_64 x86_64 x86_64 GNU/Linux Diskless Workstation's kernel and root are extracted from...

...r,127.0.0.1): user search: base=ou=Users, dc=mydomain, dc=com, dc=br scope=subtree filter=(&(mail=my.user at mydomain.com.br)) fields=homeDirectory,uidNumber,gidNumber Aug 27 15:03:42 radamante dovecot: auth(default): ldap(my.user at mydomain.com.br,127.0.0.1): result: uidNumber(uidNumber)=3478 gidNumber(gidNumber)=513 homeDirectory(homeDirectory)=/home/samba/my.user Aug 27 15:03:42 radamante dovecot: auth(default): master out: USER#0111#011my.user at mydomain.com.br#011uidNumber=3478#011gidNumber=513#011homeDirectory=/home/samba/my.user Aug 27 15:03:42 radamante dovecot: pop3-login: Lo...

...log. requests are coming in from SipGates server etc but my asterisk box is not transfering the calls to the phones. I have the register string in my sip.conf as so: register=6698221:(MYSECRET)@sipgate.co.uk/6698221 Port on my IPCOP box as follows: UDP/5060 UDP/10000:20000 UDP/8000:8012 UDP-TCP/3478 Thanks for your time. Paul. -------------- next part -------------- Sip read: INVITE sip:6698221@MY_ISP_IP:5060 SIP/2.0 Record-Route: <sip:6698221@217.10.79.219;ftag=as6a04ebdf;lr=on> Max-Forwards: 9 Record-Route: <sip:6698221@217.10.79.8;ftag=as6a04ebdf;lr=on> Via: SIP/2.0/UDP 217....

I am moving some messages from an IMAP account to another. During the process I got this message: dovecot: IMAP(mailstore): Disconnected: EOF while appending bytes=263276/427306 (no core dump) I deleted dovecot.index.cache and dovecot.index.log and the error disappeared Ciao, luigi -- / +--[Luigi Rosa]-- \ Committee: a life form with six or more legs and no brain.

Scott said: >> I''d like to output my rcov data to the screen in ascii instead of >> html...is that possible using the rake spec:rcov command? > Look at the RSpec gem''s rdoc. I believe it is something like > Rake::RcovTask. An option can be provided to give rcov options. Run > rcov --help. This told me I needed the -T option to get what I wanted.

...e = no nat = yes username = 8703405315 secret = xxxx context = from-sip-provider ----- With 'sip debug' on, I can see it sending the REGISTER requests and getting back a response with STUN headers like so (also edited): ----- SIP/2.0 407 Proxy Authorization Required X-Stun-Server: w.x.y.z:3478 X-Observed-Adr: a.b.c.d ... ----- However, when Asterisk sends the auth it doesn't sends the REGISTER again to the same address without seeming to take into account the STUN details, a la: ----- REGISTER sip:sip-provider.not SIP/2.0 Via: SIP/2.0/UDP 10.20.15.4:5060;branch=z9hG4bK43e3ead5 ......

Hello all, I have a new Sipura SPA-2000 that I am trying to configure beind a NAT. The SPA is able to register to the asterisk server without a problem and the SPA can make calls to other extension that are not behind a NAT. However, when I try to call the SPA from another extension, the extension connected to the SPA rings, the user at the SPA answers, and there is no audio in either

...Recently I decided to test the same sip.conf with another VSP (SIPphone) and it worked fine! No issues on the reinvite. Note: clients and server using ULAW (only), no NAT or firewalls, public ip address and no extra parameter on the dialplan. I even tried to use STUN server at FWD (stun.fwdnet.net:3478) with no luck at all. The only option available on the market for me at the moment is Broadvoice because of an unlimited international plan with a flat rate, so I'm stuck here. Please, send me ideas. Thanks in advance, -Dhennys Pestana ### sip.conf [general] context=default recordhisto...

...e maxexpiry=50 defaultexpiry=30 register => user:passw@sip.voipstunt.com/user [useruser] type=friend username=user secret=passw host=sip.voipstunt.com fromdomain=sip.voipstunt.com canreinvite=yes insecure=very nat=yes context=incomingsip.voipstunt.com dtmfmode=rfc2833 stun=stun.voipstunt.com:3478 [13] type=friend username=13 secret=13 callerid="13" <13> host=dynamic mailbox=13@default dtmfmode=rfc2833 canreinvite=yes context=13 --------------------------------------------------------------------------------------- my extensions.conf [general] static=yes writeprotect=no...