similar to: SIP and symmetric NAT

hi all, I known taht this ML is about *, but lot of you are using BT telepnones. I'm using FW 1.0.6.7 for all phone. This firmware support VLAN tagging for QoS on the Layer2. I use it to separe the PHONE network form the PC network, which are in PC connector in the BT. And I also use the TFTP server for provisioning (software, configuration). The problem is, if you set the VLAN for QOS.

Hello, I'm using the "sem" package to do a confirmatory factor analysis on data collected with a questionnaire. In the model, there is a unique factor G and 23 items. I would like to calculate the standardized residual variance of the observed variables. "Sem" only gives the residual variance with the "summary" function, or the standardized loadings with the

Hi there, I searched in mailinglist and in web, but no answer to my problem... Only this post with no answers: http://lists.digium.com/pipermail/asterisk-users/2004-March/038994.html I'm using CVS Asterisk (05/17/04) with chan_capi 0.3.1. (multiple controller support). In my Asterisk-box there are 2 Fritzcards (module for second card compiled with changes on sourcecode found in the web).

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]

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

Hi, I have been making an RPM of FLAC to bundle with GStreamer. In order to get it working I had to make some rather hackish solutions in the SPEC file. The flac Makefile does to build into the correct directories while creating an RPM for some reason. I have attached the SPEC file I ended up with if it is of interest. Of course it didn't help me much cause it turned up we had a bug in the

Hello Routing Gurus ;-) I''d like to know if it''s possible to make a routing decision for pakets originating from a specific port of the local machine without using ipfilter/iptables to mark the pakets. I read about the tc filter stuff but that seems only to be able to sort the pakets to a different queue on the same interface and not choose a different interface for example. Is

Hello, I cannot get JGR installed. Here is what I have tried so far http://wiki.ubuntuusers.de/R 1. sudo -s # root werden R CMD javareconf root at gunnar-laptop:~# R CMD javareconf Java interpreter : /usr/bin/java Java version : 1.5.0_18 Java home path : /usr/lib/jvm/java-1.5.0-sun-1.5.0.18/jre Java compiler : /usr/bin/javac Java headers gen.: /usr/bin/javah Java archive

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

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

hi, for some unknown reason i cannot install the package ca in R running in a ubuntu mint system. i keep getting the following error message: configure: error: missing required header GL/gl.h ERROR: configuration failed for package ?rgl? * removing ?/home/kat/R/i486-pc-linux-gnu-library/2.12/rgl? ERROR: dependency ?rgl? is not available for package ?ca? * removing

Hi, on AIX, mkstemp doesn't open a file with the O_LARGEFILE option, so you can't transfer files > 2GB to an AIX machine. Here is a fix: diff -c -r rsync-2.5.6.orig/syscall.c rsync-2.5.6/syscall.c *** rsync-2.5.6.orig/syscall.c Sun Jan 26 21:09:02 2003 --- rsync-2.5.6/syscall.c Wed May 14 13:55:15 2003 *************** *** 151,157 **** if (dry_run) return -1; if

It gets definitely better every day. List of bug fixes follows: Release 1.0.5.9 7/26/2004 If SIPRegister doesn't proceed due to conditions unmet, release channel resource Fix the LED flashing issue when connection to the SIP proxy is lost. Fix the issue where the device will not resume registration when it loses connection to the outbound proxy for some time. Fixed the

I currently have a data frame whose rows correspond to each student and whose columns are different variables for the student, as shown below: Lastname Firstname CATALOG_NBR Email StudentID EMPLID Start 1 alastname afirstname 1213 *@uark.edu 10295236 # 12/2/2008 2 anotherlastname anotherfirstname 1213 **@uark.edu ## 10295236 9/3/2008 Xattempts Q1

Hallo Zusammen, hatte gestern die Schnauze voll von Vista und habe mir daraufhin Ubuntu installiert. Bin also mit Ubuntu ?berhaupt nicht vertraut. Nun habe ich ohne Erfolg versucht, R zu installieren. Soweit ich der Anleitung folgen konnte habe ich es so wie auf http://wiki.ubuntuusers.de/R gemacht. Im Terminal kann ich R problemlos starten. Allerdings bei der Installation von

Having some trouble with sip behind a nat. So tried: stunaddr = numb.viagenie.ca in sip.conf. Didn't help so tried stun debug: asterisk*CLI> stun set debug on STUN Debugging Enabled STUN Packet, msg Binding Response (0101), length: 36 Found STUN Attribute Mapped Address (0001), length 8 Ignoring STUN attribute Mapped Address (0001), length 8 Found STUN Attribute Changed Address (0005),

I am trying to "dial" a GTalk, ie @gmail.com, address. I inscribed this address in jabber.conf on the buddy= line. Upon executing the Dial application, I hear only a brief brief ring, then nothing. What might be the trouble? As the Dial application starts trying, the JABBER chatter on the console includes some "INCOMING" entries that name IP addresses. The one with

Hi, I''ve got a rather confusing problem. My linux router box has several internet uplinks of various kinds (pppoe, ippp, ethernet). These uplinks are used by a tunnel to another location. It kinda looks like this: eth0 - internet uplink eth1 - LAN tun0 - tunnel device ppp0 - another internet uplink ... Routing is setup with iproute2 in a way that pakets with a source IP from the LAN

examples of "interesting" information like ICE result and howto make "minimal" configuration of pjproject.conf i.e. for  debugging app_queue.so core set debug 5 app_queue.so for debugging RTP core set debug 10 rtp_engine core set debug 10 res_rtp_asterisk rtp set debug on logger.conf rtp => debug,verbose(5) so i mean in

I have been trying out several stun servers with Asterisk 1.6.0.9 and 1.6.1.0 and I keep getting the following message: [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: