similar to: OPTIONS to query endpoint capability

I have been sending OPTIONS requests 1) programatically (my own code), 2)manually via SIP VERIFY PEER x and 3)automatcially by setting verify=yes in sip.conf. The trouble is I do not see anything except an ACK 200 come back from endpoints and it does not contain any SDP/codec info. . My goal is to determine audio and video codec capability in advance of a call INVITE. I notice in both 2 and 3

Hey everyone, I have several RStudio (.rnw) files that I am using a script to assemble into a composite document. For this purpose, it would be helpful if the script could automatcially make a call to whatever is called when a user opens up one of these files and manually clicks "Compile PDF". Basically, this would allow me to refresh all of the reports before assimilating their

The VC visualizers are automatcially installed. 2016-12-20 19:45 GMT+02:00 David Blaikie via llvm-dev < llvm-dev at lists.llvm.org>: > Added something in r290186. > > Say, Reid - do you know anything about the MSVC formatters that are also > provided in LLVM? Perhaps you could add a brief bit of documentation about > them, if they need an explicit wiring up (if they just

Dandy :) I looked into ways to do this for the GDB visualizers - but was unable to come up with a totally automated solution, unfortunately. On Tue, Dec 20, 2016 at 9:51 AM Yaron Keren <yaron.keren at gmail.com> wrote: > The VC visualizers are automatcially installed. > > > 2016-12-20 19:45 GMT+02:00 David Blaikie via llvm-dev < > llvm-dev at lists.llvm.org>: >

> On Dec 20, 2016, at 9:55 AM, David Blaikie via llvm-dev <llvm-dev at lists.llvm.org> wrote: > > Dandy :) > > I looked into ways to do this for the GDB visualizers - but was unable to come up with a totally automated solution, unfortunately. I've wanted the same. Best I could think of was to embed them (or a reference to them) in the binary for the debugger to look at.

Hello The end-of-line or new-line character is not interpreted when I open a shared file using MS notepad. The file was created on a Sun Solaris system - The contents of the file is " I newline am newline testing newline samba" when I do a hex dump of the file on Unix I can see the 0d 0a at the end of each line and the same on the Windows side but when I open the file with Notepad I get

Hi All First time poster, long time reader. I just ran into something really bizarre. I've enabled videosupport and have been testing sip calls with Xten Eyebeam software, it works (mostly) However, when I have Videosupport=yes In the [general] section of my sip.conf, broadvoice calls fail w/ "We're sorry your call cannot be completed at this time" So... I've

another issues with cisco 7975 I have phone registered on asterisk have 2 different issues on different versions of firmware, on 9-4-2-1S I have not working 3way conference, when I trying to connect second call, phone says ?unable to set up conference? and sending some cisco xml data to asterisk which cannot be handled, thats the problem, I know on firmware 8-5-4 3way conference works just

Fedora: Linux abcde 2.6.11-1.1369_FC4 #1 Thu Jun 2 22:55:56 EDT 2005 i686 i686 i386 GNU/Linux Asterisk: 1.2.4 SIP Problem 1. Asterisk sends SIP messages to Softphone. 2. Softphone receives SIP messages and replys back. 3. Asterisk doesn't receive these replies and keeps on sending. Asterisk: Reliably Transmitting (no NAT) to 192.168.1.4:5060: OPTIONS sip:192.168.1.4 SIP/2.0 Via:

Hi everybody, I have some problems making calls from a sip user (HT286) to the pstn trough Digium Wildcard TE110P, i allways have an error : SIP 403 INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd From: "test" <sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79 To: <sip:0170708959@192.168.1.4;user=phone>

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

When using the global options: ================================ interfaces = eth0, lo bind interfaces only = Yes hosts allow = 192.168.1. 127. ================================ and a secondary loopback address: ================================ ~ # ip addr show dev lo 3: lo: <LOOPBACK,UP,10000> mtu 16436 qdisc noqueue link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00 inet

-------------- eth0------eth1 eth0------------ |211.241.219.xx | --- | ROUTER | --- |192.168.1.4 | --------------- --------- ------------ when i send traffic from ROUTER to 211.241.219.xx or 192.168.1.4(masquraded), the filter works fine... In ROUTER, tc filter policy is like this: tc filter add dev eth0 parent 1:0 protocol ip u32 match ip dport 80 0xffff

hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get thorugh: here is my sip debug outout: thx for ur help!! <asterisk-users at lists.digium.com> --- (13 headers 16 lines) --- Sending to AA.BBB.CCC.DD : 28127 (NAT) Using INVITE request as basis request - Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk. Found user '701' for '701' Found RTP audio format 107 Found

Stuck with TLS transport, I have 2 phones (both in local network for tests) one connected with up second with tls when I calling TLS to UPD everything is fine, but when UDP calls TLS I getting an error ERROR[44230]: pjsip:0 <?>: tlsc0x7f143012 TLS connect() error: Connection refused [code=120111] pjsip log: -- Called PJSIP/601/sip:601 at 192.168.1.55:5075;transport=tls <---

Hi Here is a short description of my network: ppp0 (adsl) ppp1 (adsl) | | | | --------------------- | Router | | Firewall | | MASQUERAD | | DNAT | | | | eth0 | --------------------- | | | ---------------------- |

OS- windows vista 32-bits Control Panel -> Network & Sharing center Network discovery ON File Sharing ON Public folders Sharing ON Password protected sharing ON Media Sharing ON Window firewall turned OFF McAffee firewall turned OFF Case 1: Connect to LAN using wireless connection (network type= private) I can open the windows share, look at the list of shared folders & open them.

Hi, I am working on a cluster of PCs, and I want to boot them from an SGI IRIX machine. The PCs use Supermicro XEON MB, with dual GigE built in. They have Intel Boot Agent GE v 1.1.09 and PXE 2.1 (083). I have DHCP and TFTP on the IRIX machine, pxelinux.0 loads, but it can't load the configuration file? tftp on the host tries to send the file, but data transfers do not receive ACKs. I got

Hay guys, got trouble with registration with cisco 7975 Here is the debug : <--- Received SIP request (576 bytes) from UDP:192.168.1.61:49533 ---> REGISTER sip:192.168.1.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK35076381 From: <sip:111 at 192.168.1.4>;tag=0c8525a68961001f44d503e2-d9359bd3 To: <sip:111 at 192.168.1.4> Call-ID: 0c8525a6-89610004-b972d038-5864c98e

Given a DC environment where very few (1-3) hosts actually need to be discovered via browsing is there a good way to limit what is browseable? I'm thinking of something like a read-only WINS - where WINS provides only those servers that need be contacted and doesn't allow client registrations. Such as a wins.dat that only contains the following: ========================================