similar to: Monitoring SIP with Whats Up Gold

Is there a traceroute equivalent in the VoIP world? I would like to see the route a call takes after it gets to the gateway. Basically showing all the hops until it reaches it's destination or PSTN termination. -Dave

Hi, I am trying to install WUG; This requires a number of prerequisites to be met from the get go. An installation of IE6 or above...done An installation of MS Host Script 5.6 or above....done An installation of MDAC 2.6 or above..again..done An installation of client for Microsoft Networks and File and Print Sharing...failed :( I have the current version of wine running on Ubuntu Lucid and

I know that there are some patches being worked on to cache realtime users that might ultimately fix this problem, but until then, here is a little script that brings back the MWI when using the excellent mysql realtime architecture with sip: http://www.cheapnet.net/~mike/asterisk/send_mwi.txt This script relies on sipsak utility found at http://sipsak.berlios.de/ Download, rename to

>Date: Sat, 11 Oct 2003 22:07:49 -0700 >To: asterisk-users@lists.digium.com >From: John Todd <jtodd@loligo.com> >Subject: RE: [Asterisk-Users] SIP / IAX over satellite > >[post re-ordered chronologically] > >>-----Original Message----- >>From: asterisk-users-admin@lists.digium.com >>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tilghman

I've got a client who would love to be able to leave an asterisk server running sompelace, and achieve telephone connectivity using an IP phone from within his Motorhome in his words "I want to be able to work from a mountaintop in Glacier National Park" I've done some initial testing, and a SNOM200 SIP phone comes close enough to working that I have not ruled out the idea as

I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had

I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest as my PSTN interface. I'm experiencing random dropped calls on the various SIP devices I have tested. Network connectivity to the SIP devices looks ok, and I have tried a variety of the devices including all of the following. Grandstream 286 Grandstresm 486 Sipura SPA 1000 Mediatrix 2102 Some example lines from my logs

Does anyone have some decent Nagios scripts out there that do more than monitor the proc itself? Rather than reinvite the wheel, figured I'd ask. I already saw the one on the wiki. Matt

Hello, Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. Thanks by advance.

Hello, you can have a look on Homer http://sipcapture.org/ regards On 27/09/2016 10:39, Gholamreza Sabery wrote: > Hello, > > For service monitoring you can use tools like sipsak in combination > with Zabix or Zenoss. Also using Zenoss or Zabix you can monitor the > health of your servers. This way you have both top-down and bottom-up > monitoring. For monitoring call

I am a bit confused by the documentation of the gold plugin (http://llvm.org/docs/GoldPlugin.html). I want to use the gold plugin on my Linux system so I can run "llvm-gcc -use-gold-plugin." In the documentation it talks about building gold with plugin support (which involves running "make gold-all") and building the LLVMgold plugin (which involves configuring LLVM with

When I build LLVM 2.6 after configuring it with the --with-binutils-include option so it points to the gold plugin src/include directory, I get the errors below. I used "cvs -z 9 -d :pserver:anoncvs at sourceware.org:/cvs/src co src" to check out the gold-plugin, as is documented on llvm.org. -------------------------- make[2]: Entering directory

In an earlier post, it was suggested that the developers of the gold plugin do not necessarily frequent the LLVM dev mailing list. So, how should I go about contacting the developers of the gold plugin? Regards, Ryan

The errors are /usr/lib/libstdc++.so.6: version `GLIBCXX_3.4.9' not found /lib/libc.so.6: version `GLIBC_2.7' not found I can not use it in the command line... Rafael Espindola wrote: >> However, I got >> "... >> checking for gcc... llvm-gcc -use-gold-plugin >> checking for C compiler default output file name... >> ... >> configure: error: C

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Hi, I've tried to use SIPSAK to understand the troubles i'm having about sending my voice to the person I've called (extension), after doing this tests below I always got this error "error: this FQDN or IP is not valid: voicegw". This could cause problems (namely audio problems)? Best regards, Helder voicegw:~# sipsak -C empty -a password -s

the symlink ld is already pointing to /usr/bin/ld. Also -fuse-ld=gold does not works on clang in freebsd. I am not sure where is the problem?should I remove /usr/bin/ld and create a new symlink to /usr/local/bin/ld.gold? Regards Aditya Verma Junior Undergraduate IDD Computer Sc & Engg IIT(BHU), Varanasi(UP) On Mon, Jun 22, 2015 at 5:25 AM, Rafael Espíndola < rafael.espindola at

I have been trying to enable the gold linker on FreeBSD to use the link time optimizations. I made gold from the binutils under /usr/ports. After building binutils using make -k install clean i got ld under /usr/bin and in the directory /usr/local/bin i got ld, ld.gold and ld.bfd. Now while trying to use link time optimization for the simple example programs

Hello, Is it still possible to produce the LLVM bitcode after linking using the LLVM gold plugin? I get the warning "ld: warning: Ignoring flag also-emit-llvm" and the LLVM bitcode is not dumped. I installed the LLVM gold plugin using the latest binutils from CVS and LLVM 2.6, since I read on the discussion list that this was possible with LLVM 2.6. This is the command I use:

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the