similar to: OT: sorry ... list problem

I am setting up a small Asterisk system for use at home. I have one Budgetone 101, one Cisco 7960 and two Xten lite softphones. So far everything is working expect for an issue with the Budgetone. When a call is placed between the Budgetone and any other phone, the call is setup and sounds good. If I hang-up on the Budgetone, everything is ok. If I hang up on the other phone, the Budgetone

> On Wed, May 31, 2017 at 12:36:47PM -0700, Steve Edwards wrote: >> I want to capture all SIP messages. >> >> I have about 30 hosts in about 6 colos. >> >> My first thought was dumpcap, but the output file name format bugs me. >> >> What do you use for long term SIP capture? On Wed, 31 May 2017, Daniel Tryba wrote: > What bugs you about the output

On Wed, 31 May 2017, Daniel Tryba wrote: > On Wed, May 31, 2017 at 01:39:25PM -0700, Steve Edwards wrote: >>> What bugs you about the output format? >> >> It's been a while, but as I recollect, it included the date/timestamp in the >> file name of the 'ring buffer' which meant that each time the host was >> rebooted, dumpcap didn't know the

I recently received a GoIP-32 for a client project -- primarily outbound calling. How should a GoIP be configured for Asterisk? No fancy shmancy Elastix or FPBX GUI -- just using the configuration files. Single Server Mode, Config By Line, and Trunk Gateway Mode all seem likely suspects. How did you configure your GoIP and why? What do your relevant sip.conf section(s) look like? What does

On Wed, 31 May 2017, Barry Flanagan wrote: > sngrep? Isn't sngrep a great tool? Since discovering it my use of tcpdump/wireshark has cratered. Being able to compare an INVITE that worked with one that didn't (with color highlighting) rocks. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com

Anyone with any experience resurrecting a fried / ext2fs partition in Linux 2.2.x, drop me a line please? :-(

In article <alpine.DEB.2.20.1705311339370.15080 at ws.sedwards.com>, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Wed, 31 May 2017, Steve Edwards wrote: > > > I want to capture all SIP messages. > > > > I have about 30 hosts in about 6 colos. > > > > My first thought was dumpcap, but the output file name format bugs me. > > >

I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

I was debugging a turnup with Global Crossing the other day and they presented me with a web page that displayed a 'ladder diagram' of a call including a ton of detail all neatly organized in tabs and links so you could drill down to any level of detail needed. The copyright notice says 'Copyright? 2008 Empirix.' Is there any free software available to analyze a pcap or

I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :)

A client asked me to come with a system that will pass certification with PPP in the UK. Google is not being helpful :( It has something to do with recording calls in case the PPP requests a copy. Supposedly their rules were relaxed on August 1st if that helps. Any clues (links?) will be appreciated. -- Thanks in advance,

On Wed, 25 Feb 2015, A J Stiles wrote: > The limiting factor with a switch carrying IP telephony traffic is not > bandwidth, but routing table entries; and even cheap switches nowadays > will usually take 1024 entries, if not 4096. Are you referring to the MAC CAM table? Saying 'routing table' and 'switch' in the same sentence seems confusing. Do VOIP devices take

Does anybody have contact info for Dan Toma, the author of Diax? I've tried danto at clicknet.ro and danto at rdslink.ro without success. Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

Anybody care to share a script or snippet of what they use to remove leading and trailing silence from customer recorded files? I've fiddled with sox to remove the leading; reverse the file; remove the now leading; and reverse the file again, but I'm not really happy with my results. -- Thanks in advance, ------------------------------------------------------------------------- Steve

I have a client that wants to put a phone on their web page for customers to call them via their Asterisk server. ) A keypad is needed to enter credit card details. ) "Speed dial" buttons like "Tech Support," "Sales," etc. are a requirement. Actually, passing the SIP address in the HTTP link would work with a bit of arm twisting. ) Free is preferred, but not a

I have started to read r-help on Gmane using Thunderbird as a reader. Before that I had messages sent to gmail account. The first post through Gmane needed moderator approval, but then I got a message from Your friendly autoauthorizer at Gmane.org saying You are now authorized to post to the gmane.comp.lang.r.general newsgroup. But my messages are still needing moderator approval, which I

A lot of people are going for the "VOIP only" approach, but SBC says you have to have an active analog voice circuit before they will sell you DSL. Does anybody know which DSL providers will sell you DSL without making you pay for a voice circuit? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com

Hello, I work in a corp with a lot of active directory domains (20 in total) in a forest, these 18 are in windows 2003 with '2003 level mode' and twin in AD 2012 with '2012 level mode'. There is a domain hierarchy like this: . Domain2003A <----------Trust relation------------->Doman2012X<------Trust relation----->Domain2012Y |- Domain2003B |- Domain2003C .

I've installed Sun's SUNWski package on Solaris 8 (32-bit) that provides a /dev/random interface. It appears to as cat'ing it gives me a bunch of well, random data. However, when I ran my configure, it gives me the WARNING.RND message to the effect that I'm using the built-in. I've seen allusions on this list to building openssl with to get random support, so I rebuilt it

As the OP for this thread, it saddens me to see that the thread I started has now been used as a forum for behavior of the worst kind seen in professional circles. I'm a longtime user of CentOS and merely wanted to know of users' past experiences transitioning between SL and CentOS. My first experience with CentOS was as a transition from Whitebox to CentOS, and the process was as