similar to: Originating Client Request

I was just wondering how many people out there are using FreeBSD as a pdc. I see a few guides on the net mostly followed by a load of posts of problems people encounter. Is it like most things that once you have done it once you can soon set up a machine at the drop of hat as you encounter and remedy any problems. I have a few customers at the moment one of who requires a pdc with roaming

This isn''t even close to ready for release, but in case anybody is interested, has feedback, finds it useful as is, or even wants to collaborate... I''ve been working on a "personal information server", something of a protocol multiplexer. Right now it serves calendar feeds for: - a local calendar folder (iCal 1.x''s Library/Calendars) - rss for local todos

Our CEO, Dr. Dan Steinberg, is planning to visit Europe in May. He would like the opportunity to introduce statisticians (and statistically minded people) to data mining, data mining applications and to forefront data mining tools. Our algorithms are probably familiar to many statisticians (CART, MARS, MART, TreeNet and RandomForests), although it isn't necessary to be a statistician to

Hi, I would like to know what is the easiest way to compile two or more set of vectors or data frame, according to their index. They are interrelated to one another by their assigned index. for example: #data set 1 abc #output: X403 X408 X410 X415 X418 X419 X420 X423 X424 X425 X426 X427 549.58 541.91 544.18 549.37 555.54 540.83 543.26 544.26 546.85 548.98 553.10 556.49

Hello ! I have two lans (LAN A & LAN B) connected via freeswan vpn gateways (VPNGW-A & VPNGW-B) over DSL. Logged in at VPNGW-A, I can only connect to the clients *behind* the VPNGW-B but not the VPNGW-B itself. ---- example VPNGW-A# ssh -l remoteuser 172.20.7.1 [IP 80.133.197.130.40523 > 172.20.7.1.ssh: SWE 343226483:343226483(0) win 32440 <mss 16220>] ---- no connection I

Hello, I need to redirect web traffic originating from the firewall host itself addressed to the webserver x.x.x.x (in the net zone) to be redirected to an ip address on the local net interface of the firewall. How is this possible with Shorewall? I couldn''t find anything like that in the Documentation or Faq. PS: Pls cc me, as i am not subscribed to this list. Regards Stephan

EHLO, Somehow I don't think it makes any sense to test whether the gid/egid can be changed, if the original uid happened to be root. Root can always change the gid/egid anyhow. So, I would like to propose the following change to 3.9p1... --- uidswap.c.orig Sun Aug 29 15:43:57 2004 +++ uidswap.c Sun Aug 29 15:44:05 2004 @@ -201,7 +201,7 @@ #endif /* Try restoration of GID if

simple setup... two * boxen, X100P in each pots(1) --> ZAP/1 --> *(1) --> IAX2 --> *(2) --> ZAP/1 --> pots(2) with busydetect=yes and callprogress=yes in zapata.conf, calling from pots(1) through to pots(2) we get the following -- Zap/1-1 is ringing -- Zap/1-1 is ringing etc, and if nobody picks up, then it timeouts and hangups, good however, if during the ringing,

I have several Cisco SIP gateways sending calls to Asterisk. Because the gateways don't have user-agents, they don't authenticate with Asterisk. And because they don't authenticate, they use the default context in the sip.conf file. Is there a way to either: A) identify the inbound gateway with a variable, in channel info, or the manager interface? If there was a ${SIPDOMAIN} for

I am not sure if it is the right list for the post. Please excuse my lack of expertise, if it is a bad post. Is there anyway to detect the originating network identity of the call in Asterisk? For example, if the Asterisk gets a call from Cingular Network, is there anyway to find out that the call came from a Cingular subscriber. Thanks __________________________________ Do you Yahoo!?

I've got around 45 people who need to place calls from our inhouse app. What is the considered "best practice" for placing these calls: 1) All clients connect to astmanproxy, and use AMI API Originate command 2) All clients connect directly to the astersik AMI and use the API Originate command 3) All clients create a db record, some process reads the record and writes out a call

I have an application that uses the Asterisk Management Interface to bridge two calls using the Originate command with Dial as the action. Using one SIP server, there is no ringback on the second leg of the call. The first person is called, answers, and hears silence until the second person picks up, even though the second person's phone is ringing. When the call goes to another SIP gateway,

Hello, There is a bug in SOGo, as it sends the original IP after successful login, and not before the login process. I traced the bug to the source code. https://sogo.nu/bugs/view.php?id=2979. Then, in my research, I found this old thread: https://forum.nginx.org/read.php?29,237299,237367#msg-237367 Can I use Nginx as an IMAP proxy to add the missing ID. I suspect this is something that can be

On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy <mulitskiy at acedsl.com> wrote: > Hello, > > After I have re-read the "PJSIP Advanced Codec negotiation" document, it > occurred to me that the desired behavior should actually happen > automatically, just due to the codec negotiation logic, but it looks like > asterisk doesn't actually follow the described

Oh, that's great. It wasn't clear from that page, at least not for me. :-( Having it clearly stated on the document would save me (and probably others) lots of time. Thanks for clarifying it. Any idea on the timeframe of implementation? *Michael Ulitskiy* Ace Innovative Networks, Inc. Main/SMS: 212-868-2366 Direct/SMS: 212-812-1203 https://www.aceinnovative.com On 7/6/23 12:47, Joshua

Hello, I finally got to look at chan_sip to chan_pjsip migration again. This time I’m having problems with influencing codec selection on originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only works on outbound (called) channel and has no affect on calling channel. My experiments and function documentation (which says “Media and codec offerings to be set on an outbound SIP

Well, I'm trying to migrate to chan_pjsip so that I don't have to do that. It's so surprising that the issue so seemingly obvious and trivial hasn't been addressed yet that I wanted to query the collective wisdom of this list to verify my observations. Thanks for github pointer. Michael On 7/5/23 16:46, asterisk at phreaknet.org wrote: > On 7/5/2023 4:19 PM, Michael

Why do attacks from the Internet get shown in the Asterisk logs with myAsteriskServerIP instead of the attacker's IP?! Really useful for blocking them, that is... Example: [Mar 6 00:00:00] NOTICE[1926] chan_sip.c: Failed to authenticate user 5550000<sip:5550000 at myAsteriskServerIP>;tag=ab8537ae (I replaced our IP address with myAsteriskServerIP. The attacks are not coming from

Hi list! I'm trying to make a SIP test call from Bria and/or 3CXPhone from a PC behind NAT. From Bria/3CXPhone I connect to an Asterisk 11.25.0 server on the internet at 100.100.94.210 with a SIP account "3333" created in sip.conf: [3333] type=friend secret=something host=dynamic nat=yes qualify=no disallow=all allow=alaw allow=ulaw canreinvite=no context=voipin I dial +1234

Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider: g711 -> asterisk drops g722 and passes g711 to internal callee -> no transcoding. As far as I know,