[Asterisk-Users] Re: [Asterisk-biz] Case studies for Asterisk Voicemail
On Thu, Jun 16, 2005 at 03:27:49PM +0100, Alistair Cunningham wrote: > I'm planning an Asterisk Voicemail system of around 3000 users spread > across several sites, each site connected by a fast network to a central > site. We're considering 2 models: > > - Central Voicemail with VoIP calls from remote sites (easier to > administer the system(s)). This will work. > - Voicemail server at each site with shared database and NFS server at > the central site (easier to connect to the existing PBXs for MWI, etc). I really don't think that you want to run NFS over the wide area. Not only do you have to be very very careful security-wise (i.e. do it over IPSec or something and make sure your NFS is not visible from the Internet itself) but do you really want to deal with the local VM server wedging when something funny happens on the network between the remote and central sites? It's not impossible but IMO you're asking for trouble doing it like this. -w ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[OT] Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
On Mon, Jun 13, 2005 at 12:48:46PM -0700, Robert Hajime Lanning wrote: > > http://www.0xdecafbad.com";> > > Protecting freedoms by putting limits on (thus restricting freedoms). > > Interesting concept. > > It maybe an interesting concept, but it is absolutely true. > True anarchy (no rules what so ever) cannot exist. Actually, anarchy means absence of hierarchy. It does not mean no rules. The "no rules" was a slag by the monarchists who called the capitalist merchant class dangerous anarchists because they were causing all kinds of worry with their "no rules" free market ideas. Anarchist societies, where and when they exist, actually tend to be quite organized. -w ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
On Mon, Jun 06, 2005 at 05:11:42PM -0400, William Waites wrote: > > If you're interested, take a closer look. chan_sip.c, some time ago. > Miscellaneous bug fixes. But a whole lot, and not for a long time. ^^ should read "not a whole lot". argh. 73 -w ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
On Mon, Jun 06, 2005 at 03:31:31PM -0400, Andrew Kohlsmith wrote: > > > So Digium has leveraged the community to build for them a > > proprietary product. Correct? > > > > Nice. > > Others have commented on this, so I'll refrain short of saying you need some > serious clue. I'm not sure I see your name in any of the CVS commit logs, so > other than whinging on about it, what really do you have to say? If you're interested, take a closer look. chan_sip.c, some time ago. Miscellaneous bug fixes. But a whole lot, and not for a long time. The reason for the "not for a long time" is because I feared that Digium would create a proprietary version. The disclaimers seemed to explicitly allow for that. It was claimed at the time that this wouldn't happen, but now it apparently has. Unfortunately. Though I may have written bluntly, the fact that people are resorting to ad hominem attacks, suggests that I have a point. -w ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
On Mon, Jun 06, 2005 at 12:20:13PM -0400, Andrew Kohlsmith wrote: > On Monday 06 June 2005 11:25, William Waites wrote: > > So is there at least a cvs tag? Can I "cvs co -r ABE asterisk"? > > Honestly, what part of "the source is not available" do you have trouble > comprehending? Sorry, due to the high traffic on these lists, I didn't read the entire thread. So this is a version of Asterisk that is released by Digium but is not released under the GPL. Correct? If it were released under the GPL, the source code would be available. Correct? So Digium has leveraged the community to build for them a proprietary product. Correct? Nice. -w -- William Waites, Consulting Technologist Consultants Ars Informatica S.A.R.F. [EMAIL PROTECTED] / +1 416 848 1527 x514 +1 514 963 4096 (Direct) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
On Sun, May 29, 2005 at 10:06:02PM -0400, Andrew Kohlsmith wrote: > On Sunday 29 May 2005 20:59, Aidan Van Dyk wrote: > > 1) Simply CVS head (as of some point in time) with certain features or > >bug fixes "backed out" > > > > 2) In addition to CVS head, some important features and bug fixes. > > I think it's simply #2. They are taking HEAD and maintaining a version where > they are extraordinarily careful about what goes in. Similar to what > "stable" was supposed to be. So is there at least a cvs tag? Can I "cvs co -r ABE asterisk"? -w -- William Waites, Consulting Technologist Consultants Ars Informatica S.A.R.F. [EMAIL PROTECTED] / +1 416 848 1527 x514 +1 514 963 4096 (Direct) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call forwarding
t(fwnum=CFIM/${fwext}); check if
> already forwarded
>
> ; ext is forwarded
> exten => s,10,Playback(fw-is-forwarded-to) ; play
> forwarded number from database
> exten => s,11,SayDigits(${fwnum})
> exten => s,12,Read(resp,fw-cancel-1-change-2,1); 1 to cancel
> fwd, 2 to change #
> exten => s,13,GotoIf($[${resp} = 1]?17:14) ; 1 entered,
> goto delete
> exten => s,14,GotoIf($[${resp} = 2]?111:15); 2 entered,
> jump to change number
> exten => s,15,Playback(fw-invalid-response); invalid
> response, loop back
> exten => s,16,Goto(s,12)
> exten => s,17,DBdel(CFIM/${fwext}) ; delete
> entry from database
> exten => s,18,Playback(fw-call-fwd-canceled) ; give status
> & end call
> exten => s,19,Playback(fw-goodbye)
> exten => s,20,Hangup
>
> ; ext is not forwarded
> exten => s,110,Playback(fw-is-not-currently-forwarded) ; say number
> is not forwarded
> exten => s,111,Playback(fw-enter-new-forwarding-number); ask for new
> number
> exten => s,112,Read(fwnum,fw-press-pound-when-finished); accept new
> number, since variable length, ask for #
> exten => s,113,GotoIf($[${LEN(${fwnum})} < 2]?114:116) ; if len < 2
> then bad number
> exten => s,114,Playback(fw-invalid-response)
> exten => s,115,Goto(s,111)
> exten => s,116,Playback(fw-you-entered); repeat back
> number
> exten => s,117,SayDigits(${fwnum})
> exten => s,118,Read(resp,fw-if-corr-press-1-otherwise-2,1) ; confirm 1
> if correct, 2 if not
> exten => s,119,GotoIf($[${resp} = 1]?120:111) ; if 1,
> proceed and update db, else loop back
> exten => s,120,DBdel(CFIM/${fwext}); delete db
> exten => s,121,DBput(CFIM/${fwext}=${fwnum}) ; add new db
> entry
> exten => s,122,Playback(fw-ext-is-forwarded) ; give status
> & end call
> exten => s,123,Playback(fw-goodbye)
> exten => s,124,Hangup
>
>
>
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Re: [Asterisk-Users] Traceroute equivalent
On Wed, Mar 17, 2004 at 09:47:34AM -0600, David Zuzga wrote: > 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. Note that sipsak will show you the path that the signalling takes -- i.e. the list of proxies that are traversed. It says nothing about the path that the audio data itself takes. -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Traceroute equivalent
On Wed, Mar 17, 2004 at 09:47:34AM -0600, David Zuzga wrote: > 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. For SIP, there is a tool called sipsak from http://sipsak.berlios.de/ that can do this. -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: There is no FORK! (WAS Re: [Asterisk-Users] BETA RADIUS support for Asterisk)
Jeremy, I am really not interested in rehashing this again. You know my views on the matter, I know yours. We disagree. On Wed, Mar 10, 2004 at 08:04:52PM -0500, Jeremy McNamara wrote: > > I seem to recall your http://www.gnutel.com publicly discussing a fork > of Asterisk. This is no secret (though it was gnutel.net, not .com). I am still in favour of a GPL fork that does not encourage people to develop non-free software, and if someone were to take up that project I would support them. But I am not working on this having chosen another course of action some time ago. > I have one shared object module that quite a few people have expressed > interest in, yes, but this is is far from a fork of Asterisk. This is > absolutely nothing different than chan_dialogic.so or even > codec_g729b.so. If you want it, you can pay for it. If not, write your > own, you have the damn code. Also note that the software ABSOLUTELY DOES NOT need to be split-licensed in order to be able to use OpenH323 and g729. Your arguments to that effect earlier in this thread are FALSE. The ONLY reason for the software to be split-licensed is to leave the door open for future development of proprietary extensions -- as you have done. > How about James Golovich aka citats? (sorry James) Yes, on this I stand corrected. Sorry James for the omission. /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: There is no FORK! (WAS Re: [Asterisk-Users] BETA RADIUS support for Asterisk)
On Wed, Mar 10, 2004 at 05:01:38PM -0500, Jeremy McNamara wrote: > > That fact is not the problem. It the fact that there is no FORK of > Asterisk that Digium secretly maintains. This is how rumors get > started. If memory serves, you were the one who started that rumour. I remember you claiming publicly that (1) you had a private fork and (2) you had licenced Asterisk outside of the GPL from Digium and had the right to distribute a proprietary version if you chose. While it is undoubtedly true that Digium does not secretly maintain a fork, your company, NuFone is closely associated on a business level with Digium, and you are as far as I know the only person outside of Digium with commit privileges to the source tree... 2+2 == ? /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[OT] Genetic Diversity (was Re: [Asterisk-Users] Asterisk on Feebsd , pls. HELP !)
On Thu, Mar 04, 2004 at 02:49:52PM -0600, Steven Critchfield wrote: > > > > Genetic diversity in operating system support is a good > > thing. It makes for more robust code. Following standards > > is a good thing -- POSIX was written for a reason. If you > > only support one OS you are less likely to notice when > > you do something non-standard. > > Ahh then you don't believe the SCO FUD that Linux sprang forth from SVR4 > or 5 or something else they supposedly own that is also the foundation > of the BSDs. I don't know what the big deal about that is. Remember 4.4BSD-Lite? In the unlikely event that SCO gains any legal traction whatsoever, any alleged SVR4 stuff in Linux can just be taken out and rewritten from scratch, it's been done before... /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Feebsd , pls. HELP !
On Thu, Mar 04, 2004 at 11:24:12AM -0600, Tilghman Lesher wrote: > > You've already answered your question. As chan_h323 does not > work on FreeBSD, and as you need chan_h323, you are therefore > required to not use FreeBSD. > > Install Linux, like everybody else. Genetic diversity in operating system support is a good thing. It makes for more robust code. Following standards is a good thing -- POSIX was written for a reason. If you only support one OS you are less likely to notice when you do something non-standard. However, I would recommend to the original poster that they use a FreeBSD 4.X release rather than -current for stability unless they need something specific that is only available in the 5.X series. /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Feebsd , pls. HELP !
On Mon, Mar 01, 2004 at 11:37:59AM +0100, Serge wrote: > Thanks William, it's get. > but new problem: > server dont have any sound device ( I think:) ) > Why noone make normal Makefile and FAQ for FreeBSD Asterisk.. als many for > Linux. I don't know why the Asterisk crowd is resistant to using GNU Autoconf, it solves these problems very neatly. OSS doesn't work on FreeBSD, just erase chan_oss from the Makefile in the drivers directory... /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
On Mon, Mar 01, 2004 at 05:32:34PM +, WipeOut wrote: > > Currently connecting more than 3 analog lines to asterisk can be > problematic unless you get hold of a channelbank (not that availible in > the UK).. > Of course there is a 12 port configurable FXS/FXO blade from VoiceTronix /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Feebsd , pls. HELP !
On Sun, Feb 29, 2004 at 04:26:17AM +0100, Serge wrote: > Hello, > > Pls. help ! > I have server on Freebsd 5.2 and don't may install asterisk , following errors: ( > gmake clean ; gmake install ) > - > include/mpool.h:53: error: syntax error before "CIRCLEQ_ENTRY" > include/mpool.h:64: error: syntax error before "CIRCLEQ_HEAD" Asterisk bundles an obsolescent version of the Berkeley DB for silly copyright reasons. Just erase any reference to db1-ast in the Makefile -- FreeBSD includes the relevant routines in libc, so you don't need it. /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling from Iaxtel to FWD users always busy
On Wed, Feb 11, 2004 at 02:02:03PM +0100, dkwok wrote: > > -- Format for call is G729A ^ I suspect that if you use a standard format your call will go through. Also keep in mind that there is no reason to go through IAXTel for this -- it is just necessary to dial SIP/[EMAIL PROTECTED] you don't need to set up a peer. /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and fax over ip - concept
On Mon, Feb 09, 2004 at 09:31:30PM +0100, Philipp von Klitzing wrote: > > Why exactly would hylafax be a "worst case" solution only, why would you > tink that that the Asterisk solution is better at all? The "worst case" would be the modem hairpinned into an FXS port, not hylafax per se. > > Instead of a fax machine, the people could have a scanner > > Hmpf... I've always found that to be a very bad replacement for an analog > fax, at least as soon as you have to deal with more than 1 page. Plain > old analog fax machines are a very well designed devices... I suppose you're right. Perhaps one of those unified printer-copier-scanner things might be better, but the models that will take multiple input pages start getting expensive. I guess it depends whether the people are doing more print-to-fax sort of things or feeding the fax machine with large amounts of paper... -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and fax over ip - concept
On Mon, Feb 09, 2004 at 02:28:02PM +0100, Dawid Mielnik wrote: > > Would the t.38 transmission be properly handled by the t.38 supporting end > points whith mediastrem passing through Asterisk ? (dont have much > experience with t.38) Has anyone ever tried anything similar / different / > wierder to try and deal with fax over ip and Asterisk ? Any suggestions and > comments are welcome. What I would do i this situation is work out a fax <--> email gateway. Best case this could be done entirely with software on the asterisk box, worst case a faxmodem hairpinned into an fxs card using hylafax. Instead of a fax machine, the people could have a scanner, and they would send their fax as an email attachment. Likewise recieved faxes could be sent directly to a printer, or to an email address, or even stored and made accessible on a webserver... Cheers, -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Code Hosting...
On Wed, Feb 04, 2004 at 10:18:10AM +0100, Andy Powell wrote: > but apparently this will never make it into CVS > (since the engine is not GPL)... GPL code is not allowed in the Digium CVS repository. Only split-licensed code is. /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Smallest server continued...
On Tue, Feb 03, 2004 at 10:46:50AM -0700, [EMAIL PROTECTED] wrote: > This thread got me thinking of other servers that would run asterisk. The > obvious question comes up if Xebian (the xbox version of Debian) would run > as a SIP only server? Asterisk on an XBox would be a small box! Cheap too. I see no reason you couldn't run it on some of the handheld pcs... Perhaps one with audio hardware and wireless ethernet... It'd make a great softphone... /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using a Dial Statement with option m and t
On Tue, Feb 03, 2004 at 01:04:27PM -0500, Matthew B Marlowe wrote: > > exten => _NXXNXX,6,Dial(SIP/611&SIP/612&SIP/613&SIP/614,30,t,m) > > The music on hold will not work I believe you do not want a comma between the t and the m. -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling while * is running
On Sun, Feb 01, 2004 at 04:51:30PM -0600, Steven Critchfield wrote: > > This isn't intended as a flame bait. The original message should have > been more clear that I thought you where experiencing crap in windows. Heh. I haven't used windows since 1995 :) In fact, with HP-UX you cannot delete or rename or overwrite a shared library if it is in use, so you would *have* to stop the process before doing a "make install". For example, http://web.gat.com/comp/analysis/mdsplus/textfilebusy.html Talks about this phenomenon. > How the hell did HP-UX get trusted status for military use if that is > true? HP was/is a big military contractor long before HP-UX came into being, so perhaps that has something to do with it... /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling while * is running
On Sun, Feb 01, 2004 at 04:21:23PM -0600, Steven Critchfield wrote: > > Dude maybe you need to learn more Unix programing and leave those toy > OSes alone. Once a module is loaded, there should be no need to read the > version on the file system again. Your problem would be loading new > modules into a running version where there may have been an api change. Steven, stop flame-baiting. HP-UX, for example, might be an ugly proprietary SysV monster, but it's far from a toy. There do exist broken dynamic loader implementations based on mmap(2). /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling while * is running
On Sat, Jan 31, 2004 at 07:43:46PM -0600, Brian West wrote: > Nope I do make install all the time with asterisk running without ONE > problem. As I said, this behaviour is specific to some implementations of dynamic loadable modules. It depends what OS (and in some cases what version of the OS) you are running. /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling while * is running
While your problem is most likely bad RAM as other replies have suggested, there is another thing to keep in mind. Some implementations of dynamic module loading have problems if a loaded module is overwritten on the disk. What this means is that it is safest to stop Asterisk just before running "make install", else the running instance may mysteriously segfault at that point. /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk with big number of extentions.
On Thu, Jan 29, 2004 at 06:27:33AM -0600, Rich Adamson wrote: > > We are thinking of making network of about 25000 extension numbers. > > These extension will be SIP phones. Asterisk will be connected to some VoIP > > gateways through H323 which will allow to > > terminate calls. > > > > Can Asterisk handle such kind of load? > > No problem, as long as none of them make any calls. The number of extensions is relevant. I am not sure the level at which begins to matter, but the comment reproduced below from pbx.c gives some idea that eventually it may be an issue worth considering. It may be that at 25k extensions the O(N+M) search starts to become noticeable. /* * I M P O R T A N T : * * The speed of extension handling will likely be among the most important * aspects of this PBX. The switching scheme as it exists right now isn't * terribly bad (it's O(N+M), where N is the # of extensions and M is the avg # * of priorities, but a constant search time here would be great ;-) * */ But of course there are ways around using ENUM and the like. -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian Packages and Mirrors
On Fri, Jan 23, 2004 at 08:45:07AM -0800, Kostur, Andre wrote: > > v0.1.11 in stable, v0.5.0 in testing, v0.7.1 in unstable (unless you're not > on an i386) Ah, I didn't realize 0.7.1 was in unstable -- I run mostly testing here. > What do you have different in your packages? Nothing in particular as far as I know... -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Debian Packages and Mirrors
FYI and to whom it may concern, I have made Debian packages of Asterisk et. al. You still need to build a new kernel and the zaptel modules from source, but Asterisk and libpri are manageable with dpkg. The debs as well as mirrors of the source distribution are here: http://www.ntgos.com/Projects/Asterisk/Download http://parc.styx.org/asterisk I would also like to mirror the CVS repository as well as set up a cvsweb... -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 Licenses from Digium
On Tue, Jan 20, 2004 at 03:09:54PM -0600, Tilghman Lesher wrote: > > The specific issue is that VoiceAge uses a copy protection method > that binds the license to the filesystem. Solution: don't use proprietary software. Then you don't have to worry about the stupid things that they do to keep their code secret. -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS and Releases
On Wed, Dec 17, 2003 at 04:45:14PM -0500, C. Maj wrote: > > My nose is bleeding from CVS. Same thing with a > T400, had to comment out all "fax" extensions. Updated > to CVS of 12/16. We really need to get this organized. 0.5.0 is too old to be useful, and having people run CVS snapshots in production causes no end of nosebleeds and headaches. the default should not be to tell people to run CVS code, that should only be for people interested in hacking on the code and trying out bleeding-edge features. I think we need more stable releases more often. Perhaps CVS commit access for some people should be discussed as well -- my impression is that the workload on Mark is rather high. And with stable releases for people to use, introducing bugs in the experimental stuff in CVS is not as critical. Mark? Cheers, -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 using non standard port
On Tue, Dec 16, 2003 at 10:59:50PM -0600, Walker Haddock wrote: > > edit iax2.h file and change line 73 as follows: > #define IAX_DEFAULT_PORTNO 80/* 4569 */ this is *really* the *wrong* way to fix it. the correct way is to set port = 80 in iax.conf BUT... you will notice near the beginning of the load_module() routine in chan_iax2.c, it does sin.sin_port = ntohs(IAX_DEFAULT_PORTNO); sin.sin_addr.s_addr = INADDR_ANY; in other words, the local address the iax2 process binds to, as well as the port, are hardcoded in the source. not good. these should come from the config file with INADDR_ANY and IAX_DEFAULT_PORTNO as defaults. can you create a bug for this on bugs.digium.com? -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD and (multiple) internal IPs
On Mon, Dec 15, 2003 at 10:05:56AM +0200, Peter Zeltins wrote:
> My Asterisk box also does NAT for internal network, and
> establishes site-to-site VPN tunnel(s). As a result I have
> several internal interfaces with private addresses on them, and
> only one public interface. By trial-and-error I've found out that
> FWD (SIP) won't work unless I disable my VPN tunnels - it would
> send the internal IP address to FWD's SIP server instead of public
> one. I assume "bindaddress" in SIP.CONF is what I need (bind only
> to public IP), but the problem is that my public IP is dynamic!
> Any ideas? Or have I missed something?
This can be a tricky one. If you only use one address range internally,
i.e. 192.168.0.0/16 broken up into subnets, then you should be fine
with the SIP+NAT patch from bug #104.
Since your public IP is dynamic, you will need to give it a stable
name -- perhaps set up Dynamic DNS or use one of the DDNS providers
so that you will know that the name, myhost.myip.com always maps
to the correct address.
Then, put
externip=myhost.myip.com
localnet=192.168.0.0
localmask=255.255.0.0
in sip.conf. as long as localnet is a superset of your internal
address ranges, it should be fine.
If you are using multiple RFC1918 address ranges, more than one of
10.0.0.0/8, 172.16.0.0/12, 192.168.0.0/16, then you will have a problem
because at the moment, we only support one internal address range
in the localnet parameter. In the future it may be possible to
do something like
localnets = { 10.0.0.0/8, 172.16.0.0/12, 192.168.0.0/16 }
but for now, not.
hope this helps,
-w
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Re: [Asterisk-Users] Re: Asterisk behind NAT << How to do it.(Leif Madsen)
On Mon, Dec 08, 2003 at 07:46:50PM -, David J Carter wrote: > Hi, > > I have chan_sip.c version 1.259 do I still need the patch. yes. -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P echo problems - seem to be fixed now
On Mon, Dec 08, 2003 at 10:27:15AM -0600, Dave Weis wrote: > > > > What about McLeod USA? > > They aren't necessarily evil, just incompetent. "Any sufficiently advanced incompetence is indistinguishable from malice" -- Jamie Reid -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX error messages in log
On Mon, Dec 08, 2003 at 11:05:17AM -0600, Steve Dolloff wrote: > > Local server: > > register => [EMAIL PROTECTED] > ; > [voip2p] > type=peer > host=dynamic > port=4569 > trunk=no > qualify=yes > context=IAX > > Remote server: > > register => [EMAIL PROTECTED] > ; > [voip1p] > type=peer > host=dynamic > port=4569 > trunk=no > qualify=yes > context=IAX this is fine if you never want to place calls between the servers you'll need two statements -- one of type 'user' for inbound calls one of type 'peer' for outbound calls on each *. that may have something to do with the log messages as well... otherwise you might use type 'friend' but that doesn't scale very well. setting the port number is not necessary. cheers, -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
On Fri, Dec 05, 2003 at 11:58:44AM -0600, Andy Hester wrote: > > The guy did leave open the possibility that he could be wrong, and said that > he'd be glad to answer any further questions or if we had some other way of > doing it. If you or some of the others think that this should be possible > then perhaps we could get together a list of more specific questions to ask. > Did he have the impression that the idea was to terminate the voice traffic on the box with the DS3, or just switch it out as IAX2 or TDMoE? My impression is that it should be a question of just dealing with line timing and reading/writing bits, which is not all that different from data. But then again I am ignorant of the design and capabilities of these cards... -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian Testing / 2.4.22 / zaptel problems.
On Thu, Dec 04, 2003 at 10:35:13PM -0800, Andrew Gillham wrote: > Well as far as I can tell, the only version I have on the box is 2.4.22-1. > I certainly only have 'kernel-headers-2.4.22-1' installed and 'linux' > symlinked > to that directory in /usr/src. i have not gotten the zaptel drivers to link properly against the packaged kernels. grab a kernel from ftp.kernel.org, build it, then build the zaptel drivers against that. -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] correct way for cvs update?
On Thu, Dec 04, 2003 at 02:20:20PM -0600, Rich Adamson wrote: > What's the correct way to do cvs update now? > > 'cvs update' seems to work in the asterisk directory, but not the zapata > or other source directories. I use 'cvs update -PAd' AFAIK it should work in the zapata and libpri directories... -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
On Thu, Dec 04, 2003 at 06:25:16PM -0500, William Waites wrote: > > btw, jason thorpe at nasa has benchmarked gige cards on netbsd/i386 > doing well in excess of 500Mbps so it /is/ possible. > Just another data point: We also made measurements in November 2000 from a Pentium III running Linux with a Gbit interface at SLAC, via an OC12 (622Mbps) link provided by the experimental NTON network from SLAC to Caltech to another Pentium III host. Over this link we achieved about 500Mbits/s with a single stream and a window size of about 800KBytes or more. The results are shown to the right. http://www-iepm.slac.stanford.edu/monitoring/bulk/caltech.html Note that they are doing the tests with TCP which needs window size tuning at these speeds. That wouldn't be an issue for IAX2 or TDMoE... -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
On Thu, Dec 04, 2003 at 03:31:06PM -0800, David Boreham wrote: > There are DS3 (and OC-3) PCI cards available > with Linux drivers (for data). Might be worthwhile > contacting a vendor of those things to see if there's > a way to suck the TDM voice data > off a channelized DS3. I know of OC3 ATM cards for linux, but AFAIK few telcos want to do VoATM these days, do you know of an OC3 SONET card? I can't find one even for POS... -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
On Thu, Dec 04, 2003 at 04:58:03PM -0600, Steven Critchfield wrote: > > > > a standard 32 bit 33MHz PCI bus has a maximum bandwidth of > > 133MBps == 1Gbps. a DS3 is 45Mbps. even if you pass the data > > over the bus 10 times, you're still only using up half the > > peak bandwidth. > > Thats only if you could get full theoretical speeds. I have friends who > work on ranked supercomputers that will tell you how far short most > chipsets fall of the theoretical. Oh, granted. I would only do this on a reasonably high end PC with a good chipset. > Also remember the DS3 speed you > mention is a one way speed. Voice being bidirectional means that it > would pass the PCI bus in and some going out. Then if you plan on doing > any recording, there will be another crossing of the PCI bus to either > go out the ethernet cable to a drive subsystem that could handle the > speed, or to a decent SCSI system locally. I wouldn't suggest doing that! I would do something like: | cluster of asterisks OC3/DS3 <---> * <---> TDMoE <---> | for recording, vm, | voip gateway, etc. and keep the config on the DS3-TDMoE box as simple as possible. ideally the DS3 interface is plugged into a 64bit 66MHz bus as well. btw, jason thorpe at nasa has benchmarked gige cards on netbsd/i386 doing well in excess of 500Mbps so it /is/ possible. -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
On Thu, Dec 04, 2003 at 10:34:02PM -, Linus Surguy wrote: > I don't want to criticize your idea, but you do have to consider certain > points. Starting from (as has already been mentioned) the bandwidth of DS3 > is far too much to reasonably shove down the PCI bus without data loss / > excessive overheads. ??? a standard 32 bit 33MHz PCI bus has a maximum bandwidth of 133MBps == 1Gbps. a DS3 is 45Mbps. even if you pass the data over the bus 10 times, you're still only using up half the peak bandwidth. -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
On Thu, Dec 04, 2003 at 02:43:40PM -0600, Eric Wieling wrote: > I believe there are boxes that will take a DS-3 from the Telco and spit > out T-1's to your telecom equipment. Not sure what they are called. you're thinking of something like the nortel access node express... doing it this way will also spread the load over multiple asterisk boxes which may or may not be a good thing depending on the requirements... fwiw, you could also take the telco circuit as a SONET OC3 which, assuming proper engineering, would put you as part of a sonet ring and give added redundancy. often the telcos will run an OC3 to the basement anyways and just peel off a DS3 for you... depending on the facilities... to add to john's question, what about the possibility of ATM or TDM OC3 cards for asterisk? at that point you could probably *build* an access-node-alike out of asterisk... -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT << How to do it.
On Thu, Dec 04, 2003 at 07:56:56PM +0100, robert ivanc wrote: > this patch seems to break my GS phones that are connecting to * via NAT. > The one before that works ok - 249 or something? They can't connect > anymore - get a Not Found error back. That is very strange -- the *only* difference between those two versions of the patch is the variable naming. Can you give me some more debugging information? Some more information on your setup and perhaps a trace of the SIP conversation? I don't have a GS phone to test with here. Thanks, -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT << How to do it.
On Wed, Dec 03, 2003 at 05:47:59PM -0200, listas iPfone wrote: > Hi! > > I need help to undestand the options: > hello. > > externip= static/ dynamic ip? can be a domain? externip can by an IP address or a domain. it uses gethostbyname(3) in the code. > > localnet= internal ip of * machine? localnet should be the internal network address not the internal ip address. i.e. if your asterisk server is 192.168.0.245, localnet should be 192.168.0.0 > > localmask= 255.255.255.0 ? that is correct. (unless you have a different netmasks of course) cheers, -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LCR with ENUM and DDNS: half the story
Ok, so you've read the Wiki and gotten call routing using ENUM to work
(http://www.voip-info.org/tiki-index.php?page=Asterisk%20E164%20Call%20Routing)
with your own ENUM-alike domain, e164.example.com.
But how do you populate it with data? You can do it manually, but that gets
very tedious very quickly. Or you can use the nifty DDNS updating program
that comes with bind9.
The first thing is to set configure your e164.example.com to allow ddns updates.
A very good document describing how to do this (just ignore the DHCP stuff) is
http://ops.ietf.org/dns/dynupd/secure-ddns-howto.html
In a nutshell (I used TSIG keys for simplicity, the procedure is analogous with
SIG(0) asymettric keys) this is how you do it.
On the client computer that will be allowed to update the database do:
% dnssec-keygen -a HMAC-MD5 -b 512 -n HOST client.example.com
Kclient.example.com.+157+13404
This creates the shared key, which will live in a file called
Kclient.example.com.+157+13404.key and .private
% cat Kclient.example.com.+157+13404.private
Private-key-format: v1.2
Algorithm: 157 (HMAC_MD5)
Key:
I9FvX+F3fcSVLkzlPSVR9THww+oN6o0mj/JgKTu9auzMx0IM7lmBd9RIfk2cbHvoV9drGQVsk+svkrf+AeN0JQ==
Now on the server, let that key update e164.example.com. To do this, change named.conf
to have
key "client.example.com." {
algorithm HMAC-MD5;
secret
"I9FvX+F3fcSVLkzlPSVR9THww+oN6o0mj/JgKTu9auzMx0IM7lmBd9RIfk2cbHvoV9drGQVsk+svkrf+AeN0JQ==";
};
zone "e164.example.com" {
type master;
file "dynamic/e164.example.com";
update-policy {
grant client.example.com. subdomain e164.example.com. ANY;
};
};
and restart the nameserver.
That's it for the configuration.
Now, say you have just found a very good IAX2 peer, FooFone that offers /wonderful/
rates
to the ficticious country code 666. You can use a script like this, to tell the
asterisk application EnumLookup (see the howto above) to use this peer for that
country:
#!/bin/sh
TTL=3600
SERVER=nameserver.example.com
SERVER=sparx
ZONE=e164.example.com
KEYFILE=Kclient.example.com.+157+13404.key
nsupdate -v -k ${KEYFILE} << EOF
server ${SERVER}
zone ${ZONE}
update delete *.6.6.6.e164.example.com.
update add *.6.6.6.e164.example.com. ${TTL} NAPTR 100 100 "u" "E2U+IAX2"
"!+(.*)!iax2:foofone/1!" .
update add *.6.6.6.e164.example.com. ${TTL} TXT "greate $0.00/minute rate from
FooFone!"
show
send
EOF
the first update line deletes any existing records for +666, the second adds the NAPTR
record for ENUM call routing, and the third adds a nice informational message in the
DNS
which is useful if you want a quick way to find out how much a call will be billed at.
Note the escaped-escaped-escape characters. The first is because the shell will try to
interpret \, so what actually gets sent to nsupdate is \\ which is correct for what
BIND
wants.
And the second half of the puzzle? Figuring out how to know what to put in the DNS,
calculating the best rates...
Hope someone finds this useful,
-w
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[Asterisk-Users] Multi-port FXO anyone?
Query: why does Digium not make a 2 or 4 port FXO interface card? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hold, park, transfer, etc-- How is it done?
On Wed, 5 Nov 2003 17:46:16 -0600, Don Pobanz wrote > If you want to ring multiple > phones > (extensions) simultaneously, then you need to specify that as part > of the dial command. exten => 353,1,Dial(SIP/192.168.50.188&Zap/10,18) Actually, I have had problems trying to do this, at least if Zap/10 is a FXO. What happens is, both are dialed, but then the FXO gets marked as "ANSWERED" before the remote end has picked up (I read somewhere that this has to do with preventing reuse of the channel...), and the SIP phone stops ringing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
On Mon, 3 Nov 2003 19:33:36 -0800 (PST), Chris Albertson wrote > > Did your patch make it to CVS? Sorry for being lazy and not looking. > >From the sounds of thing maybe only half the patch made it > But I'm not at the right machine to look at present. No, but it may have something to do with the corporate machinery churning out a faxed disclaimer -- I'm not sure if that's actually been done yet. The patch (against then current CVS -- there have been some other changes since then) is at http://lists.digium.com/pipermail/asterisk-dev/2003-October/002150.html -w ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
On Tue, 4 Nov 2003 13:00:44 +1100, Anthony Wood wrote > > Internals can use the IP address of the NAT box as the Asterisk > Server IP and then it should work. > > i.e. don't set your internal SIP UAs to connect to the internal IP > address of the Asterisk Server. > > The fix allows asterisk to work together with the NAT box to appear > to all concerned as if it has a real IP address. That only true for some NAT implementations and configurations. It is not robust in general. It would require double the ipfilter configuration and double the traffic on my NetBSD gateway, for example. The patch I submitted last week addresses this problem. Cheers, -w ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
On Mon, 3 Nov 2003 17:10:10 -0600 (CST), Martin Pycko wrote > It doesn't care about the phones. If you phones are behind nat use nat=yes > for each defined account. The fix is incorrect. Asterisk chan_sip.c must distinguish between SIP peers that are behind the firewall (together with the *) and those that are on the outside. Either the configuration flag use_extern_ip must be specific to a peer, or it must be figured out in some other way. A global variable won't do since it creates a situation where either external or internal peers will work but not both. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] two NAT patches and STUN
On Fri, 31 Oct 2003 09:09:22 -0800 (PST), Chris Albertson wrote > > Stephens, I think preferably, introduces one new sip.conf > line for the internal _network_ which acceprts a "network > address in the form inside=192.168.111.0/14 Where the "14" > would be the number of zero bits in a 32-bit mask > > Waites used two .conf lines one for the IP address and one > for the mask. IMO Stephens' approach is more cleaner. I agree, I was worrying more about the functionality than the config file. > Both of these have an "if" statment that checks to see if > the public address needs to be stuffed into the outbound > SIP packet. I would replace this "if" with one that checks > the result of a STUN query. STUN simply makes Asterisk > more self-configuring. Careful though -- we don't necessarily want to send a stun query each time ast_sip_ouraddrfor() is called. That would introduce unnecessary traffic as well as delay in setting up the call. > Ho, and one more thing. I think the NAT configuration stuff > needs to go in a more global place and not in sip.conf > as part of my STUN integration I'll look for a logical place > to but NAT stuff. I could add a nat.conf file but, "Oh no > not yet another *.conf file!" Suggestions > We need a place to list known STUN servers and a place to > put manual "overrides" to handle cases whereeither STUN fails > or gives a misleading result stun.conf? in order for other protocols to use stun, maybe it should be its own module, exporting ast_stun_ouraddrfor() or similar... > The STUN license is quite good. It is basically "BSD-like". > or X11-like and reads in short "do what you want with this > but keep this notice and don't blame us if this is broken" having it as its own module would also isolate the C++ stuff (in which the reference implementation is written) as well as the differently licensed code... -w ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN and Asterisk
On Thu, 30 Oct 2003 16:18:23 -0800 (PST), Chris Albertson wrote > > This would be VERY much like the two current patches do except > that we would no longer need the new lines in sip.conf as STUN > would figure this out for us. > you would still need the lines to specify the internal network/mask. either that or an ioctl() to get that info from the interface -- although using ioctl() that would break in the case of a subnetted internal network. without this there would be no way to distinguish between an internal and an external address even with stun. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
