Michael Wallette wrote:
My only gripe is the initial configuration, although even that isn't too
terribly bad. You must download and unpack a C program, then edit a
config file that the C program pushes to the Iaxy. If you want to change
You can do provisioning from within Asterisk, after
Free Software/Open Source Telephony-Summit 2006
Tuesday, May 2nd 2006
Wiesbaden, Germany
For the third time the German Unix User Group (GUUG - www.guug.de)
organizes the Free Software/Open Source Telephony-Summit, an
international workshop
Rich Adamson wrote:
Is this worthy of opening a bug assuming the above comment is still
valid? Would the individual(s) maintaining res_snmp want to log into
either of these internet accessible boxes to identify the root cause?
The module loader in trunk is undergoing changes that will
RumaTech wrote:
And it keeps running like that. Call usually come through OK. If i try
to use show g729 command, it shows that all codecs are in use. Well,
this is fine, I am using one, but I do not want to see those warnings.
Once is quite enough. Those continuos warnings make it impossible
Rudolf Ladyzhenskii wrote:
I am not. I have one license and use i channel.
It seems to detect the fact there are no more channels left and keeps
warning me about it in case I want to use more.
I reviewed the code for that module after reading your original message,
and confirmed that it will
Steve Kennedy wrote:
Each channel needs TWO licenses, one for each way (I think).
Nope. The encoder/decoder licenses are counted separately, and each
license you purchase entitles you to one encoder and one decoder.
___
--Bandwidth and Colocation
Due to an error in the configuration of the mirroring tool we are using
to mirror the repositories from our internal commit server to the public
read-only mirror, the revision numbers were not being properly kept in
sync (so rev 14381 on the internal server was not the same as on the
mirror).
Aaron Daniel wrote:
Of the people in here that have hinting working with the polycom 601's
(or any phone for that matter)... do you have it working so that the
shared line appearance shows that there's someone on the phone? If so,
any hints on how to do it?
It's not a shared line appearance.
Aaron Daniel wrote:
Ok, with the buddies, what device do you hint to? The last line of
the phone?
I don't understand the question... the 'buddy' is effectively a
speed-dial, the same thing you would dial to call that person/extension.
___
--Bandwidth
mustardman29 wrote:
The Sangoma has a very nice mechanical design on their new analog cards that
will fit into standard PC cases or 2U rack mount cases. Don't know about
the Digium as it's a nonstandard oversized card that takes up one full
length PCI slot and then some.
This is
Matt wrote:
Ooo! Good point! Even though 100 instances of the same
mp3/wav/etc may be opened... linux really only has it opened 'once'.
Forgot about that :) Good point!
And there aren't additional threads created for this either; the thread
already servicing the channel handles
Dov Bigio wrote:
When I call a mailbox in a context company is doesn't play my busy
message... It goes directly to the temp message...
Am I doing something wrong?
If you have a temp message, it is supposed to override your other messages.
___
Lorentz Hinrichsen wrote:
I've had very poor results with the Digium cards, I am using a couple of the
new Sangoma ones now (they are cheaper and have hardware echo cancellation).
Which boards are cheaper _and_ have hardware echo cancellation?
___
A_ Navone wrote:
speech rec what works ?
anything out there with established dictionary, eg medical ?
don't want to pay $3-4K for Nuance API
thx in advance
The LumenVox SRE is going into beta testing with Asterisk integration
this week, and we expect to be able to release it to the public
Steve Totaro wrote:
Just looking for unsolicited thoughts on the Aheeva product? Anyone
have anything to say?
H soliciting unsolicited thoughts. Interesting :-)
___
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Asterisk-Users
Eric Lyons wrote:
I'm unable to set any cdr fields except userfield and accountcode using
Set(CDR(lastapp)=foo) [for example]. This is 1.2.4. I'm using the
cdr_addon_mysql plugin, which handles userfield, accountcode, and
uniqueid fine.
The documentation shows that those fields are
Daniel Hazelbaker wrote:
Does anybody know how big a presence Asterisk and/or Digium will make at
Networld Interop this year? I have a part-time guy that is building an
Asterisk system for us (in a proof of concept fashion before we do a
full switch to it) that I would like to take, but I
Mark Quitoriano wrote:
What model can i use for an xseries 346 server, i think the pci slot is
64-bit? Im just going to use it for asterisk timing so the cheapest will be
the best.
The x346 has PCI-X slots that supply 3.3V signaling, so any Digium 3.3V
compatible card will work just fine.
Anton Krall wrote:
I was wondering, besides recording the queues, I also use mixmonitor on my
dialplans for some extensions, does mixmonitor also use sommix to mix the
call legs are is mixmonitor mixing realtime using inernal asterisk
functions?
MixMonitor mixes the audio internally.
Sam Tam wrote:
I have used 729 and find those problems are gone. I am currently using the
free IPP g723 and 729 license and I can't see why it works for 729 and not
723.
You have clearly identified that the problem is with the G.723 codec you
are using, so you should contact the provider of
Mark Edwards wrote:
I have a single PRI span setup at present and need to dial a prefix number
in order to suppress outgoing caller ID.
Really? Normally you would set the calling presentation to 'restricted'
on a PRI, no prefix would be needed.
___
René Enskat [Teamware GmbH] wrote:
Apr 10 10:21:18 VERBOSE[5873] logger.c: [codec_g729a.so]Apr 10 10:21:18
DEBUG[5873] loader.c: Unexpected signature: 8e 93 22 83 f5 c3 c0 75 ff
8b a9 be 7c 43 74 63
Apr 10 10:21:18 WARNING[5873] loader.c: Unexpected key returned by
module
Marco Mouta wrote:
I have this to access directory of Asterisk:
exten = *411,1,Answer
exten = *411,2,Wait(1)
exten =
*411,3,AGI(directory,general,ext-local,${DIRECTORY:0:1}${DIRECTORY_OPTS})
exten = *411,4,Playback(vm-goodbye)
exten = *411,5,Hangup
The Asterisk directory is an
Erik wrote:
IMHO Asterisk B should change it SSRC to tell Asterisk A the RTP source has
changed (and fix this timestamp gap)?
That's an interesting question; since Asterisk is not actually a proxy,
in point of fact the SSRC has _not_ changed, since Asterisk B is still
the source of the RTP
Douglas Garstang wrote:
exten = 2001,1,Page(SIP/3254105)
does strange stuff. The caller's phone immediately drops into the call, while
the callee's phone is still ringing. I'd think it was a SIP messaging issue,
except that the Dial() command is working fine, which makes me wonder if
[EMAIL PROTECTED] wrote:
Wanted any /all used out of service Digium boards
Please stop posting commercial content on this mailing list, as you have
been told previously it is off-topic.
___
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Ronald Wiplinger wrote:
It does not go to the next provider. Is there a settings for timeout
to go to the next provider???
Uhh... yeah. That is why there is a timeout parameter for the Dial()
application.
___
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Waldo Rubinstein wrote:
Can anyone provide any further info on External IVR application? It
seems interesting. I currently have a heavily used AGI script that I use
for a custom IVR. It is written in Perl. I wonder if it would be more
efficient to migrate it to this External IVR. Will it be
Matt Roth wrote:
The last point also brings up a question. Does anyone know how
gracefully Asterisk handles attempting to write leg files to a full disk?
I suspect it would fail in an ugly way
___
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[EMAIL PROTECTED] wrote:
I changed from a TE410P to a TE411P and fax carriers weren't detected
anymore !
I have tried everything (recompile zaptel+asterisk+spandsp ;
echocancel=yes/no ; rxgain=0/txgain=0 ; increase Wait() ; ...), nothing
worked.
The only solution that worked for me was to
Matt Roth wrote:
These statements seem contradictory. I know of no way (short of a
custom patch) to tell Monitor() to mix the in and out legs prior to
writing them to disk. On the other hand, MixMonitor() does just that
and I believe it also buffers the writes in a way that circumvents the
Rob Lith wrote:
Kevin - if you stop it from tone detection with 'vpmdtmfsupport=0' will it
detect the fax cgn?
Yes, that was the point of my message; with that setting, the software
tone detector will be used, just as it was before the OP's VPM got
installed.
Tamas wrote:
Kevin, does MixMonitor have buffering? How big is the buffer? Is it
possible to change the size? I guess, we are talking about buffering
voice samples and writing only a bulk of them to disk (e.g. in every 50
packets - 1second).
It buffers the data in memory, there is no fixed
marek cervenka wrote:
can you someone explain this bug? (or point me to number from
bugs.digium.com)
2006-03-28 19:07 + [r15699] Olle Johansson [EMAIL PROTECTED]
* channels/chan_sip.c: Fix breakage of NAT support for peers with
qualify=yes. Thanks Damin for access to your system,
George Pajari wrote:
For the moment, if you need FAX tone detection, you will need to use
'vpmdtmfsupport=0' in your module configuration for loading the wct4xxp
module; this will not disable the echo canceler, just stop using it for
tone detection.
Any idea if/when this will be
Rich Adamson wrote:
In the US, you can't.
Yes, you can. You just set the 'presentation' bits to show that the
number is not known or is restricted. However, you can't control the
actual words that show up on the recipient's device instead of the CNAM...
Douglas Garstang wrote:
I just upgraded to Asterisk 1.2.7 from 1.2.5.
Page() is behaving differently.
I'm getting an error - Incomplete destination '' supplied.
This was a bug introduced in 1.2.7. I have just fixed it in Subversion,
so you can update to the latest branch-1.2 code from there if
Tim Jackson wrote:
I have a TDM2400P with hardware echo cancel. We seem to have static on
some calls but not others and the receive audio appears 'choppy'.
Transmit side works fine and does not have any audio problems. I had to
turn up the RX gain to 18 or the receive audio volume is too
Josh McAllister wrote:
http://svn.digium.com/view/asterisk/trunk/apps/app_amd.c?rev=14714
Applications from the development trunk should not be expected to
compile against 1.2.x, since there have been API changes (among other
things).
___
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Giorgio Incantalupo wrote:
I'd like to change 3 TDM400p with one TDM2400p to avoid echo (my
intention is to use a TDM2400P echo cancel module). It TDM2400p working
good with asterisk 1.2.1? Or I need to install a new asterisk version?
There is no reason not to upgrade to the latest Asterisk
Giorgio Incantalupo wrote:
Hi Kevin,
I know upgrading is better, sorry, maybe my question was malformed...the
exact question is which is the minimum asterisk version supporting
TDM2400P?
(I have 10 pbx and I want to change 3 TDM400P with one TDM2400P on
every pbx without reinstalling a new
Rusty Dekema wrote:
If this works, I don't see why a fax transmission wouldn't work. Is it
because the fax protocol doesn't have error correction? Is that even
true?
FAX transmission is massively more complex than modem transmission. At
higher speeds, it involves 3 or 4 different 'carrier'
Giorgio Incantalupo wrote:
I'm sorry..I was wrong again...when I wrote Asterisk I meant Zaptel (I
always use Asterisk x.y.y + Zaptel x.y.z + Libpri x.y.z, same version
for all!)
FYI... those version numbers are no longer kept in sync. The Zaptel and
libpri version numbers are incremented only
Jeff Gustafson wrote:
I was looking at using a Dell server for running Asterisk and noticed
that Dell has started using PCI-X on a lot of their new systems. Does
this newer bus standard help the situation with faxing?
No. PCI-X is just a wider/higher-speed version of PCI, not a new
Jeff Gustafson wrote:
My fault. I meant to say PCI-e, which is a newer bus that Dell is
shipping on their server class machines.
Right. That is not supported by any Digium products yet, but it still
won't help the FAXing issue, since the issue is _not_ PCI bus bandwidth.
In fact, the
Jeff Gustafson wrote:
Is there any reason an easier implementation of the same, basic, idea
could be created for the Asterisk generation? According to a quick
search of H.100 it's just a TDM bus. It handles 2,048 full duplex
calls. Would a lightweight version that only supports 512
George Pajari wrote:
I'm sure you didn't quite mean to write what you have said above. Fax
transmission builds upon exactly the same ITU-T standards as data
transmission. For example, 33.6 kbps fax transmission (so called Super
G3) uses the same V.34 standard as 33.6 data modems. At slower
Dov Bigio wrote:
Is this message normal???
Apr 18 16:26:29 WARNING[1229]: channel.c:1323 ast_hangup: Hard hangup called
by thread 51792816 on Local/[EMAIL PROTECTED],1
mailto:Local/[EMAIL PROTECTED],1ZOMBIE ZOMBIE, while fd is blocked by
thread 51792816 in procedure ast_waitfor_nandfds!
Steve Kennedy wrote:
In sox terms is SLIN .ul (as in unsigned linear).
No. ul is ulaw. SLINEAR is .raw, or .sw (signed word).
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Dmitry Ivanov wrote:
Apparently, Playback(invalid,noanswer) does not work with Zap/PRI. Is
this bug?
Yes it does work. However, if your telco will not allow you to send
'early audio', then you can't do it.
A better solution is to set the PRI hangup cause before dropping the
incoming call; if
Mongi LASSOUED wrote:
I am trying to use asterisk with an Aculab card using ss7 protocol. i
have a problem when configuring zaptel and zapata files. could you give
me the right configuration of this files to get asterisk functionning
with ss7 protocol? I hope that you could help me!
Aculab
William M Conlon wrote:
3. To beat a dead horse, the Polycom 501 itself, is NOT a POE phone,
IMHO. Caveat emptor.
It is if you buy the PoE injector cable instead of the self-powered one.
___
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[EMAIL PROTECTED] wrote:
Where can we find a roadmap of asterisk 1.4 release ?
Harry... please use proper mailing list etiquette when posting to these
lists. It is very tiresome to see you quote an entire long message,
without changing the subject, and insert a one-line unrelated comment at
the
Rich Adamson wrote:
Oh, and if shielded T1 cable is used, the shield at each end of the
cable must be grounded. (Let's see how many can figure out how to do
that via an rj45 plug. ;)
You use shielded plugs and jacks, of course :-) That is why the
TE405P/TE410P have shielded jacks (as of about
Andrew Kohlsmith wrote:
Insulation (especially such thin insulation) does not prevent crosstalk.
Distance, shielding and tighter twists do.
Ever looked at the underground cable in the street outside your
building? If it's more than 20 years old, it's probably paper-insulated
gel-filled
Rich Adamson wrote:
I've never bothered to check to see if cat5 cables use the appropriate
mating twisted pairs or not. Since the pinouts are different for cat5 vs
T1 cables, I'd have to guess a single strand is used from two different
twisted pair groups. That wouldn't be cool, but in short
Alexander Lopez wrote:
6-8 spans? That's the number that I have been trying to get, and why the
limit. Is it X-talk?
I think so. I've had clients before who had to have spans brought in via
different routes even though the pairs in the underground cable were in
otherwise acceptable condition.
hugolivude wrote:
1)Will I need a digital or analogue interface card? I expect digital
is the answer, but the Digium web site said something about analogue
cards being able to support provider T1 lines whatever that means,
so I was thinking about a TDM2400P because I want on board echo
Douglas Garstang wrote:
I think a 'sip reload' will keep your sip subscriptions.
It will now, yes. The OP said he was using Asterisk 1.2.4, which was
released long before this bug was fixed. That's why it usually wise to
update to the latest release before posting a question like this to the
Peter J Dean wrote:
You are stilling going to need to answer the call before you can play
any message or music or other.
That is not necessarily true. On PRI circuits, you can usually play
'early audio' towards the caller for some period of time.
___
Peter J Dean wrote:
Would the Asterisk-PBX command Page() honour it (i had a quick look at
the app_page.c source, nothing stood out to me)?
Yes, because the 'absolute timeout' is handled in the Asterisk core, not
in any application.
___
--Bandwidth
hugolivude wrote:
Funny you mention that Kevin. I was on the web site this morning and
I saw it here:
http://www.digium.com/en/products/hardware/analogcards.php
Later on the same day, that page had changed. The text was gone and
the TDM2400P TDM400P had swapped positions...
I'll
Carlos Chavez wrote:
If I remove the eco cancellation module from a TE411P card, will it
work as a plain TE410P?
Yes. You can also the 'vpmsupport=0' module parameter to disable the use
of the module without physically removing it.
___
Ian White wrote:
Anybody have suggestions on having a 56K dialpool and VOIP connections
with an Asterisk box over the same set of PRIs? We've done the PM3 with
PRIs for just dialup, but are looking for a way to integrate our
Asterisk box and move our voice calls onto the same PRIs.
There are
[EMAIL PROTECTED] wrote:
I've been playing around with a new system I'm going to install in
another office. In setting up the Polycom's, I accidently used a new
power supply from a new 601 (24VDC) with an 600. The 600 only require
12VDC. Now, I get nothing on the screen of the 600 when I
Frank Attard wrote:
I am pasting 3 SIP messages between the Mediatrix (192.168.0.27) and
Asterisk (192.168.0.6) upon an incoming call. Asterisk is returning 407
error.
407 is not an error. SIP errors are in the 5xx and 6xx range. 407 means
Asterisk is expecting the SIP device to provide
Raymond Chen wrote:
Do anyone know to setup asterisk's SIP channel to use an outbound proxy
outside of asterisk's network to proxy the SIP message?
This is documented in the sample sip.conf file in the configs directory
of your Asterisk source tree.
Andrew Kohlsmith wrote:
First of all, don't load every Asterisk module under the sun. Load the
modules for the hardware you have, and if you're using something like [EMAIL
PROTECTED]
which loads everything, edit your /etc/modules.conf to alias the ones you do
NOT have to 'off' to
Chris Bagnall wrote:
I think what's happening is that the ADSL router is reconnecting after a
break in the connection (as it should), getting a different IP, but the
phones don't seem to be recognising they've got a different IP and updating
the asterisk server with the good news.
Douglas Garstang wrote:
When the user enters the queue again, they are being put at the back of the
queue. It seems this new variable does not work.
It works fine; lower numbers mean higher priority, and the default
priority is 1 IIRC, so you asked the queue application to put the caller
at
Douglas Garstang wrote:
I originally had a lower number, 1, but changed to a higher priority after it
did not work.
I was wrong; higher numbers are higher priority. Your example should
have resulted in the caller being placed into the queue ahead of any
callers with a priority less than 10
Marc Scheuffler wrote:
Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different
mobile network providers. Nothing.
There was a bug in various versions of Asterisk when outbound calls were
placed using spool files and then could not detect DTMF from the called
party. Without
Chris Bagnall wrote:
Okay, so assuming I've got to drop the re-registration to a much shorter
time than the default of every hour, what are the implications of doing so
(in terms of network traffic, load on the asterisk box, etc.)? What's the
lowest one can reasonably take it? 10 minutes? 1
Trond G. Andersen wrote:
Has there been done any work to support ISAC ?
ISAC is a proprietary codec from Global IP Sound. There will not be any
support for it in Asterisk unless GIPS wants to either open-source the
codec (not likely) or allow Digium to license it in the same method as
the G.729
cristian wrote:
My asterisk is Asterisk SVN-trunk-r20297 built by xxx@ xxx on a i686 running
Linux on 2006-04-20 01:02:07 UTC
This has been covered many times on this list already. The G.729 codec
binary is not compatible with current SVN trunk. If you are running SVN
trunk in production, stop
Matt Schulte wrote:
Is there something obvious I am missing? I googled this to death and
cannot find anyone with a similar issue. I remember running into this
issue in the past and quite frankly am unsure if it has *ever* worked.
Seriously? This has been discussed many times on this list and
Emmo ather wrote:
In older version of freebpx if you write somethng manually in the
configuration files it was flushed by amp, i.e. you can configure it
through the interface only. Is this this thing still present in freepbx?
Why don't you ask that on a FreePBX support list?
Mimmus wrote:
Where is the problem? Asterisk or ARI?
Since Asterisk has no control over permissions, where the files are
located or the users' names/passwords, it can't possibly be Asterisk.
___
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Douglas Garstang wrote:
I know there's bugs open on this.
This is not a bug. There is no practical way to handle a SIP client who
tries to transfer a call between Asterisk servers directly. The proper
way to handle is this to ensure that your proxy/load balancer ensures
that all SIP calls
Douglas Garstang wrote:
Then, can you please explain to me what this is all about:
http://bugs.digium.com/view.php?id=3710
This certainly appears to be a work in progress to fix this issue.
No, it is not. That work will allow a phone to transfer a call to
another server, but the request to
Jason Lixfeld wrote:
Any red flags or anything I should know? Should I bother installing a
64 bit OS? (gentoo-amd64)? Does asterisk work in 64 bit mode? Should I
turn hyper threading off? Etc?
This was just asked and answered earlier on this list today (and
yesterday), at least the 64-bit
Aaron Daniel wrote:
Has anyone done any work with implementing dbsecret for chan_sip?
No. That would only be useful for DUNDi, but chan_sip dial strings
cannot contain the secret (unlike chan_iax2 dial strings), so it would
not be useful to pass it to the remote party, and thus they would not be
Rodney G. McDuff wrote:
Is the TE411P just a TE410P with hardware echo cancellation?
Yes. Same for a TE405P and TE406P.
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[EMAIL PROTECTED] wrote:
Does digium provide a snmp solution to monitor their
telephony cards ?
Not at this time, no.
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Giorgio Incantalupo wrote:
I cannot make my TDM2400P (with Echo Canceller module) detect faxes. I
tried with a TDM400P and it worked at 80% (20% of faxes were lost). My
test conf.files are:
This is a known problem with the hardware echo canceler. For the time
being, load the wctdm24xxp module
Steve Davies wrote:
In the cases previously mentioned, the user is doing an attended
transfer using the handset features, and not Asterisk. I do not know
whether SIP even allows the Caller ID to be changed at the point when
two separate calls are bridged to one...
It does, but Asterisk does
Wes Santee wrote:
The first problem is obviously that the curly braces used in regex patterns
to
denote repeating patterns means something different to Asterisk. I would
expect
back-slashing to fix this. So...
This was just recently fixed in SVN branch 1.2, and the fix will be part
of
Samuel Tardieu wrote:
Is Asterisk supposed to honor this field and retry with the proposed
minimum Expires: field? It looks like it doesn't, and I had to change
the default_expirey globally.
Yes, it should. Please open a bug report on bugs.digium.com with a 'sip
debug' trace of this
Klaus Darilion wrote:
I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not
accept the 200 OK responses. E.g in the following example, Asterisk
retransmits the CANCEL although the 200 OK is received.
SVN trunk is not Asterisk 1.2.
There is no way to help you with this
Roger Schreiter wrote:
is it possible to let asterisk issue a SIP redirect?
A SIP invite command by a SIP client should be answered
by 30X Temporarly moved to SIP/
Have you read any documentation on the applications available in
Asterisk, or on the voip-info wiki? The Transfer()
Klaus Darilion wrote:
Shouldn't there be some error indication if Asterisk discards a response?
Probably, although it's not clear here that Asterisk actually discarded
anything. Without seeing the entire dialog, there's no way to be sure
whether there were multiple Call-IDs, multiple tags, etc.
Peter Beckman wrote:
1. Can I modify variables set in the parent context in the child
context and read them again in the parent context?
Not 'context', 'channel'. The macro you are supplying is running on an
outbound channel created by the Dial() application, so the simple answer
is
Peter Beckman wrote:
I've poured over the docs, README.variables, some of the doxygen docs, the
wiki, and it doesn't seem there is an easy way to do this.
Then there isn't, and I was mistaken.
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Ronald Wiplinger wrote:
I would like to test my extensions.conf before I give it to my users.
Is there a dialplan emulator available?
Sorry, no. There is a parse tester for the new AEL interpreter, but
there is no emulator.
___
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stoffell wrote:
Aside from being available.. What driver does it use?
Will it be needing bristuff ? (that wouldn't work I guess)
The Digium B410P will use the mISDN stack and chan_misdn for Asterisk.
Or will the near future integrate BRI ( and hfc?) drivers in asterisk?
And thus, making
Klaus Darilion wrote:
Is Asterisk not able of handling multiple early dialogs with pedantic=yes?
Asterisk is not capable of handling multiple dialogs in response to an
outbound INVITE at all. The code is not prepared for requests that it
sends to be forked by a proxy.
The next major version of
Mojo Jojo wrote:
Is there any way to use/allow SIP reinvite and still track the length of
the call?
This is discussed nearly every week on this list, it's well covered on
the wiki, and various other places. Have you tried to research this
before asking here?
The simple answer: you are
Craig Guy wrote:
From the picture on the web site it looks like it uses a cologne chipset.
Any idea if these cards will be available in Australia?
(Please to trim your replies and not reply in the middle of quoted text...)
The cards will be available through all normal Digium distribution
Bruno de Assumpção Loureiro wrote:
how can I know which version my TE405p Digium card is?
It will be reported in the kernel message log (dmesg) when you load the
driver and it binds to the card.
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Erick Weber V. wrote:
Dose someone know if the latest version of asterisk support H.264?
Asterisk SVN trunk (which will become Asterisk 1.4) supports H.264, and
I have a Grandstream H.264 phone on my desk right now which I am testing
with it (and it works fine!).
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