1.4 to
learn what the differences are.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing lis
ow the new registration way works on asterisk 1.4 and 1.6?
Of course... it's already present in the tree, amazingly it is even
called 'app_skel.c'.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
>
> Any help will be greatly appreciated.
You will probably have a much greater chance of getting a useful
response on a HylaFAX mailing list instead of an Asterisk mailing list.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The
ish 7-vs-10 digit numbers by the number
pattern. In other words, this will work fine if you are dialing from a
SIP phone, but not if you are dialing from an analog phone.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
used at
the moment; this situation should be rectified in the next week or two.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation Provided by http://www.
f day in the near future. It has been heavily tested against many
browsers, including IE7 and works well in all of them.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwi
this one:
> http://www.thinkgeek.com/tshirts/itdepartment/60f5/ ?
This is actually caused by a minor bug in zaphpec_enable itself, it will
be fixed in a future release. Thanks for reporting it :-)
--
Kevin P. Fleming
Director of Software Technologies
Digiu
commercial purposes.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUB
and it
would have to be relinked into Asterisk if it got upgraded anyway?
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation Provided by http://www.api-digi
t actually being set.
This is a bug in Asterisk, which has been corrected in Subversion branch
1.4 and will be in the next 1.4 release.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
7;TDM410P' was a
TDM400P with a single FXS module on it. The part number for that product
is now TDM410B, which is a TDM410 with a single FXS module on it.
Unfortunately some sites persist in referring to the TDM410 as a TDM410P
(VOIPSupply, for example), so confusion reigns.
--
Kevin P. F
official, and is located here:
http://blogs.digium.com/2008/05/19/zaptel-project-being-renamed-to-dahdi/
In addition, there will be a Frequently Asked Questions page located at:
http://www.asterisk.org/zaptel-to-dahdi
We will post details on that page as we can over the next few days.
--
Kevin P
'dahdi show channel' as it will show you
that the echo canceler was disabled automatically.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Co
Johann Steinwendtner wrote:
> I thought the ec gets disabled only by the ec disable tone and not the CED
> tone.
The CED tone *is* the echo canceler disable tone.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Expe
tful that *any*
residential phone service will work in the way that chan_zap is
expecting; FSK and neon MWI signaling are generated by legacy PBX
systems with analog ports, not telco (CO) switches.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Aster
/telephony/codec_g729/unsupported/asterisk-trunk/linux/
Although I see only v33 there... we seem to be running a bit behind.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-
27;s both
easier and cheaper for the user to purchase, especially if they are
installing this card into an existing server.
At this time Digium isn't selling a single-port T1 card with a hardware
echo canceler so this is somewhat of a moot point, I suppose :-)
--
Kevin P. Fleming
Director of S
Paul Hales wrote:
> But a single port E1 card with hardware echo cancellationpossible?
Yes, I would say that is definitely possible (wink wink).
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experie
io will have gaps when
Asterisk drops the incoming CN packets. What effect this really has on
your system will depend on where you are sending that audio and what it
can do in terms of jitter buffering and other magic.
--
Kevin P. Fleming
Director of Software Technologies
e message that
you got then your source tree is corrupted.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
ast
it
and aren't using chan_ooh323c any longer.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mai
build every
module that is capable of being built on your system).
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
--Bandwidth and Colocation Provided by http://www.api-digital.
g various stress tests on the software
> to try break it.
Digium uses an Empirix Hammer (which is an actual product, not just a
codename) to test Asterisk Business Edition and verify that it will
handle the call loads and scenarios we sell it for.
--
Kevin P. Fleming
Director of Software
and of course larger CompactFlash cards can
easily be used.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asteri
s well; the Linux kernel on the AA50 does not have NFS
support nor SMB support, and there are no userspace tools present to
handle NFS or SMB mounting of filesystems.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
__
criptions and they can replace the card. Users can
also, of course, make a backup copy of the card on a new card when they
receive the unit and have a ready-to-install replacement should any
problems occur.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "T
require a dd?
Yes, using tar/untar would work fine.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk
'-l' option to see license information for software
> included in this program.
>
> Could not generate Host-ID
This is a problem (apparently) with the brand-new register version 3.0
tool; I'll send this to the developer who has been working on it and he
will get in contact w
using HPEC since you
purchased it.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To
but '200 OK', even when a redirect
(forward) is in place.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
--Bandwidth and Colocation Provided by http://www.api-digital.com-
standard PBX with FXS ports can do,
unless it has special 'long line' drivers.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation Provided by http
Armin Schindler wrote:
> sorry for beeing off-topic here. But can anyone confirm that
> there is a problem reverse resolving lists.digium.com (216.207.245.17) ?
Our IT department reports that this has been corrected.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc.
the *prefix* for PCI-Express analog cards from TDM to AEX, but
they still follow the rest of the model naming scheme (no suffix letter
and no different model numbers that indicate included optional modules).
--
Kevin P. Fleming
Director of Software Technologie
n so that users don't get compilation
failures :-)
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asteri
d would be using the Transfer() application in the dialplan.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation Provided by http://www.api-digital
, or they restrict their
users to not performing actions that will break the billing process.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation Prov
zoa wrote:
> Asterisk does not support that yet.
Yes it does, and it puts G.722 into the SDP the way that RFC3551
specifies. To the original poster: please read RFC3551 and you will
understand why G.722 appears in the SDP with an 8000 sample rate instead
of 16000.
--
Kevin P. Fleming
Direc
tential CPU load running HPEC on 24/30 channels in 128ms mode is
quite high and could cause problems on the system. However, if you don't
have that many active channels at once, or you have a very powerful
system, or many other variables are in your favor, you can certainly
give it a try.
--
penvox products (card and zaptel versions, etc...)?
Every distributor that carried the TDM400P should have TDM410s in stock
already. Where are you located, and who do you buy Digium cards from?
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "
rmation as part of DIAL, so they would
process this sort of call with an empty CLID and CNAM. We can of course
enhance chan_iax2 to understand this method of doing things, but it
won't be backward compatible with previous versions of Asterisk or any
other IAX2 clients.
--
Kevin P. Fleming
Dir
nection can be
'hijacked' as you put it.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mai
27;t any better
> card that I already have).
It will most likely work just fine, yes.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation Provided
release cycle.
Once kernel 2.6.25 has been released, we'll fix up Zaptel to accommodate
any changes required.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth
sean darcy wrote:
> That is, is port 1 = channel 1 and slot 1?
Yes, they are. However, 'UNCONFIGURED' means you haven't run ztcfg yet,
so Asterisk cannot use the channels.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine A
s problem in the morning.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNS
Kevin P. Fleming wrote:
> I've just located an E400P from our graveyard of old cards... if it
> works, I'll be able to solve this problem in the morning.
This has been fixed in revision 3863 of the 1.4 branch; it's a one line
fix that you should be able to easily apply to e
rying to build,
Asterisk won't use it. If the version of the kernel you have isn't
compatible with the version of Zaptel you are trying to build (which is
unlikely), Zaptel won't build against it. Asterisk does not care about
kernel versions.
--
Kevin P. Fleming
Director of Software Tec
mentation.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or u
Olivier wrote:
> Do you know if Digium's B410P has an inner TDM bus so that an 8 BRI
> ports subsystem (2 PCI slots used, but 1 one set of interrupts) could be
> made out of 2 B410P ?
No, the card does not support that mode.
--
Kevin P. Fleming
Director of Software Technologi
at I am aware of that support media-only transfers are Asterisk 1.4
and Asterisk 1.6 beta releases.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Colocation
o signal is heard.
> everything else looks ok, and all other functions are ok
The Queue() application has an option to generate ringback to callers
instead of music on hold, why don't you just use that instead of trying
to craft a new solution?
--
Kevin P. Fleming
Director of Software Technol
Johansson Olle E wrote:
> That's a feature that doesn't exist in Asterisk today, but could
> easily be added.
Actually, it is there... setting 'mohinterpret' to passthrough will get
as close as Asterisk can get to 'proxy mode' for this purpose, but it
will s
arkda wrote:
> [Feb 25 12:54:01] WARNING[8661]: chan_zap.c:1437 zt_enable_ec: Unable to
> enable echo cancellation on channel 1 (Argument list too long)
Can you tell us what versions of Asterisk and Zaptel you are using?
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc.
, but chan_sip hasn't
been upgraded to support putting the remote end on hold (chan_iax2 has
been). Never mind.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth an
Zaptel running in memory is the one you downloaded and compiled? Have
you tried removing all previously built Zaptel modules from
/lib/modules/*/* and reinstalling it?
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Expe
r ability to dial out on those
ports. That has been fixed in Subversion (see issue 11855 on
bugs.digium.com) and will be in the next release.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
_
ranch 1.4 code of Asterisk will no longer generate them,
and once my battery_alarms branch has been merged into Zaptel 1.4
(scheduled to be part of the 1.4.10 release) then Zaptel will stop
generating spurious battery alarm events.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc.
(Start of common instructions)
exten => _NXXNXX,n,etc
Since there is an implicit 'Goto' from priority 1 to priority 2 anyway,
you might as well take advantage of it :-)
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experie
correct DTMF
generation for all digits then you were lucky; the platform had nothing
to do with it :-)
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and C
n source Asterisk that are not in ABE;
primarily this is channel drivers for uncommon channel technologies and
other rarely used modules.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
tery_alarms, although
one tester has reported that incoming calls don't work properly using
that branch, so it still needs some work.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
__
t you will be able to accomplish what you
want using an analog interface card, and you may not be able to do it at
all, even using a digital interface card and a channel bank.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc
> idea?
I don't believe that any version of the Polycom firmware provides that
sort of access control.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
-- Bandwidth and Co
Terry Gilsenan wrote:
Can we get that IP blocked in the postfix access list at digium? (lists.digium.com
[69.16.138.164])? Ozemail are not what you would call "active" in stopping spam.
The list had no incoming spam filtering for some time; that has now been
fixed.
__
Remco Barende wrote:
- even if * will do the software conversion will the quality of the
channel remain sufficient to provide rock solid faxing. Maybe the
channel bank is considerably better than the sipura's I have been using
so far and there would be no ethernet involved.
There will be far
Aaron Daniel wrote:
We had the bug on 1.2.2, but when I rolled back to 1.2.1 to fix the
problem, everything started working. Doesn't seem like it's a bug in
1.2.1 :)
It is not. The bug was introduced during the 1.2.1->1.2.2 transition.
___
--Bandwid
Sean Cook wrote:
Ok... I am having a serious brain fart this evening. IIRC, the next sip
draft addresses shared lines and I thought I remembered something on the
list about support for it in the near future.
'the next sip draft'? There are probably 150+ IETF drafts circulating
regarding SIP
Juan Carlos Castro y Castro wrote:
Ah -- for all intents and purposes, assume I can obtain the most kickass PC
server hardware in the known Universe. So -- any real-life experiences out
there?
We successfully ran four fully-loaded TDM2400Ps in a server here during
development testing. The conf
Rob McKrill wrote:
I am hoping someone from Digium is monitoring this thread and that they
might comment on when the new edition of ABE will be released so that we
can actually utilize the full capabilities of the IP601's attendant
consoles. Right now they (the attendant consoles) are pretty
Leo Ann Boon wrote:
I'm looking at version.h installed by Asterisk 1.2.3/4 - and the default
value is 00. I thought the value should be 010200. I know many
people have problems compiling chan_bluetooth because of this
inconsistency. Anyone has the last word on this?
That is probably a mis
Juan Carlos Castro y Castro wrote:
Sweet! So -- what are the specs of that server you tested on? Specifically,
the power supply wattage? My intended use here is all FXS, so I suppose a
limit of 2 would be reasonable.
I believe it was an HP ML350, with only a single CPU and hard drive.
By the
Leo Ann Boon wrote:
I'm looking at version.h installed by Asterisk 1.2.3/4 - and the default
value is 00. I thought the value should be 010200. I know many
people have problems compiling chan_bluetooth because of this
inconsistency. Anyone has the last word on this?
What is ASTERISK_VERSI
Michael L. Young wrote:
I have a TE411P card in my * box. I am running FC4 x86_64. I used to have
two TE110 cards in the same box that worked without any problems. Since
changing to the TE411P cards, I am getting random DTMF tones being produced
on a bridged connection through the same Channel B
Imran Ahmed wrote:
Even though no IAX client supports inband dtmf, An IAX client can send
inband dtmf which would have corrected your problem.
No, it won't. No IAX2 client will start a DSP to listen for inband DTMF,
because IAX2 is defined to always send out-of-band DTMF.
At best, if the re
Damon Estep wrote:
Does anyone know what date this memory leak was introduced and/or how to
check source code for it?
I am running a pre-1.2 CVS head version and would like to know if the
potential problem exists.
It has been present since we switched to the 'new' expression parser a
few mont
Matt wrote:
You will need a minimum 3.4Ghz Dual xeon with 1G ECC DDR, and hardware voice
processing capable E1/T1 card, such as the sangoma 104d quad pci card, in order
to run 120 PSTN calls, 1000 calls is impossible for 1 server. Centos Linux
should be fine.
This has to be some of the poore
Accursio Avona wrote:
Step 2: The IAX client make a second call executing again
Dial(ZAP/g1/${EXTEN})
an IVR answer this call and the IAX client have to send some
DTMF stil now everything works very well.
At this point call is transfered to the previous conferen
Matt wrote:
Can anyone explain why this call dropped?
The person dialed a number, the call WAS completed and connected to
the PSTN through a PRI, but they never heard audio and the call was
disconnected by Asterisk.
Very difficult to guess without any information about your system. If
you are
Matt wrote:
I guess I'm more trying to figure out what
Feb 1 22:13:43 DEBUG[18623] channel.c: Didn't get a frame from
channel: SIP/102-9fda
means.
Pretty much what it says... the SIP endpoint dropped its end of the call
and the Asterisk channel was hung up as a result.
Given the sheer numb
Accursio Avona wrote:
The IVR records the conversation between the other partecipant to the
conference and wait '#' to stop recording and a '1' to save the file.
Then I really don't understand at all... this is not functionality that
I would call an 'IVR'.
Can you show us the portions of t
Leo Ann Boon wrote:
/*
* version.h
* Automatically generated
*/
#define ASTERISK_VERSION "1.2.4"
#define ASTERISK_VERSION_NUM 00
This was a bug in the Makefile; it has been corrected in Subversion and
will part of the 1.2.5 release. Sorry for the inconvenience.
__
Michael Collins wrote:
My questions to the Asterisk user community: are you at all concerned about
the complaints made by the "forkers?" Or are they the ones who are all
forked up? ;) What about Freeswitch? Do you see that as a threat to
Asterisk or simply as yet another competing product? (
Accursio Avona wrote:
but maybe also (i'm not sure, i had not the time to study well enough
the source, and over all i'm not a so good c programmer)
that this part of code prevents asterisk to broadcast the sound to other
channels when it is not inband.
MeetMe is not designed to pass DTMF thr
Steve Totaro wrote:
Another question, If Signate is not using ABE, what are their
requirements for releasing source as far as the GUI?
The Asterisk GPL has no bearing on the external tools used to
manage/configure it, unless those tools require changes in Asterisk
itself or loadable modules
Stagg Shelton wrote:
I just implemented a system using a TE411P hardware echo cancellation
card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as
I always have. To my surprise calls out to the PSTN had a terrible
echo. 1 - 2 second delay, and quite clear. The echo was so bad th
Phone Dev wrote:
Can SuperMicro slot (that is a 133Mhz slot) be used for TE210P card ?
Yes. Digium PCI cards work in a PCI slot capable of 133MHz, but the
cards operate at 33MHz and will slow down the PCI bus to that speed.
___
--Bandwidth and Colo
Tzafrir Cohen wrote:
do the zaptel tarball (1.2.2 and 1.2.3) miss the file .version? Without
it version.h will be generated with an empty version number and some bad
things will happen, IIRC.
It was a bug in the release script; the script has been fixed for future
releases.
_
kurtz wrote:
I've had no luck using a Zap extension as a member in a queue.
member => Zap/123444 doesn't seem to ring.
That is not a valid member string for a queue. Zap/1 (as in channel 1)
is, but Zap/1/1234 is not. What you specified would look for channel
'123444', which I'm sure d
Sam Tam wrote:
The long waited Ultimate GSM Gateway is finally out. This time we have managed
to source a new patch of brand NEW GSM Gateway at prices that is only 50% of
what the market rate. And with the SMS Function and many more...
What part of 'non-commercial discussion' is hard for you
Ken D'Ambrosio wrote:
They use the G.168-2002 algorithm; I, personally, had never heard of it
before, but I bounced it off a friend of mine (he's a hardware architect
for a major VoIP switch manufacturer -- they sell to places like Time
Warner), and he was of the opinion that G.168 is the _ONLY_
Tim Connolly wrote:
I wonder if Digium has any intentions of fixing this. I brought this to
their attention shortly after purchasing a pair of TE411's. You can issue a
loopup on span 2 only to get a message saying "looping span1" which is to
say, a bit scary when you only have two active PRI and
Tim Connolly wrote:
Hmm.. I'm running CVS-head from a few days ago. TimeWarner said they
couldn't loop me, so I plugged in a router and they were able to loop and
test the PRI. Is there any way to do loops from within the Asterisk console?
I typically use zttool.
You cannot be involved in a loo
Douglas Garstang wrote:
Thanks for the reply Kristian, but you've completely confused me. Asterisk-sounds is the default set of sounds on digium's website?
No. The default sounds are in the Asterisk distribution itself. The
asterisk-sounds package is separate, and none of the built-in
applica
Steve Underwood wrote:
More than that, in their fine print some only claim to pass maybe two or
three of the tests. There is nothing that defines what you must achieve
before you can claim G.168-2002 compliance.
Well, isn't that just wonderful :-) Standards are amazing things, from a
marketi
Kevin Collins wrote:
Any More news on this from Kevin ?
The only news is that I have not had time to work on it since last week.
However, this is the development trunk. You should _not_ be running it
in production, and realistically there is no reason to be discussing
issues with it on this
Darren Sessions wrote:
Is there a way to retrieve the Call-ID from a call made using the 'Dial'
command on a SIP channel without CDRs (i.e. variable) ?
(sometimes I wonder why we write documentation)
doc/README.variables has ${SIPCALLID} documented to be exactly that.
_
Darren Sessions wrote:
If try and read in the SIPCALLID variable (which I already do on the
incoming call) after the dial, I still get the incoming call's call-id.
Your explanation could have been much clearer.
Are you saying that you initiate a dial, which succeeds, and then after
the call
Darren Sessions wrote:
Exactly.
That will be difficult. You cannot set a channel variable on the parent
channel from inside the child channel. At best, you can pass the parent
channel's name into the child channel, then in a macro run by Dial() you
can store the Call-ID into the database key
Hans Witvliet wrote:
> Does the TDM400 not only fits, but also functions in a 3.3V only slot?
>
>>From what i detected so far, is that some MOBO manufactures have
> pci-slots that provide 3.3 Volt AND 5.0 Volt, thus can handle all kind
> of cards.
The TDM400P and TDM2400P will work in any PCI or
Dov Bigio wrote:
> Any way, if any developers are reading this, I don't think that rotating
> asterisk logs is the best way to handle this problem!
> Maybe a more user-friendly message could be logged, infoming which file
> reached the 2.0GB.
Unfortunately when we receive SIGFSZ from the kernel,
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