I also get an error following the README when I run sh
build.
[EMAIL PROTECTED] iaxmodem-0.0.5]# sh buildiaxmodem.c: In
function `printtime':iaxmodem.c:139: warning: passing arg 4 of `strftime'
makes pointer from integer without a castiaxmodem.c: In function
`main':iaxmodem.c:671: warning:
hello all,
is there any free pocket pc softphone
Best regards/alfa
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Congratulations from Italy now back to work for 1.3 ! :)
--
Best regards,
Alessiomailto:[EMAIL PROTECTED]
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Although I posted a demo a couple of weeks back, we have a new release
of our management gui that has a lot more user friendly features and
has gone through a bunch of testing. Still no name for it as it's
mostly an internal project, but we will come up with something
asap. Right now I believe
Hi All!
My asterisk box has 2 low speed ADSL connections (using pppoe straight from
my [EMAIL PROTECTED] box). I've routed some IP ranges out over one of the ADSL
lines and the other is the default route.
The problem I have is that if one of the sip extentions that is routed over
the one
Ignore me, the build does complete.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee
ArcherSent: 17 November 2005 08:19To: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users]
receive fax with asterisk
I also get an error following the
is there any free pocket pc softphone
Try sjphone from http://www.sjlabs.com/sjp.html
Regards
Guido Hecken
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looks very nice but you get an error trying to loginto the voicemail system
--On Thursday, November 17, 2005 00:24:40 -0800 snacktime
[EMAIL PROTECTED] wrote:
Although I posted a demo a couple of weeks back, we have a new release of
our management gui that has a lot more user friendly
tor, 17,.11.2005 kl. 00.24 -0800, skrev snacktime:
Although I posted a demo a couple of weeks back, we have a new release
of our management gui that has a lot more user friendly features and
has gone through a bunch of testing. Still no name for it as it's
mostly an internal project, but we
On 11/17/05, Jan Saell [EMAIL PROTECTED] wrote:
looks very nice but you get an error trying to loginto the voicemail system
There seems to be a bug where if you were logged in as a user it gives
that error once then clears up. We just put that up today so
people would see that the voicemail login
Sorry but I had to disable the admin interface as people keep deleting
the demo user. All the good stuff is in the user interface
anyways, the admin interface is just to modify the scripts (which
arent' documented yet) and for deleting users (which seems to be the
'in' thing to do).
Chris
Hi all,
I've written a billing application with AsteriskJava (a great
package!!!) but I'm having a problem with identifying the call status
(call completed, fail, etc) - kind of important for billing.
My problem is that is seems like the return code from the exec(DIAL)
command tells me if
Thank you, but I have tried that... Then the To is:
To SIP:asterisk.tsip.lab
not SIP:[EMAIL PROTECTED] as I would like.
Any other ideas?
trond
Trond Andersen wrote:
Hi.
I have a small problem and was hoping for some pointers in the right
direction...
Try:
exten =
Is there a way to detect (via batch) if asterisk is idle i.e. is there no
active channels ? (oh323 show channels via console)
I need to reboot every day an asterisk box, but I would like to do that
only when asterisk is not doing anything.
So I would like to schedule a batch at a given time
Boris Bakchiev wrote:
I think someone needs to start some sort of wiki that everyone can enter
the details of they systems J
There is already
http://www.voip-info.org/wiki-Asterisk+hardware
probalby we should add it to this page, exactly specifying the Server HW
+ the used interface cards.
On 11/17/05, Henning Kilset Pedersen [EMAIL PROTECTED] wrote:
tor, 17,.11.2005 kl. 00.24 -0800, skrev snacktime: Although I posted a demo a couple of weeks back, we have a new release of our management gui that has a lot more user friendly features and has gone through a bunch of testing.Still no
Hi all,
I have one external VoIP terminator, I need to forward
all calls to that terminator i did some configuration
in sip.conf but i am confiused what will be the
configuration in extentions.conf to forward all calls
to that terminator.
sip.conf
[general]
register =
looking for ip phones for an office setting. The client
wants about 15 phones initially. Not counting volume
discounts, does anyone have any
recommendations. Cost is a factor, after discounts they
were thinking
about $50/phone.
I've no idea what prices are like on the GXP-2000 on
Hi all,
Looking at the asterisk bug tracker i saw a feature assigned by mattf
that he issued a patch in order HDLC to be applied on hardware zaptel
boards. http://bugs.digium.com/view.php?id=5313
Is this implemented in the stable 1.2 version of Asterisk? Applying this
patch will help with
Let me see if I got that right.
When an iax user is online, the phone rings and if not picked up, then it
goes to voicemail.
But if the iax user is not online, you get an unknown extension error?
If so, the problem and solution is on your dialplan, I suggest before doing
the DIAL on your
Drumroll
TADA!!
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Asterisk Development Team
|Sent: Wednesday, November 16, 2005 11:49 PM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Asterisk 1.2 Released!
|
|We are
How about a cron job that does:
asterisk -rx restart when convenient
I do this sometimes and does the trick.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|[EMAIL PROTECTED]
|Sent: Thursday, November 17, 2005 3:21 AM
|To:
Single port TE110p and quad port TE410p.
Craig
- Original Message -
From: Klaus Darilion [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, November 17, 2005 1:34 AM
Subject: Re: [Asterisk-Users] dell and
Any word on when 1.0.10 will be out? I saw mention that 1.0.10 would be
released concurrently with 1.2 sometime last week. I've got some issues I
am hoping 1.0.10 will help solve.
Craig
- Original Message -
From: Asterisk Development Team [EMAIL PROTECTED]
To:
On Thursday 17 November 2005 12:32, Anton Krall wrote:
Drumroll
TADA!!
Already compiling it :)
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On Thursday 17 November 2005 12:58, Dmitry Ivanov wrote:
On Thursday 17 November 2005 12:32, Anton Krall wrote:
Drumroll
TADA!!
Already compiling it :)
Unlike beta2, it works for me. I hear no noise during echo test.
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I have a problem with chan_capi-cm-0.6.1.
It works when I start * directly from the command
line with asterisk vvvgc, but not when I use
/etc/init.d/asterisk script.
I have the log bellow in /var/log/asterisk/messages :
Nov 17 11:26:43 WARNING[5337]: CAPI not installed,
CAPI
does it include the patch for VAD?
( dropping extra frame of G.729 since we already have a VAD frame at the end )
Mario
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Hi all,
I am looking for SIP hard phones to use in a call center.
The feature that I need the most is quick change of logon credentials as we
run 3 shifts. each agent will have their own extension number and password.
any suggestions would be greatly appreciated.
thank you
John Fraser
Hi,
what's the current status of jb implementation in chan_sip? Are there
any patches out there available to be applied to the brand new 1.2-release?
Cheers,
Simone.
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Asterisk guy wrote:
does it include the patch for VAD?
( dropping extra frame of G.729 since we already have a VAD frame at the end )
It does not include several important things. It does not include a SIP
jitter buffer. It does not include the ability to use Zaptel for timing
of the
Hi,I'm now using Asterisk for my voicemail together with SER. They just work fine. When the user in SER is not registered the call will be forwarded to Asterisk and the caller will record his message. Then I also made asterisk to send the wav as attachment to its email. I try using two ip
How to set asterisk in pass-through mode ?
could you give a sample configure for passthrough?
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Hello,
I would like to know if there is a way in IAX2 and SIP to tell a client
to register at a different server.
For example:
Client tries to register at server B but server B answers with some sort
of redirect to tell the client to register at server C. The client then
tries to register
Title: 1.2 chan_modem not installing?
After compiling the released version I get the follow error when I run asterisk. I didnt get the fault with the beta's or rc's using the same config. I have tried the FTP version and the CVS downloaded version and get the same error
Nov 17 11:53:50
does the following patch work for 1.2? how to apply it to 1.2? ( I
am not a programmer, don't know how to use .diff file).
http://bugs.digium.com/view.php?id=5374
silence-suppression-2.diff
On 11/17/05, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Asterisk guy wrote:
does it include
Hi,
I have a rather interesting problem with my Asterisk setup at the moment,
and was wondering if anybody could shed any light on it!
The system is initiated by placing a call file into
/var/spool/asterisk/outgoing. This file calls asterisk, so it is calling
itself.
The process then goes on
They are coming, expect to have something for testing in less than a week.
Zoa
Simone Ricci wrote:
Hi,
what's the current status of jb implementation in chan_sip? Are there
any patches out there available to be applied to the brand new 1.2-release?
Cheers,
Simone.
On Thu, 17 Nov 2005, Amaury BOSSE wrote:
I have a problem with chan_capi-cm-0.6.1.
It works when I start * directly from the command line with asterisk
-vvvgc, but not when I use /etc/init.d/asterisk script.
I have the log bellow in /var/log/asterisk/messages :
Nov 17 11:26:43
hello bastian,
you could use the patch i made http://bugs.digium.com/view.php?id=5014
frank
Bastian Schern schrieb:
I'm using the snom Phones together with Asterisk and I already able to
see which Peer is used via hint priority. Then a LED on the snom phone
is blinking. But I don't see who
Jason Pyeron schrieb:
On Wed, 9 Nov 2005, Olle E. Johansson wrote:
That is not supported yet. There is a patch in the issue tracker that
does this, but it's a proof-of-concept code. It will burden your
asterisk quite a lot if you put it to use in larger production sites.
Which issue are you
Lee Archer wrote:
After compiling the released version I get the follow error when I run
asterisk. I didn’t get the fault with the beta's or rc's using the
same config. I have tried the FTP version and the CVS downloaded
version and get the same error
Nov 17 11:53:50 VERBOSE[21915]
So you are having 3 card (TE110p + TE110p + SATA-RAID) in your Poweredge
850? AFAIK the 850 has only 2 slots. ?
klaus
Craig Guy wrote:
Single port TE110p and quad port TE110p .
Craig
- Original Message - From: Klaus Darilion
[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Hi,
I have been tryingchan_bluetooth module
for asterisk during last week. I found some difficulties configuring it, due
mainlyto my ignorance and secondly to the lack of documentation. Thanks to
the listI have managed to configurethe Audio Gateway modeand I
have a strong doubt about how
Single port GSM Gateway support 900 / 1800 GSM mode with external antenna.
Brand new unit and just been out for testing only.
Extremely easy to setup and can be used out of the box without any
configuration. So should be good alternatively of phonecell or nokia pbx
etc..
Units are located in UK
send one over
i'll wire it
On Thu, Nov 17, 2005 at 12:50:29PM -, Sam Tam wrote:
Single port GSM Gateway support 900 / 1800 GSM mode with external antenna.
Brand new unit and just been out for testing only.
Extremely easy to setup and can be used out of the box without any
[EMAIL PROTECTED] schrieb:
Is there a way to detect (via batch) if asterisk is idle i.e. is there no
active channels ? (oh323 show channels via console)
use asterisk -rx 'show channels' to execute an console command from
batch and evaluate the output.
I need to reboot every day an
From an asterisk cli:
asterisk1*CLI help restart
restart gracefully Restart Asterisk gracefully
restart now Restart Asterisk immediately
restart when convenient Restart Asterisk at empty call volume
That means, if you schedule the command:
asterisk -rx 'restart when
I did download again but the problem was that chan_modem was # out of
the Makefile so wasn't made. I removed the # and rebuild and its
running fine now.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 17 November 2005 12:38
Hi, somebody has implemented Asterisk in one organizacion with amount of
extenciones in the order of 20.000?
Thanks.
--
Ing. Darío Colombo
Ingeniería de Red
Departamento Ingeniería de Comunicaciones
AFIP
Yes you are right;
I was considering asterisk -rx stop when convenient (remember I
have to shutdown the box).
But the poit is: I don't know what exactly this command does !!
Does it stop accepting new calls ? If it is right, then I can stay for
several hours (let's say I have 100 calls
The problem is that my IAX softphone only receives calls sporatically.I am using both Firefly and Diax (not at the same time!) to connect from inside my office network (behind NAT) to my Asterisk (1.0.9) box in my home (behind NAT.) The softphone registers as an extension no problem and
On Thu, 2005-11-17 at 12:46 +, Victor Alvarez wrote:
Hi,
I have been trying chan_bluetooth module for asterisk during last
week. I found some difficulties configuring it, due mainly to my
ignorance and secondly to the lack of documentation. Thanks to the
list I have managed to
On Thu, 2005-11-17 at 10:06 -0300, Dario M. Colombo wrote:
Hi, somebody has implemented Asterisk in one organizacion with amount of
extenciones in the order of 20.000?
In one building?
--
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Hi Derek,
I don't think AGI-only is the best approach for billing.
You can easily use the Manager API for that (there you get Link and
Unlink or CDR events that you can process much better).
Using Asterisk-Java you can quite easily combine AGI and the Manager
API.
=Stefan
signature.asc
Hi,
I have a long delay when detecting hangups on the TDM400P card, with 4
FXO ports,
When an incoming call dial's in, when hanging up, the asterisk will
detect the hangup only after 10 seconds, i searched around, and found
many similar problems, but no solution, i tried some options in
cd top level asterisk source directory (where UPGRADE.txt is)
patch -p0 /path/to/silence-suppression-2.diff
On Thu, 2005-11-17 at 07:07 -0500, Asterisk guy wrote:
does the following patch work for 1.2? how to apply it to 1.2? ( I
am not a programmer, don't know how to use .diff file).
Tyler wrote:
Anyone using app_icd? I need to use some of the advanced features that
the regular asterisk Queue() application won't provide. Anyone have any
configuration examples, etc? Will it work with the current 1.2rc
release?
I played around with ICD in August. I was generally impressed
I've also had some luck with Microsoft Portrait
Guido Hecken wrote:
is there any free pocket pc softphone
Try sjphone from http://www.sjlabs.com/sjp.html
Regards
Guido Hecken
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What jitterbuffer issues are you having with connecting to 1.0.x servers?On 11/17/05, Eric ManxPower Wieling [EMAIL PROTECTED]
wrote:Asterisk guy wrote: does it include the patch for VAD?
( dropping extra frame of G.729 since we already have a VAD frame at the end )It does not include several
Hi,
Just yesterday I got an amber light on my PowerEdge 2850 saying PCI
Parity Error EB113
The on-screen message says:
Uhhuh. NMI received. Dazed and confused, but trying to continue
You probably have a hardware problem with your RAM chips
Dell is saying that I must have two devices on the same
Title: IAXmodem
Hi, I wonder if you can give me some pointers please. I have hylafax running, I've tested it with a modem off the serial port so I know the install does work, and I've installed IAXmodem to be able to fax out via asterisk. I've set everything up as in the README that comes
I think that if you store the Dial Plan in a database instead of a flat
file, there is no problem with the amount of extensions. Is this Ok?
Sixto Diaz
- Original Message -
From: Dario M. Colombo [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, November 17, 2005
On Thursday 17 November 2005 07:24, Zoa wrote:
They are coming, expect to have something for testing in less than a week.
Why is the SIP jitter buffer not just using Steve Kann's excellent generic
jitter buffer that was implemented in IAX2? It's already generic enough to
use with pretty much
Bug number 5197 I'm not experiencing the problem, but I expect to. The
bug is not closed and no indication it's 1.2.0.
--Eric
Pedro wrote:
What jitterbuffer issues are you having with connecting to 1.0.x servers?
On 11/17/05, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Asterisk guy
Asterisk Development Team [EMAIL PROTECTED] wrote:
We are proud to announce that Asterisk 1.2.0 has been released!
That is great. I might get a chance to try it out later today.
(Note: for a short time, a tarball of Asterisk 1.2.0 was present on the
FTP servers with a build problem related to
In article [EMAIL PROTECTED],
Lee Archer [EMAIL PROTECTED] wrote:
I did download again but the problem was that chan_modem was # out of
the Makefile so wasn't made. I removed the # and rebuild and its
running fine now.
chan_modem has been obsoleted in 1.2. Unless you are specifically using
On Wednesday 16 November 2005 20:17, Paul wrote:
On my home system, I can tweak all I want and I cant kill the echo on my
pots line, but my VOIP line from Vonage does eliminate most if not all of
it for me regardless of the * settings!
Not strange at all. Vonage likely has very very good
We will check that... but that would have affected us in 1.0.1.9
correct?
Im inclined to believe this is a phone problem less an * prob. I dont
understand the changing from pcmu to gsm while on hold
can someone explain if it is supposed to work like this.
Does the community have any influence
Hi,
I am having voice quality problems with SJPhone under certain
conditions. Setup is Fedora 4 with Asterisk 1.2rc2 and Digium TDM 400P
(2FXO's and 2 FXS's).
SJPhone = outside line = echo's, scratchy
ZyXel P2000WV2 = outside line = clear as a bell
SJPhone = recording voice mail = clear as
Matt ha scritto:
Hi,
Just yesterday I got an amber light on my PowerEdge 2850 saying PCI
Parity Error EB113
The on-screen message says:
Uhhuh. NMI received. Dazed and confused, but trying to continue
You probably have a hardware problem with your RAM chips
I solved it putting the digium
Sixto Diaz ha scritto:
I think that if you store the Dial Plan in a database instead of a flat
file, there is no problem with the amount of extensions. Is this Ok?
Sixto Diaz
- Original Message -
From: Dario M. Colombo [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Has anyone else on list tried this yet? I bought the personal edition
and would like to compare experience with others. The vendor is
www.rsdevs.com.
In general the application does what it promises. It receives Skype
calls through an actual Skype client, then uses a virtual audio patch
cord to
I did download again but the problem was that chan_modem was # out of
the Makefile so wasn't made. I removed the # and rebuild and its
running fine now.
chan_modem has been obsoleted in 1.2. Unless you are specifically using it,
you should edit /etc/asterisk/modules.conf to prevent it
Ah that's why then. Is there something else I should set in my
modules.conf?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: 17 November 2005 14:00
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re:
I have followed the recommendations. After some further tweaking, this is the most I've been able to get:[EMAIL PROTECTED] zaptel-1.2.0]# ./zttest -vOpened pseudo zap interface, measuring accuracy...8192 samples in 8190 sample intervals 99.975586%8192 samples in 8191 sample intervals
It would not have affected you in 1.0.1.9 if you set the DTMF mode to
another setting. Mine all got reset when I upgraded to 1.0.12 and 3
of the phones I used had to be factory reset and then apply the .12
for them to work properly.
On 11/17/05, Health Masters [EMAIL PROTECTED] wrote:
We will
I have been trying chan_bluetooth module for asterisk during last
week. I found some difficulties configuring it, due mainly to my
ignorance and secondly to the lack of documentation. Thanks to the
list I have managed to configure the Audio Gateway mode and I have a
I've set it up the way you described, and my second span is still
giving a red alarm (i.e. not usable).
On 11/16/05, Trey Blancher [EMAIL PROTECTED] wrote:
Thanks. I was just unsure about referring to the second span as
channels 25-47, since my provider refers to them as separate banks of
Rich Adamson wrote:
I did download again but the problem was that chan_modem was # out of
the Makefile so wasn't made. I removed the # and rebuild and its
running fine now.
chan_modem has been obsoleted in 1.2. Unless you are specifically using it,
you should edit /etc/asterisk/modules.conf to
I wish I lived in a 900/1800 country; I'd be all over this!
Unfortunately I am not ready to move to Europe over a £60 GSM gateway, but thanks anyway.
-RustyOn 11/17/05, Sam Tam [EMAIL PROTECTED] wrote:
Single port GSM Gateway support 900 / 1800 GSM mode with external antenna.Brand new unit and
Hi,
I have a problem with the Caller ID string, seems like asterisk will
display only 10 digits of the caller id.
If the string is longer then 10 digits, asterisk will sometimes strip
the first digit, and sometimes the last digits, in order to show a
10-digit callerid,
Is this
Dario M. Colombo wrote:
Hi, somebody has implemented Asterisk in one organizacion with amount
of extenciones in the order of 20.000?
Thanks.
2 in 1 building? WOW that's a HELL of a PBX
the maximum that we are achive in 1 BUILDING is 1000 extension
regards
Saul
Lee Archer wrote:
Hi, I wonder if you can give me some pointers please. I have hylafax
running, I've tested it with a modem off the serial port so I know the
install does work, and I've installed IAXmodem to be able to fax out
via asterisk. I've set everything up as in the README that comes
I did download again but the problem was that chan_modem was # out of
the Makefile so wasn't made. I removed the # and rebuild and its
running fine now.
chan_modem has been obsoleted in 1.2. Unless you are specifically using it,
you should edit /etc/asterisk/modules.conf to prevent it
Thanks Stefan,
This is what I'll do
Derek
Stefan Reuter wrote:
Hi Derek,
I don't think AGI-only is the best approach for billing.
You can easily use the Manager API for that (there you get Link and
Unlink or CDR events that you can process much better).
Using Asterisk-Java you can quite
We also have CDMA gateway. Do contact us
for more info if you are interested.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rusty Dekema
Sent: 17 November 2005 14:45
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GSM
Hi, somebody has implemented Asterisk in one organizacion with amount
of extenciones in the order of 20.000?
If you intend to implement an PBX with 20.000 phones attached you have
to estimate the number of simultaneous connections.
I would estimate this to be the bottleneck, not the
I'm only showing channel 8 in my grep because that's one of the less
used channels.
[EMAIL PROTECTED] libpri-1.2.0-rc2]# grep reset /etc/asterisk/zapata.conf
resetinterval=3600
/var/log/asterisk/messages:
Nov 17 09:00:31 VERBOSE[2727] logger.c: -- B-channel 0/8
successfully restarted on
Hi,
If there are two channels (SIP or/and IAX) talking to each other
(bridged at Asterisk), using the same codec. Does Asterisk decode and
recode again?
By,
Roger.
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Oh, you do? How much did you have in mind for that? What band(s) does
it run on? Alternatively, if you have any 800 (850) or 1900 MHz GSM
gateways, I'd be interested in hearing about it too.
Thanks,
RustyOn 11/17/05, Sam Tam [EMAIL PROTECTED] wrote:
We also have CDMA gateway.
(Sorry; meant to reply privately.)
-RustyOn 11/17/05, Rusty Dekema [EMAIL PROTECTED] wrote:
Oh, you do? How much did you have in mind for that? What band(s) does
it run on? Alternatively, if you have any 800 (850) or 1900 MHz GSM
gateways, I'd be interested in hearing about it too.
Thanks,
Marcus Deluigi (intern) wrote:
Great!
Is there any chance someone tries to build a debian package for it?
I took the debian testing source for rc2 and easily built packages for
debian stable. I imagine this release will appear in the debian archives
soon and I can do the same with it. Here is
Matt Riddell wrote:
So, is 1.2 the new stable now? Does that mean that CVS Head is 1.3 with the
next STABLE being 1.4?
It means that people can start posting when will 1.4 be released? :)
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I have in my iax cfg files...
[EMAIL PROTECTED] asterisk]# more iax.conf
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
mailboxdetail=yes
#include
Did you try using Sjphone with allow=alaw in sip.conf instead of g729?
In our Asterisk Installations we used:
...
allow=alaw
allow=ulaw
allow=gsm
...
and didn't have any echo or scratchy problems.
Hope it helps...
Regards
Guido Hecken
-Ursprüngliche Nachricht-
Von: Chuck Bunn
Hello Victor.
I had the same problem, but when I compile a new version of
chan_bluetooth (ones form august) it works.
Try to download and compile a new version. I`m using asterisk 1.0.9.
Regards,
José Luis
El jue, 17-11-2005 a las 14:39 +, Victor Alvarez escribió:
I have been
I work for a company that is nearing the end-of-life on its existing
Nortel Meridian switch and is considering Asterisk. We have
approximately 200 existing extensions, and probably 150 out of those 200
are using basic analog phones and would stay that way. The rest would
have VOIP phones at the
I figure this topic deserves a new subject
I know there are lots of good things that work with 1.0.x but how about
some status reports for 1.2?
I am sure we might see that some apps work in a compatibility mode. By
that I mean if you just moved your existing config from 1.0.x to 1.2 it
works but
Hi,
I'm trying to setup Sipura to work with Siemens Combiset 1009 on PSTN line
(GSM gateway - produced for germany)..
I have two problems:
- Sipura doesn't detect Caller ID
- Sipura doesn't detect hangup condition
I have 3.1.7(GWg) firmware on Sipura and Asterisk 1.0.9...
Anyone has
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