James Noble a écrit :
Thank you for the heads up. I will look into both weephone and voipover3g
I think siax -from cydia- could also be an alternative as they stated to
use natively 3g. I only test WIFI.
--
Daniel
___
-- Bandwidth and
Carlos Ruiz Diaz a écrit :
Check chan_mobile.
[...]
Or use GSM gateway
On Thu, Jul 2, 2009 at 3:20 PM, Nick Hill t...@nickhill.co.uk wrote:
I have had a search for this, but didn't come up with any results, so maybe
I am
using the wrong terms, sorry if this is an FAQ.
For those
Hi all,
we face a problem with SMS reception sended to _landlines_, at least in
France.
Normally operators -tested with France Telecom and SFR- are sending
voice SMSs from a particular CID number, so no problem. But today we
discover that -at least SFR- send from time to time voice SMSs with
Hi
Charles Solar a écrit :
Hi guys, I am new here but I have a quick question.
I have an incoming trunk that sends me calls from various usernames I have
with them. Only trouble is they send invites as s...@my.ip.addr, not as the
username I have with them. So I cannot match extensions like
Hi all,
I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from
my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so
I can use the GW. For this I use:
exten = s,1,NoOp(One of our workers (${CALLERID(number)}) is calling
office) ;callerID is the
Cory Andrews a écrit :
Anyone using Nokia E Series handsets with Asterisk? I'm trying to
deploy some e71's and am having an issue. I can get a single handset
working, but when I try to create a SIP profile on the second phone, it
won't allow me to save the profile, saying that devices in the
can we get the dtmf type?
For me it looks like a bug.
Thanks for your help.
2009/5/11 Administrator TOOTAI ad...@tootai.net
Hi all,
I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from
my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so
I can use
Hello,
in our dialplan we have some variables containing datas from our
customers calls like for instance the called number.
Now, when caller make a transfer, we would like to catch the new called
number. How to get this? Is there a return context _after_ transfer and
*before* the call is
Hello,
I opened bug #0014707 concerning audio missing between Asterisk and
Gtalk. I would like to know if someone uses successfully Gtalk with
Asterisk 1.4.24?
Regards
--
Daniel
___
-- Bandwidth and Colocation Provided by
Guillermo Salas M. a écrit :
El mié, 25-03-2009 a las 19:09 +0100, Administrator TOOTAI escribió:
Can be used to receive calls from skype?
Yes
Great,and how? Have you any link to read?
http://www.gizmo5.com/pc/opensky/
--
Daniel
Michael Robertson a écrit :
Anyone connected up to it yet?
http://www.skypeforsip.com/
This service is vaporware. It's just surveyware at this point with no actual
service. An alternative is OpenSky which is a launched service which does
SIP to Skype and Skype to SIP so you can
Stefan Guenther a écrit :
Hello,
is anyone on the list using a normal cell/mobile phone which is able to
act as a SIP client over WLAN?
Or has anyone heard of a SIP client for cell/mobile phones running
windows mobile 6.x?
The phone should use SIP, when the asterisk server is reachable
Guillermo Salas M. a écrit :
El mié, 25-03-2009 a las 10:28 -0700, Michael Robertson escribió:
OpenSky can be setup for free to allow any Asterisk system to call
Skype users. Setup instructions for Asterisk are at:
http://www.gizmo5.com/opensky Free calls are available up to 5
minutes. If
Hi,
sorry for this a bit OT.
I'm using VoiceTrading for some calls -premium route- and can't get CID
to work despite the fact that CALLERID(num) and CALLERID(name) are setted.
I ask in VT-myAccount to accept calls from my IP without checking
username secret. On incoming calls the CID is
John Knight a écrit :
make[1]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 »
WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers
is missing; modules will have no dependencies and modversions.
specifically Symbol version dump
Tzafrir Cohen a écrit :
On Tue, Mar 17, 2009 at 07:16:26PM +0100, Administrator TOOTAI wrote:
Hi,
We installed the latest 1.4.24 on a test machine and can't get zaptel
nor dahdi compile. It's a Linux Debian Etch. Errors we have:
keewi:/usr/src/dahdi-linux-2.1.0.4# make
make -C /lib
Tzafrir Cohen a écrit :
On Tue, Mar 17, 2009 at 07:16:26PM +0100, Administrator TOOTAI wrote:
Hi,
We installed the latest 1.4.24 on a test machine and can't get zaptel
nor dahdi compile. It's a Linux Debian Etch. Errors we have:
keewi:/usr/src/dahdi-linux-2.1.0.4# make
make -C /lib
Hi,
We installed the latest 1.4.24 on a test machine and can't get zaptel
nor dahdi compile. It's a Linux Debian Etch. Errors we have:
keewi:/usr/src/dahdi-linux-2.1.0.4# make
make -C /lib/modules/2.6.18-custom.2/build ARCH=i386
SUBDIRS=/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi
Gordon Henderson a écrit :
Anyone here used these phones?
We stopped to use them, not stable IAX or SIP. A customer ask us to
exchange 26 pieces after 6 month battling to make the phones working. We
did it.
[...]
--
Daniel
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Klaus Darilion a écrit :
Hi!
Hallo
I have a setup with Asterisk in front of a PBX connected with ISDN to
the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing
ENUM for outgoing calls and allows incoming calls per SIP.
Recently the IP connectivity for this location was
Rob Hillis a écrit :
Administrator TOOTAI wrote:
[MyPeer]
host=xxx.xxx.xxx.139
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142
permit=yyy.yyy.yyy.yyy/255.255.255.255
On incoming calls, when the peer address is the one terminating with
.139
Hi all,
I tested with few Asterisk versions from 1.4.18 to 1.4.21, same result.
Here is the problem: I have a peer -which is peer AND user- setted up
like this
[MyPeer]
;
type=peer
host=xxx.xxx.xxx.139
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to
Laurent a écrit :
Hello,
HI
I believe that one of the most comprehensive resources, in terms
of numbering plans, is on the ITU website :
http://www.itu.int/oth/T0202.aspx?parent=T0202
[...]
As regards France [...]
The document for France is out of date, eg +33 87x xxx xxx ar no more
Hi,
Ii try to connect an Asterisk server running 1.4.21.2 version with
gtalk2voip services. Everything is fine till the call for DTMF test:
there is no audio and Asterisk shows
[Nov 18 14:51:47] WARNING[20502]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL
Administrator TOOTAI a écrit :
Hi,
Ii try to connect an Asterisk server running 1.4.21.2 version with
gtalk2voip services. Everything is fine till the call for DTMF test:
there is no audio and Asterisk shows
[Nov 18 14:51:47] WARNING[20502]: chan_sip.c:1950 retrans_pkt: Maximum
retries
bilal ghayyad a écrit :
[...]
What about Nokia Communicator? Any other Nokia Family that accept to download
fring on it?
Why do you want to use fring on a Nokia as they have a very good SIP
client ?
--
Daniel
___
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Olivier a écrit :
Hi,
Good day
I've got this case :
When the last staff member is about to leave and lock offices, he would like
to notify everybody Offices are about to be closed so that (s)he wouldn't
lock anybody in.
Which is the smartest way to do it ?
I thought of either :
1.
michel freiha a écrit :
Hi all,
Hi
I have the below extension defined under sip.conf:
[2203]
type=friend
username=2203
secret=123456
host=192.168.0.164
mailbox=2203
context=intern
canreinvite=yes
dtmfmode=rfc2833
When trying to register from a softphone installed on a PC behind a
Tilghman Lesher a écrit :
On Tuesday 01 July 2008 10:48:55 Tilghman Lesher wrote:
On Tuesday 01 July 2008 09:20:52 Administrator TOOTAI wrote:
does anybody know how to cut a chain using the pipe delimiter? I tried
to escape it or to use $'x7c' as delimiter, no luck.
1.2 does
Hi all,
does anybody know how to cut a chain using the pipe delimiter? I tried
to escape it or to use $'x7c' as delimiter, no luck.
Thanks for any help.
--
Daniel
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AstriCon
Steve Kennedy a écrit :
[...]
Are the same rules and conditions that exist here in the States
mirrored elsewhere?
How does a person in Europe go fully VoIP and still keep the main
number?
In the UK numbers are portable, though the telco wanting the number must
have a
Hi all,
on a new installation, HP [EMAIL PROTECTED],83 with 2.6.18-amd64 image from
Debian
Etch with Asterisk 1.4.20.1 and zaptel 1.4.11 -both tar.gz from digium-
we get:
[Jun 11 14:54:29] ERROR[6686]: asterisk.c:2952 main: You have Zaptel
built and drivers loaded, but the Zaptel timer test
Tzafrir Cohen a écrit :
On Wed, Jun 11, 2008 at 02:00:50PM +0200, Administrator TOOTAI wrote:
Hi all,
on a new installation, HP [EMAIL PROTECTED],83 with 2.6.18-amd64 image from
Debian
Etch with Asterisk 1.4.20.1 and zaptel 1.4.11 -both tar.gz from digium-
we get:
[Jun 11 14:54:29
Hi all,
we are running Asterisk SVN-branch-1.4-r114299 and face following
problem: we have a main extension (102) and other extensions (104 and 107)
When extension 104 is calling extension 107 and 107 is on the phone, the
caller is parked for 5 seconds (Park-And-Announce) and the going back to
bilal ghayyad a écrit :
Hi All;
Till now I am not able to find a good IAX IP Phone or
Gateway that can be used with good quality.
Anyone can advise for good one?
We are selling IP0023 and IP0027 phones (IAX and SIP). Please contact
off line if you're interested.
--
Daniel
Good morning,
we face a problem with Atserisk 1.4.18.1 and Zaptel 1.4.9.2: calls are
frequently ended during conversation or voicemail are not registring the
entire messages given by callers.
What we have -and seem strange- is:
Module Size Used by
ztdummy
Chris Bagnall a écrit :
I had same problem in france, not much choice, i have ordered from germany
http://www.voipango.de
Unless I'm missing something, that looks like a hardware supplier. I'm
looking for someone to provide the physical lines (i.e. a Swiss Telco), but
I've no
Hi,
we are using the app_nvfaxdetect from Newman Telecom with Asterisk 1.4
and tried to build the trunk/next release 1.6 with this application, but
it failed (We are using fax stuff with iaxmodem/Hylafax).
I remember that we had the same issue switching from 1.2 to 1.4 and
someone made the
Matthew J. Roth a écrit :
Administrator TOOTAI wrote:
This is not true if you're using B410P cards. We always face timing
problem as we can't -Asterisk stability issues- add X100P or TDM400P
with those cards
Daniel,
I thought that using an empty TDM400P as a timing source may
Matthew J. Roth a écrit :
[...]
I settled on using an empty TDM400P as a timing source, because it is a
simple solution that just works. This may still be your best bet, but
I'll defer judgment on that to the list because Asterisk has evolved
quite a bit since I made that decision.
This
Hi,
we have an Debian Etch 4.0 amd64 server with 2 B410P cards. Asterisk SVN
r99777 is installed. We tried with mISDN shipped with Asterisk/Zaptel
(make b410p) as well as with the latest version from mISDN.org 1.1.7.2.
zaptel, ztdummy and crt-ccitt modules are loaded. Output of /dev/zap is:
bilal ghayyad a écrit :
Hi All;
Anyone can advise for a good IP Phone that has the
ability to support SIP firmware and IAX firmware?
Ofcourse, SIP there is a lot, but we need also the
ability to use IAX (as it is good for NAT).
We are using IP0027. Great audio, POE, 5 SIP accounts, 1
William Herrera a écrit :
Yes I did ...
The real problem I am having with it is voicecode.
The phone claims to support several codes but in reality it does not give
any trouble when using iLBC. It will give you trouble with configuring with
alaw or ulaw...
Hmm,
we're using them with
William Herrera a écrit :
Hello to you all.
Hi William
Just got my first iP0020 phone and no matter what I do to
it when I try to call I get a busy signal even though Asterisk and the phone
web gui shows that the phone is registered.
Has any body had any similar experience with this type of
William Herrera a écrit :
Sip show peers will show the phone connected.
Ok, but sip show peer IP0020 does show good datas aso? Do you use dhcp
or fixed IP? Did you run with sip set debug peer IP0020? What version
of Asterisk?
Asterisk 1.2 asked the phone to have a fixed IP, with 1.4 it's
? Do you mean we should add a parameter in
/etc/misdn-init.conf or /etc/asterisk/misdn.conf?
Meanwhile, we only have the problem with Siemens phone, others brand are ok.
Thanks for your help.
On Wed, 2007-11-28 at 09:47 +0100, Administrator TOOTAI wrote:
Good day all,
we have following
Good day all,
we have following setup: Debian Etch 64, Asterisk SVN-branch-1.4-r66244
with mISDN 1.1.3 and 2 Digium cards B410P. Our customers calls in the
office through ISDN lines and then get a possibility to join meetme
conference. It works well except when customers are using SIEMENS
Kyle Sexton a écrit :
Does anyone know who really makes this phone:
http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/
Not so mysterious: we import those phones in Europe ;-) POE, 5 accounts,
SIP and IAX able, nice audio Good product.
--
Daniel
Hi all,
I receive this error while compiling chan_mobile:
gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c
chan_mobile.c: In function `mbl_load_config':
chan_mobile.c:1745: erreur: trop d'arguments pour la fonction «
ast_config_load »
make[1]: *** [chan_mobile.o] Erreur 1
make[1]: Leaving
Jason Parker a écrit :
Administrator TOOTAI wrote:
Hi all,
I receive this error while compiling chan_mobile:
gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c
chan_mobile.c: In function `mbl_load_config':
chan_mobile.c:1745: erreur: trop d'arguments pour la fonction
Jason Parker a écrit :
Administrator TOOTAI wrote:
Jason Parker a écrit :
Administrator TOOTAI wrote:
Hi all,
I receive this error while compiling chan_mobile:
gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c
chan_mobile.c: In function `mbl_load_config
Remco Barendse a écrit :
Has anyone ever tried using a Nokia phone with SIP client as channel for
Asterisk? I mean i would like to receive calls to the mobile on
asterisk and use the Nokia phone to place calls to cell destinations.
E70 and E65 are working perfectly as SIP client through
Gordon Henderson a écrit :
On Mon, 20 Aug 2007, Steve Totaro wrote:
Well chan_bluetooth is really amazing (especially if your phone does not
support SIP).
You connect your phone via bluetooth to your asterisk box and it becomes
a channel type. You can use it as an extension(FXS) or a
Good morning,
we actually have 2 Asterisks 1.2 running on one server each of them in a
XEN Dom and connected together with a IAX trunk. This setup allow us to
use our both public IP (different ISP's) and to have failover solution
in case of a problem on one of the ISP's line.
Is it a way in
Tzafrir Cohen wrote:
On Mon, Jul 02, 2007 at 09:27:39PM +0200, Administrator TOOTAI wrote:
We have an Ubuntu Dapper with 2.6.14 kernel, asterisk 1.2.14 debs from
http://pkg-voip.buildserver.net
Does it have misdn support?
We add misdn support by compiling misdn1.1.4
When
Matthew Fredrickson wrote:
On Jul 2, 2007, at 6:02 PM, Tzafrir Cohen wrote:
[...]
That's what I would say as well. Also, what's the output of dmesg
when you releoad the card.
Hi Matthew,
this is what we get after rmmod modprobe wctdm24xxp
[Jul 4 00:25:24] ERROR[20034]: Unable to
Tzafrir Cohen wrote:
[...]
Problems at the zaptel level.
cat /proc/zaptel/*
cat /etc/zaptel.conf
Here the outputs:
[EMAIL PROTECTED]:~# cat /proc/zaptel/*
Span 1: WCTDM/0 Wildcard TDM2400P Board 1
IRQ misses: 10
1 WCTDM/0/0 FXOKS (In use)
2 WCTDM/0/1 FXOKS
We have an Ubuntu Dapper with 2.6.14 kernel, asterisk 1.2.14 debs from
http://pkg-voip.buildserver.net
When misdn stuff (misdn-init start) is not started, everything is fine,
our 8 FXO (Channel 1-8) 4 FXS (21-24) are working well.
If we start the misdn stuff (one card, port 1,2,3,4 in
Matthew Fredrickson a écrit :
On Jun 1, 2007, at 4:20 AM, Steve Hanselman wrote:
We're also seeing the same thing, our calls are bridged zaptel calls
between ISDN30 PRI interfaces on a single TE410P.
We don't' appear to have any lost interrupts.
Same as stated, 2-3 second gaps in audio.
Steve Totaro a écrit :
Hi Steve
Your Zap conf files would be helpful. Zttest results? Cat
/proc/interrupts. Sharing interrupts?
No. Zap con files should not be relevant as we are using ISDN.
[EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /etc/zaptel.conf
loadzone = us
defaultzone=us
Morning all,
we face a strange problem on a dual Xeon server with 2 GB RAM and 2
B410P cards. On ISDN calls, audio is sometimes poor (micro
cutted/scratched) even cutted during few seconds (2 or 3). Problem
appears only on some calls and isn't reproducable.
Kernel is a Debian/Etch 2.6.17-4
Hi list,
we have a dual Xeon server with 2GB RAM running Debian Etch 2.6.18.4-686
kernel. The server has 2 B410P cards plugged in. No other card.
We installed Asterisk 1.4 trunk with zaptel trunk, ran make b410p, the
install mISDN1.1.0 (for bug 9064) configure and compile Asterisk with
Marco Mouta a écrit :
did you modprobe ztdummy?
Marco, thanks for your answer. The second way we tried -see end of
message- was with ztdummy, so yes, it was modprobed ;-)
On 3/30/07, *Administrator TOOTAI* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi list,
we have a dual
Jay Milk a écrit :
Doug Lytle wrote:
[...]
Thanks to Dave and Doug for the quick responses. I'm looking forward
to hearing the response on #3, but I think I'll get get one of these
devices to play with this weekend. At worst, it'll be a usable garage
or basement phone.
Hi Jay,
sorry for
marcotasto a écrit :
I did something similar one year ago for a friend [...] If you are interested,
I can post my results and the link to my site when they will be ready.
Yes please, would be great. Many thanks :-)
--
Daniel
___
--Bandwidth and
Hi list,
I installed a fresh Debian/Etch with Asterisk 1.4 and Zaptel 1.4 from
SVN for 2 Digium B410P card. I ran configure in Asterisk dir, went in
zaptel dir and: make, make install, make b410p. Everything is ok. Now I
want to compile Asterisk but can't activate the chan_misdn channel which
Hi,
does anyone have some informations on when the SVN repository of
digium.com will be synchronized again? Since few days we are sticked
with trunk #51363.
--
Daniel
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asterisk-users
Anton Krall a écrit :
Guys, anybody has seen or is using some kind of softphone on any square
screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they
do work on Wm5 but they are designed for standard screens, anybody using
anything on square ones?
We are using PPCIAX.
--
Vernier Umali a écrit :
[...] I do not have
any luck using nokia E61 (doesn't register and keeps on hanging). I
would think it's the same with all wifi enabled nokias.
Nokia E70 with latest firmware works perfectly.
I used an Ipaq
6900 series and Asus P55 and both worked well with SIP
Morning,
we have gateways with FXO port registered as SIP endpoint in Asterisk.
To be able to use this port, the gateway ask for prefix -lets say 9-
then send dial tone and here the user enter the calling number. We want
to cancel this step for the users so they can enter the entire number
Administrator TOOTAI a écrit :
[...]
FYI, dialing Dial(SIP/exten,,D(0)) give the dial tone, let the user
enter the calling number and the call is passing smoothly.
Sorry, please read Dial(SIP/exten,,D(9))
--
Daniel
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Anselm Martin Hoffmeister a écrit :
Am Montag, den 11.12.2006, 11:29 +0100 schrieb Administrator TOOTAI:
Administrator TOOTAI a écrit :
[...]
FYI, dialing Dial(SIP/exten,,D(0)) give the dial tone, let the user
enter the calling number and the call is passing smoothly.
Sorry
Michiel van Baak a écrit :
Hi,
Hello
Anyone here has any experience with the Nokia E70 and
asterisk ?
I read on the nokia website this phone is capable of talking
SIP and do Presence based on SIP/SIMPLE.
Please share your experience, I'm thinking of getting one
but want to be sure I can
Jean-Michel Hiver a écrit :
Hi List,
I am looking for a 1 FXO analog termination device, other than the
obvious PC + FXO card, and which can achieve decent call quality. The
SPA-3000 seems an option... have you got any other ideas?
Tiger G104 has PSTN to VoIP and vice versa. Didn't had time
Guillermo Salas M. a écrit :
On Wed, 2006-11-01 at 16:05 -0500, Vladimir Montealegre Estailes wrote:
Hello list partners
you know about a softphone made in java attachable in a web page?
GNU!
I'm using JIAXClient [1] to permit to any user to join one meetme room
[2] with the
Zoa a écrit :
Lets change the question to : does somebody know good iax phones, that
are ROHS compliant and without enormous delivery problems ?
ATCOM
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Daniel
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asterisk-users mailing list
Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router. I lost track of the
email thread, if anyone is presently working with this scenario please shoot
me an email.
Cory,
OpenWRT -running on Linksys WRT-
Guillermo Salas M. a écrit :
On Wed, 2006-10-18 at 20:08 +0200, Francesco Peeters (Asterisk) wrote:
On Wed, October 18, 2006 19:03, Paul Gaffney wrote:
Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm
looking for a NAT-friendly solution and my SIP phones are good
Gregory Duchatelet a écrit :
IAXcomm should. So should wengophone and mozphone.
And Kiax and Ekiga
--
Daniel
Ekiga not for Windows platforms...
http://snapshots.ekiga.net/win32/win32.php
--
Daniel
___
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Crazy Boy a écrit :
Hi,
I want to buy a phone. That phone must have two ports. One is Ethernet
port (to connect to my Asterisk server) and second is RJ11 port (to
connect with my traditional PSTN exchange). I searched in internet,
but unable to find this phone, which contains both feautre.
Tzafrir Cohen a écrit :
[...]
Did you know a good GPLed softphones which works on Windows ?
IAXcomm should. So should wengophone and mozphone.
And Kiax and Ekiga
--
Daniel
___
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Morning all,
We're looking for hand free solution to use with Asterisk beside BT
headsets. I was thinking on Sipura 841 but it seems that the headset
jack connector is not carrying voice (microphone), only audio.
Ideal would be a headset audio+microphone with RJ11 4p female that we
could
Hi all,
I'm running Asterisk SVN-trunk-r40489 on which one I have a Sipura 1001
connected. I face a problem when sending digits to voicemail password:
each one is sended twice (eg 35 give 3355) I have the same behaviour if
I have to enter the mailbox number before.
I have no problem to
Christian wrote:
Hi all,
Does anyone know a Softphone for Windows mobile 5? Want to connect to my
Asterisk when I am away.
We are using PPCIax
--
Daniel
___
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asterisk-users mailing list
To
Morning everybody,
After having a zaptel compile error in zttranscode.c with trunk version
of 28/07/2006 I updated with todays trunk branch. The error disappears
but get now another one. Asterisk core still compile fine as well as SVN
1.2 branch. It's a Debian SARGE running on 2.4.27 kernel.
Tzafrir Cohen wrote:
On Fri, Jul 28, 2006 at 02:04:10PM +0200, Administrator TOOTAI wrote:
Morning everybody,
I try to install an asterisk test server with trunk branch and get this
error when compiling zaptel. Asterisk core compile fine as well as SVN
1.2 branch. It's a Debian SARGE
Morning everybody,
I try to install an asterisk test server with trunk branch and get this
error when compiling zaptel. Asterisk core compile fine as well as SVN
1.2 branch. It's a Debian SARGE running on 2.4.27 kernel.
zttranscode.c: In function `zt_tc_mmap':
zttranscode.c:378: warning:
Attilla De Groot wrote:
Hi all,
I have two pda's and I want to be able to make calls, but I need a
client for this. The only problem is Windows Mobile 5.0, I can't find
a freeware client for this, the only one is Sjphone. But this one is
still beta for windows mobile and it just doesn't
Matt Gibson wrote:
Hi,
I'm experimenting with a little script here, and I'm tired of seeing
my tests in the callerid logs.
Is there a way to do something like the following:
exten = s,1,Answer
exten = s,n,DoNotLogCallData()
NoCDR()
--
Daniel
___
Hi all,
could someone tell me what this does mean bad file descriptor when
trying to start asterisk. It goes till the CLI command and then die with
this message. Below an strace output from asterisk -vc
It's on debian Sarge kernel 2.6.7 with packages from debian VoIP team.
The server
Hi all,
could someone tell me what this does mean bad file descriptor when
trying to start asterisk. It goes till the CLI command and then die with
this message. Below an strace output from asterisk -vc
It's on debian Sarge kernel 2.6.7 with packages from debian VoIP team.
The
WipeOut wrote:
Hi,
I am investigating getting a wifi VoIP phone because its may be a
better option than an ATA and a cordless phone..
Does anyone have any experience with the whats out there??
Do they support things like WPA etc??
I have heard the battery life can be a problem.. Is this
[EMAIL PROTECTED] wrote:
hello,
I have to test asterisk/gnugk is their somebody, sur
cette putain de liste, with a h323 terminal ?
No need to be aggressive like that, I don't think it will help your
request. And if you think what you wrote, feel free to unsubscribe.
--
Daniel
Información Capa Tres S.L. wrote:
Hello,
anyone has tested the TigerNetwork IPH202A or IPH202B Ip Phone ? I'm
very interested in known if the quality of this phones is OK, and if are
any problem with asterisk with this Phone.
I tested IP202 and ATA104, both are working well.
--
Daniel
Bjørn O wrote:
In my extensions.conf I’ve got an entry for the phone number that I’m
supposed to receive calls on:
[default]
Exten = 11223344,1,Dial(SIP/1000)
exten =
and not Exten =
--
Daniel
___
--Bandwidth and Colocation provided by
[EMAIL PROTECTED] a écrit :
Oops !
I have upgraded TRUNK again via SVN and all was seeming to be fine, no more
invalid IAX2 frames and able to place and receive calls.
I was happy..
But, few calls later (about 5 minutes) : INVAL frames again and no more
possibility to place or receive calls,
Anton Krall a écrit :
Why is iaxmodem with hylafax more stable than spandsp?
Can you run iaxmodem and hylafax together with spandsp (for running E1
r2mfc)?
You're mixing thinks: iaxmodem+hylafax is equivalent to rx_fax/tx_fax,
both are based on spandsp which is the library.
Olle E Johansson a écrit :
21 feb 2006 kl. 21.00 skrev Chris Bagnall:
What's the benefit of using stund vs nat=yes in your sip.conf
for that device? I haven't had any issues behind firewalls
when I enable that option, and no ports are needed to be opened.
For some strange reason, even
Fabrice a écrit :
Le Vendredi 3 Février 2006 13:54, Dave Cotton a écrit :
On Fri, 2006-02-03 at 09:52 +0100, Wilson Pickett wrote:
Have you seen that 3 Asterisk servers were running during this show ?
François,
I was there (had a coffee with Dave in fact) but was wondering,
Hi list,
I try to connect to a GW which have one domain eg sip.mydomain.com and
have few IPs related to this domain. I register * to this domain with
host=sip.mydomain.com and type=user. So DNS will decide on which IP of
my domain I will register (or redirection on the GW side).
If an
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