Hi there
I have a Dialogic VFX/40 ESC plus installed
on Redhat Linux 8.0
and looking for Channel drivers for this
Card.
where cann i found channel drivers for VFX/40
ESC ?
Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
h
I reloaded my asterisk and found some red lines flushing by. When I
stopped it I see:
WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring signalling
WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring echocancelwhenbridge
WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring echotraining
Reconfigur
Karl Brose wrote:
There is no such thing as "subscribecontext" parameter in SIP.
I have updated the wiki with the correct current information to make
this work.
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+snom
The subscribecontext is part of chan_sip2, not in standard Asterisk
ch
> Unfortunately it seems that when asterisk tries to ring the other
> number on Zap/2 it thinks the call has been answered and can therefore
> stop ringing IAX2
I wonder if you could fool it by
ringing IAX2/100
then after 15, ring ZAP/2/1234567&IAX2/111
and make 111 ring the same IAX2 device.
_
On 22-Dec-2004, Daryll Strauss wrote:
> What I'd like to do is write a dialplan that recognizes:
> @sip.earthlink.net
> and does a dial like the one above.
This is possible, but it's a bit convoluted what needs to be done.
Fundamental to asterisk design seems to be the notion that ${SIPDOMAIN}
i
Hi guys,
I got 2 licenses of g.729 and while running the asterisk with Monitor
(for recording a channel) and using one channel for the call... I
receive this error:
WARNING[23826]: codec_g729.c:180 g729tolin_framein: Out of G.729
Decoder Licenses!
many times
it starts only when the call throug
It seems you have setup an asterisk sys would like to help someone new to
asterisk get started. If so contact me at [EMAIL PROTECTED] My name
is Shelton Wardsworth.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI)
Sent: Wednesday, December 22,
"What OS Release of Phone System were you using to do the setup?
G3si R6?
If I remember off top of my head. I'm out of the
office until after the new year.
I'm also not using a CSU/DSU, not needed, but shouldn't
matter anyway.
Which ISDN options are enabled under customer options?
Do you have or h
Hi !
I am facing a serious discennection problem with asterisk queues. Now I am
basically running a call center where users land from allover the place
through internet. I am using asterisk queues to queue them and then they
are forwarded to agents. Now what happens is that after some
We changed from Fedora to Debian and saw the following increase in
performance:
Before:
G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC
G723 - - - - - - - - - - -
GSM - - 6 615 6 522 -
I had tried g3 first...the individual address was at guess that g3 didn't
work.
Thanks.
- Original Message -
From: "Jason Kawakami" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, December 22, 2004 9:43 PM
Subject: [Asterisk-Users] RE: Zaptel/Zapata config from T410p to
BrooktroutT1
-O
> Richard wrote:
> > Hi,
> >
> > Has anyone tried to use cdp to push the voice vlan tag to polycom
phones?
> > The document says that it is supported, but I can't make it work.
> >
> > Thanks,
> > Richard
>
> Richard,
>
> I can't either. I've tried using HP Procurve switches and even my
>
Rodolfo Grave wrote:
Can you please try to set up X-Pro or a Budgetone with g729 (In case you
have one of those at hand, of course)? If you're succesful I'll know
it's Asterisk having problems. I dont have much resources to test (just
the Budgetone and X-Pro).
Thanks again,
RODOLFO
Rodolfo,
Can you please try to set up X-Pro or a Budgetone with g729 (In case you
have one of those at hand, of course)? If you're succesful I'll know
it's Asterisk having problems. I dont have much resources to test (just
the Budgetone and X-Pro).
Thanks again,
RODOLFO
Kristian Kielhofner wrote:
Rodolf
Hello Rich,
First of all, thank you very much for your help and patience.
I've just arrived home from work (yes, I'm one of the midnight oil burners :-))
and implemented and tested your suggestions; unfortunatelly it didn't work, the
same behaviour remains.
More details follow below, in-line:
O
I am running the Stable 1.01 version of Asterisk on a Dell SC420, RedHat
9. I have a PRI to the Telco and the phones are SIP 7960's. Both the
phones and the asterisk box are on a private subnet behind the firewall.
No NAT involved here.
Intermittently, the users complain of the conversation 'fad
There is no such thing as "subscribecontext" parameter in SIP.
I have updated the wiki with the correct current information to make
this work.
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+snom
--khb
Peer Oliver Schmidt wrote:
Hi,
does the hint extension work together with the Snom
Rodolfo Grave wrote:
I've seen other threads about the topic here (just that with a
Sipura)... this is the thread subject:
[Asterisk-Users] G729 and Sipura.
Digium's answer to this person was to blame the device, so I haven't
even try to contact Digium support. I'll do it now, and I'll let you
I've seen other threads about the topic here (just that with a
Sipura)... this is the thread subject:
[Asterisk-Users] G729 and Sipura.
Digium's answer to this person was to blame the device, so I haven't
even try to contact Digium support. I'll do it now, and I'll let you
know. However, I did
Rodolfo Grave wrote:
Yeap... :(
Has (or are they willing) Digium support looked at it? If they haven't,
I would be willing to SSH into the machine if it is available.
Let me know, either here or directly to me.
--
Kristian Kielhofner
___
Asterisk-Users
-Original Message-
Message: 8
zaptel.conf
span=2,0,0,esf,b8zs
e&m=25-48
zapata.conf
signaling = em_w
context = faxserver
group = 3
channel = 25-28
exten.conf
exten => 1231231234,1,Dial(Zap/2-2/${exten})
I think if you change the last line to :
exten => 1231231234,1,Dial(Zap/G3/${E
With this trimmed down versions of sip.conf and extensions.conf. I can
now receive calls from outside. But audio will not traverse out to the
internet. I can hear the caller no problem. Also I still cannot
dial-out. Any ideas?
Regards,
Norman Zhang
I removed all the PSTN stuffs. As I'm only try
Yeap... :(
Kristian Kielhofner wrote:
Rodolfo Grave wrote:
Hi.
Has anyone accomplished to use the g729 codec? I have the license
installed, and I have tried with X-Pro and a Grandstream Budgetone
configured to use g729 only. This is what I get from Asterisk:
Dec 23 02:38:07 WARNING[21176]: chan_
Sorry, Replied to wrong message:)
--
Christopher Dobbs
Christopher Dobbs wrote:
Here is a list of the libs I am using
ld-2.2.5.so*
ld-linux.so.2@
libc-2.2.5.so*
libc.so.6@
libcom_err.so.2@
libcom_err.so.2.0*
libcrypt-2.2.5.so*
libcrypt.so.1@
libdl-2.2.5.so*
libdl.so.2@
libe2p.so.2@
libe2p.so.2.3*
l
Norman Zhang wrote:
I can't receive/send calls with Asterisk. Could someone please give me a
few pointers on my configuration?
I removed all the PSTN stuffs. As I'm only trying to make SIP work.
Would someone kindly give me a few pointers?
Regards,
Norman Zhang
[general]
disallow=all
allow=ulaw
Here is a list of the libs I am using
ld-2.2.5.so*
ld-linux.so.2@
libc-2.2.5.so*
libc.so.6@
libcom_err.so.2@
libcom_err.so.2.0*
libcrypt-2.2.5.so*
libcrypt.so.1@
libdl-2.2.5.so*
libdl.so.2@
libe2p.so.2@
libe2p.so.2.3*
libext2fs.so.2@
libext2fs.so.2.4*
libm-2.2.5.so*
libm.so.6@
libncurses.so.5@
libn
Richard wrote:
Hi,
Has anyone tried to use cdp to push the voice vlan tag to polycom phones?
The document says that it is supported, but I can't make it work.
Thanks,
Richard
Richard,
I can't either. I've tried using HP Procurve switches and even my
Catalyst 2950T-24. Neither work. I have been
Hi,
Has anyone tried to use cdp to push the voice vlan tag to polycom phones?
The document says that it is supported, but I can't make it work.
Thanks,
Richard
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mai
Arthur B Olsen wrote:
But whats the future of iaxy. Are these problems being fixed. Or is the whole
project dropped?
I don't know. I see ocassional updates to the iAXY firmware in CVS, but
it's on binary so there is no real way to know. I really wish Digium
would respond to some of this stuff.
But whats the future of iaxy. Are these problems being fixed. Or is the whole
project dropped?
On Thursday 23 December 2004 01:38, Kristian Kielhofner wrote:
> Arthur B Olsen wrote:
> > So i guess were screwed. These unusable thingies are quite expensive.
> > Good thing i only bought two.
> > Now
Russell Horn wrote:
Alexander,
I'm afraid it's POTS (X101P) and from what I have seen since I posted
this is my problem.
I wouldn't mind it hanging up the IAX2 channel and then calling it
again, but it seems that everytime the new call via Zap/2 means no
other calls are possible.
There is ISDN in t
FXS ports are always answered immediately.
Do you have to dial out over PTSN?
If so you are going to have an interesting time breaking this problem.
--
Christopher Dobbs
Russell Horn wrote:
Alexander,
I'm afraid it's POTS (X101P) and from what I have seen since I posted
this is my problem.
I wouldn
Rodolfo Grave wrote:
Hi.
Has anyone accomplished to use the g729 codec? I have the license
installed, and I have tried with X-Pro and a Grandstream Budgetone
configured to use g729 only. This is what I get from Asterisk:
Dec 23 02:38:07 WARNING[21176]: chan_sip.c:2764 process_sdp: No
compatible
I have an embedded Linux Distro that is specifically designed to fit on
as small as a 32MB CF card.
Includes:
HTTP Server
SSH Client and server
DHCPCD
DHCPD
bash
and more
Contact me off list if you are interested.
--
Christopher Dobbs
Michael G
Hi.
Has anyone accomplished to use the g729 codec? I have the license
installed, and I have tried with X-Pro and a Grandstream Budgetone
configured to use g729 only. This is what I get from Asterisk:
Dec 23 02:38:07 WARNING[21176]: chan_sip.c:2764 process_sdp: No
compatible codecs!
Dec 23 02:38
Arthur B Olsen wrote:
So i guess were screwed. These unusable thingies are quite expensive.
Good thing i only bought two.
Now im really nervous about the isdn pri card i bought. Gonna try it out
tonight. Hope its different.
Is the software for iaxy open source. Then maby it can be fixed.
I hate t
So i guess were screwed. These unusable thingies are quite expensive.
Good thing i only bought two.
Now im really nervous about the isdn pri card i bought. Gonna try it out
tonight. Hope its different.
Is the software for iaxy open source. Then maby it can be fixed.
On Wednesday 22 December 2004
Hi all,
A new version of the Asterisk Management Portal is available for
download. AMP now supports SIP and IAX2 trunks.
Please visit the AMP homepage at http://amp.coalescentsystems.ca
Join the AMP mailing list for discussions about AMP.
Changes:
- Added Call Group CID Name prefixing
- Renam
Steve Underwood a écrit :
Hi Jean Denis,
I think you are using spandsp-0.0.1 It has this problem with some
Philips, Canon, and a few other machines. Try spandsp-0.0.2pre6. That
fixes the problem, and has various other improvements.
Sorry for forgetting that important info in my first message, bu
Not such a stupid question :)
The paper instructions included with the card failed
to mention that needed to be done manually :)
Up and running :)
-Dorn
On Wed, Dec 22, 2004 at 10:34:17PM -0800, Shahed wrote:
>
> >I can see contents in /dev/zap but any attemp to
> >touch for example /dev/zap
No the card does not have to ring 20 phones at the same time. At least in
North America, the ring cycle is 6 seconds, 2 seconds on, 4 seconds off.
Stagger the ON between the ports, you can divide the 4 ports by three and
you really only have at the most two ports ringing at once or 10 phones.
I t
Henry,
I noticed on a post on the asterisk user's forum, you are versed with
both Asterisk and the Toshiba CTX product. I have a customer that would
like to integrate an existing location using a CTX to a new location
with asterisk. Are you aware of a way to link the two systems via the
intern
No. It means it can ring a set of phones that add up to 5 REN. This number
does not equate to watts.
The unit has to power the electronics, send out talk battery, add
superimposed DC on the ringing voltage. Generate the ringing circuit, etc.
>From + and - 12 volts, we get -48v talk battery and
HI, I'm trying to config a span from port 2 of a
DIgium T410p to a Brooktrout TR114-P8V-T1 card. I have a T1-PRI from the
TELCO in port 1 (thru first port) working just fine with Asterisk 1.0.3 - been
working fine for some time now. No problem with dialplan PSTN calls.
Now I'd like to "ro
On Thu, 2004-12-23 at 11:08, Andrei (MPI) wrote:
> Hi there,
>
> We are using 10+ Polycom SP IP 600 phones with Asterisk and TMD400P with
> 4 FXO lines.
>
> So far we have 3 phones with following problem: more or less frequently,
> for every call or ever other call, user of the phone would hear
Hi Jean Denis,
I think you are using spandsp-0.0.1 It has this problem with some
Philips, Canon, and a few other machines. Try spandsp-0.0.2pre6. That
fixes the problem, and has various other improvements.
Regards,
Steve
Jean-Denis Girard wrote:
Hi list,
Just installed spandsp. In my limiting te
Hi there,
We are using 10+ Polycom SP IP 600 phones with Asterisk and TMD400P with
4 FXO lines.
So far we have 3 phones with following problem: more or less frequently,
for every call or ever other call, user of the phone would hear brief
interruptions on the line when dialing out via PSTN, lik
Well give oej and steve some time here ... the project sure couldn't hurt
from more enterprise funding... lets just hope some of that makes it way
back to the root of the project. Also I was quick to judge their intentions
and I shouldn't have been... so guys lets give them some support and see
wh
What registration failure is that?
The only way to tell is a complete SIP trace of what's going on.
The registration timeout on the phone and in Asterisk should be the same,
unless the server goes down and reboots. The server usually has no way
to tell a phone to
re-register (no real need to do so
All,
I am from Brazil, and I am no finding the Digium boards to sell here.
I am currently using FreeBSD 5.3 to run asterisk. Digium cards could be
used with this OS ?
How is possible to use a generic modem (like INTEL MD3200) with FreeBSD
5.3? It has the total supported drivers for it ? Does som
Alexander,
I'm afraid it's POTS (X101P) and from what I have seen since I posted
this is my problem.
I wouldn't mind it hanging up the IAX2 channel and then calling it
again, but it seems that everytime the new call via Zap/2 means no
other calls are possible.
There is ISDN in the office, but I
Russell,
What kind of zap cards do you have??
If T1, is it PRI or RBS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russell
Horn
Sent: Wednesday, December 22, 2004 6:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-U
Ken Godee wrote:
I'm currently playing with a Digium T100P card and 2 Grandstream
phones, things are working well. I wanted to move on to linking our
Definity G3R Rev 8.2 to the T100P. Everything that I've read so far
shows that you need a TN464 to accomplish this. We have a TN767E
availabl
Hi,
Anyone on this list is in Moscow and interested, e-mail me off list.
Regards,
Alex.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
h
I have a problem with ringing simultaneous channels where one is IAX
and one is Zap
I have two Zap channels and a single extensions on IAX2
I'm trying to take incoming calls on Zap/1 and if not answered in 15
seconds by IAX2/100 to keep ringing IAX2 and also try another number
on Zap/2
Unfortuna
I've got soft phone that allows me to dial SIP URI's. I'd like to
route these calls through a provider to be completed, because I'm
beind a NAT box and doing it directly doesn't work.
Right now I've got an extension defined like this:
Dial(IAX2/${FWDUSERID}:[EMAIL PROTECTED]/**356)
This will conne
Hi
I have something like this but it's in french and it uses teh res_config
Best regards
Thierry wehr
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Nabeel
Jafferali
Envoyé : mercredi 22 décembre 2004 22:57
À : asterisk-users@lists.digium.com
Objet :
Hmm..perhaps because SIP and IAX use different registration
methods?
-Matthew
- Original Message -
From: "Greg - Cirelle Enterprises" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, December 22, 2004 4:34 PM
Subject: [Asterisk-Users] register_verify defined in 2 files?
> I know I'm getting ti
which docs are you talking about?
At 06:15 PM 12/22/04, you wrote:
Yeah, I d like to get those docs too.
--
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan
Sent: Wednesday, December 22, 2004 4:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Su
The criteria are published in RFC 3435, range is from 1 to 999,999,999.
there is no requirement
of starting from 1. Call agents may allocate certain ranges for certain
groups of gateways.
Asterisk (the call agent) simply increments the id numbers monotonically
for each new request.
Most likely
I had this problem when I used the TDM400P on an FC3 system. The Wiki
helped:
http://www.voip-info.org/wiki-Asterisk+Linux+Fedora
[snippet]
Additional Notes for Zaptel if you get error message
Notice: Configuration file is /etc/zaptel.conf
line 143: Unable to open master device '/dev/zap/ctl'
I know I'm getting tired of looking at code, but
why is the function register_verify defined in 2 different
files?
chan_iax2.c
line 3860
static int register_verify(int callno, struct sockaddr_in *sin, struct
iax_ies *ies)
chan_sip.c
line 4869
/*--- register_verify: Verify registration of user */
I can see contents in /dev/zap but any attemp to
touch for example /dev/zap/ctl gets a no such
device or address ...
May be a stupid suggestion, but have you loaded the
wctdm driver by doing a "modprobe wctdm" ??
Regards
Shahed
___
Asterisk-Users mailin
David Ishmael wrote:
Yeah, I’d like to get those docs too.
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Paul Rodan
*Sent:* Wednesday, December 22, 2004 4:03 PM
*To:* 'Asterisk Users Mailing List - Non
I'm curious as to the minimum to have a perfectly normal asterisk running,
with SIP and IAX support, no zaptel or anything out of the ordinary.
VoiceMail and such must still be there. I don't know which modules I can
unload safely and not compromise the performance/ability of the server. I
only use
I just installed a new TDM400P with one FXO interface
in slot 4 (how it came from Digium). This box is
running Debian with a 2.6.10-rc2-mm3 kernel. After
the make linux26 and make install in /usr/local/src/zaptel,
I can see contents in /dev/zap but any attemp to
touch for example /dev/zap/ctl g
Hi, I'm setting up * to extend Norstar locals to our off premises support
personnel. It works quite well as a simple extension extender.
When the Norstar connects to the extension that is tied to the zap fxs port,
I automatically dial the voip phone number and the call is connected.
I would like
Yeah, I’d like to get those docs too.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan
Sent: Wednesday, December 22, 2004
4:03 PM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Can
somebody email me the Sipura SP
Osvaldo Mundim wrote:
Hi,
We currently have an Asterisk box (P4 2.4 Ghz, 512Mb RAM, 1 T100P,
Zhone Channel Bank and 1 E100P) connected to an ISDN running without
any problems. That machine is working for about 1 year.
Two days ago, we decided to switch that machine for two PowerEdge
600SC (HA)
I heard Sipura had really awesome documentation on the
SPA-2000 and SPA-3000, but you have to email them for it. When I did, they said
I had to get it from a reseller. It’s been a while since I bought my
units, I don’t even remember where or who they were bought from. Can
somebody email me
Bruno Hertz wrote:
Did anybody already attempt to strip down an asterisk config
to an absolute minimum for a specific use?
Let's say I have a home installation and want to use capi and
iax exclusively, and load only the channels, apps, codecs,
file formats I really need.
I normally just rm the u
That answered my question. My question should have been "does IAX use codecs
like SIP does?" and since you send gsm over IAX then the answer is yes.
Thanks,
Matthew
- Original Message -
From: "Rich Adamson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
I've been working with Grandstream support on the caller ID thing and after
looking at a packet capture using Ethereal, it seems that * is sending the
characters to the phone rather than the number. I've got the following
configured in the extensions.conf file:
exten => s,1,SetCallerID(${CALLERID
If anyone is experiencing this type of registration error:
Registration from '' failed for '192.168.70.25'
try adding the following line to your modules.conf file
noload => app_adsiprog.so
This error is clearly asterisk trying to register with the phone
and not the other way around.
the app_adsipro
Hello.
I am looking for a simple Asterisk billing solution. I expect about
50-100 users (a mix of IAX and SIP) through 3-5 outgoing providers (all
IAX).
I need something that can handle monthly fees and per call charges
(depending on destination, obviously), and should provide a web
interface for
My * is sitting behind my firewall and NAT'd. From the log I see
incoming calls are answered
AstInt->69.90.155.70->sip->accept
but then
69.90.155.70->AstExt->Sip->reject
Reason: sip reason: Illegal redirection 69.90.155.70->65.39.205.114
May I ask how can I solve this?
May I ask is Asterisk or FW
> If I send a call to a provider over SIP, I have to designate which codec to
> use (711 or 729).
> Having never used IAX I have to ask: If I send a call via IAX does IAX use
> 711/729 as well?
> If a SIP 729 call uses 20Kbps, what does an IAX/729 call use?
Depends on how you have asterisk setup.
Are there any good
SIP softphones with support for SUBSCRIBE/NOTIFY, like the SNOM phones have? I
tried one from Estara, but it costs $100 and it crashed when I tried to run it
(maybe I could have make it run, but any software that crashes after running
install/config wizard is not something
If I send a call to a provider over SIP, I have to designate which codec to
use (711 or 729).
Having never used IAX I have to ask: If I send a call via IAX does IAX use
711/729 as well?
If a SIP 729 call uses 20Kbps, what does an IAX/729 call use?
Thanks,
Matthew
_
I know this is not the most appropriated list to
this, but I will try:
Does anyone know what is the criteria to the
generation of the transaction identifiers in MGCP? I mean, are they generated by
a randomic method?
I'm using Asterisk and MGCP eyeP Phone and observed
that the RSIP and NTFY
On Wed, 22 Dec 2004 12:39:46 -0600, Kristian Kielhofner wrote:
>Michael Graves wrote:
>
>> They look cute, but not enough RAM for *. Someone already has * ported
>> to the Soekris 4801. Have a look at www.soekris.com.
>>
>> Michael
>
>I didn't have to "port" Asterisk, the Soekris boards have 586'
Eric Wieling aka ManxPower wrote:
For one thing there is no mask= setting. If you look in sip.conf.sample
you'll see the line below as an example of the correct use of localnet.
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local
networks
Thanks. I made the changes. Do I need to
Hello,
My company TeleNova is offering a new service - NovaPhone IP - directly to
Brazilian companies already using Asterisk.
The rates are very attractive: to give you an idea it is possible to terminate
calls to any fixed phone in Brazil for R$ 0,20 (charges included).
We are licensed, genera
Norman Zhang wrote:
My * is sitting behind my firewall and NAT'd. From the log I see
incoming calls are answered
AstInt->69.90.155.70->sip->accept
but then
69.90.155.70->AstExt->Sip->reject
Reason: sip reason: Illegal redirection 69.90.155.70->65.39.205.114
May I ask how can I solve this?
May I a
On Wed, 22 Dec 2004, Rich Adamson wrote:
> That's an answer to the wrong question. See example below.
> Rephrased: Why do folks think they have to use Answer in the sequence
> when Playback (etc) is _not_ used?
And even if you do play back some audio you may not want to answer anyway.
In most m
As much as I appreciate the work done by Digium on Asterisk, it
appears as though the IAXy is not ready for prime time.
1) IAXy has no security of any kind, anyone with iaxyprov can
reprovision your device without so much as a password!!!
2) The IAXy doesn't work with regular dhcp, it uses bootp (
Norman Zhang wrote:
Hi,
I can't receive/send calls with Asterisk. Could someone please give me a
few pointers on my configuration?
Regards,
Norman Zhang
; sip.conf
[general]
disallow=all
allow=ulaw
port=5060
bindaddr=0.0.0.0
externip=x.x.x.x
localnet=192.168.22.0
mask=255.255.255.0
For one thing
Shahed wrote:
Hi All,
I am wondering if I will be breaking the GPL,
if I write for example, a channel driver or
make some modifications to the astrisk source code,
to interface at RUN TIME, through sockets, with
a proprietary system.
First, a warning - I am NOT a lawyer. I don't play one on TV, ei
Rich Adamson wrote:
That's an answer to the wrong question. See example below.
Rephrased: Why do folks think they have to use Answer in the sequence
when Playback (etc) is _not_ used?
[voiptalk.org]
;forwards any calls starting with an "8" thru voiptalk.org
exten => _8.,1,Answer
exten => _8.,3,Se
Here's a thought.
If you could get the gumstix to have 4 FXS ports and one Ethernet
interface, then you can sell an Asterisk voicemail/auto-attendant
solution.
You can already buy these with some proprietary hardware and software.
This can be integrated with legacy PBXs in a minute.
If Digium c
Hi,
I can't receive/send calls with Asterisk. Could someone please give me a
few pointers on my configuration?
Regards,
Norman Zhang
; sip.conf
[general]
disallow=all
allow=ulaw
port=5060
bindaddr=0.0.0.0
externip=x.x.x.x
localnet=192.168.22.0
mask=255.255.255.0
context=inbound-sip
maxexpirey=180
Hope this is the right maillinglist.
I would like to know how i can secure the iaxy. Or is the the sad truth that
anyone with an iaxyprov program can change any box not behind a firewall?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Try sending 5350 config and oh323.conf, versions, etc...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Goran Dj.
Sent: Wednesday, December 22, 2004 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk-
That's an answer to the wrong question. See example below.
Rephrased: Why do folks think they have to use Answer in the sequence
when Playback (etc) is _not_ used?
> There's a wiki tip that suggests you always put answer and even wait
> before playback, cause asterisk can
You are not quite correct.
sip-friends is the STABLE 1.0 method of storing sip.conf info on
peers/users.
sip_buddies is the CVS (RealTime) method of storing sip.conf info on
peers/users.
-Matthew
- Original Message -
From: "Giovanni Powell" <[EMAIL PROTECTED]>
To: "Asterisk Users Maili
*** SIP Channel fixed in CVS stable
---
During a few days there's been a buggy SIP channel in CVS STABLE, but
not in the 1.0.3 release tarballs on the FTP server and mirrors. We have
now removed the patch that was integrated by mistake so CVS should be
ok again.
My configuration is:
[ISDNPRI] -- [CISCO AccessServer AS5350] [ASTERISK] --
[CISCO ip phone 12SP+/Skinny]
When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN
everything working ok (RTP is ok).
But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone
IP phone party can
Guys I think I DONT GET THE POINT ,, or all of us didnt get the point.
there is too much diference between
Certified training and learning course of asterisk.
and
Course to be certified in asterisk.
yes I also know order of factors do not afect the result :)
to the guys of metrotel,
There's a wiki tip that suggests you always put answer and even wait
before playback, cause asterisk can pickup a milisec after you've
finished dialing, unlike legacy PBXs that always ring at least once.
Take a look at
http://voip-info.org/wiki-Asterisk+tips+answer-before-playback
> -Original
There was talk on the list... some time ago... an iax firmware has yet
to be released.
On Wed, 22 Dec 2004 13:35:23 -0500, Dorn Hetzel
<[EMAIL PROTECTED]> wrote:
>
> I can't get the link to work. Does this mean that there is
> some IP phone available which if loaded with the right
> firmware ca
1 - 100 of 216 matches
Mail list logo
