Hi,
We have two of these 4 port FXO cards.
However, we are having some problems with incoming/outgoing calls.
The latest version of Asterisk/zaptel from CVS is being used. Voicemail,
internal SIP - SIP calls between Pingtel xpressa hard phones work
terrific, echotest is fine, and so on.
The zaptel
Am 01.06.2004 um 21:11 schrieb Eric Wieling:
I suspect that the only providers that support free codecs are ones
running Asterisk. Any commercial VoIP system will only support G711,
G729 and G723.1. Your problem is very common.
That´s true ...
The idea was to fallback to G711 if G729 runs out of
Hi all,
I'm working on a setup for a small office. I'd like to use SIP/iconnecthere
most of the time, because they're cheap. But they only allow a single call.
When the single iconnecthere line is in use, I'd like to use IAX2/voicepulse
instead:
exten = _1NXXNXX,1,Dial,SIP/[EMAIL
John Todd wrote:
If the files don't seem to exist in the CVS asterisk distribution, or
the asterisk-sounds distribution (from the same CVS server) then I
suggest you record them yourself, or if you want the voice to match then
pay Allison (http://www.theivrvoice.com/) to record them for you.
hi,
what happens if you change the rxgain and txgain to something lower than
1.0 ?
best regards
Wolfgang
Am Mo, den 31.05.2004 schrieb Stefano Finetti um 17:11:
Hello all,
I've just finished to install chan_capi with 3 AVM Fritz PCI cards.
It correctly loads the 3 drivers, and * starts
Hi,
if the G.729 codec runs out of licenses does * fallback to another
codec?
TIA,
Mike
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Eric Wieling wrote:
On Mon, 2004-05-31 at 10:16, Duane wrote:
Andy Powell wrote:
Anything that's added to * that breaks how protocols work should be by default OFF not ON,
but that's just IMO...
I agree 100%, this has been very frustrating trying to work out why
Asterisk suddenly stopped
From: Wolfgang Pichler [EMAIL PROTECTED]
hi,
what happens if you change the rxgain and txgain to something lower than
1.0 ?
best regards
Wolfgang
Well, actually it just change volume on the headset...
This is not a distortion case, sound is choppy and trembling, but volume is
just
http://www.aca.gov.au/telcomm/telephone_numbering/enum_nsg2/index.htm
Gary
.
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Adam,
works now :-)
Just one further question. In my understanding Firefly's RTP Port is the
SIP listen port. So there is no chance to influence the RTP/RTCP
Portrange for the audio channel.
Please correct me if I'm wrong.
jo
Adam Hart wrote:
I just put up another version - fixed that issue and
You maybe have to create a SIP user called like it is declared in your
UAC/UAS xml file. I think it should be 'sipp' or something like that...
[EMAIL PROTECTED] a écrit : -
Pour: [EMAIL PROTECTED]
De: C. Johnson [EMAIL PROTECTED]
Envoyé par: [EMAIL PROTECTED]
Date: 31-05-2004 08:03
Firefly's RTP port option is for RTP, not SIP listen port. All RTP goes
to the one port. There's currently no option to set the SIP port (coming
shortly)
jo wrote:
Adam,
works now :-)
Just one further question. In my understanding Firefly's RTP Port is the
SIP listen port. So there is no
AFAIK.. it shows up a crazy error...
The G.729 crying for more licenses...
Isamar
On Tue, 1 Jun 2004, Mike Heininger wrote:
Hi,
if the G.729 codec runs out of licenses does * fallback to another
codec?
TIA,
Mike
___
Asterisk-Users
Add insecure=very to your FWD peer context in sip.conf.
The in-between fix really wasn't a fix. Chan_sip was modified some time
ago
(5/24 or so) to require authentication for inbound calls also. To turn this
required authentication off, you need to add insecure=very to your peer
definition.
Hello all
I've discovered that SIP channels sometimes get stuck in *.
I've read some posts from Fri 29 Aug 2003 which mentions this issue, but
there doesn't seem to be any final answers
I don't know if this is related to the 0001604 bug?
Below is a list from one of the incidents:
I know the
Has anyone had any experience with the D-Link DPH-100S SIP phones and *?
http://www.dlink.com/products/?pid=319
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
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[EMAIL PROTECTED]
The in-between fix really wasn't a fix. Chan_sip was modified some time
ago
(5/24 or so) to require authentication for inbound calls also. To turn
this
required authentication off, you need to add insecure=very to your peer
definition.
5/24 is some time ago? The age we live in!
-M
Hello Adam,
Hi Adam,
two features I would really like to have:
- the textbox from Dial a URL in the normal client (maybe optionally) so
that you could easily copy and paste numbers in
- a function that replaces +49 or wathever to 00. maybe it would be also
possible, to recognize that +49 (333)
Are they any issues still with hyperthreading processors, I've read and been
told by a few people to make sure its disabled in bios if I want to use * on
a hyperthreading machine.
Kind Regards,
Chris Bond
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[EMAIL
Hi Carlos,
Try HTB. It is better than CBQ, requires less CPU and have a better
help:
http://luxik.cdi.cz/~devik/qos/htb/
Daniel
Carlos Arnt wrote:
Hi all,
Reading
about CBQ on internet i can say "I dont understand well" ;)
So anyone
that has a good background can help me
Hi All
I found that after I played with the settings the Recall button button
started to work as flash so that is great.
Thanks for your help.
Simon
Kevin Walsh wrote:
Simon Chappell [EMAIL PROTECTED] wrote:
I have fiddled about and managed to get some of the phones to work, only
the fixed
On 28/05/2004 at 19:58 usedcanon wrote:
Hi Andy,
I am most certainly interested. If you have some example code using a DB
(MySQL maybe) that would be extremelly helpful.
BTW, I am new to fpc(Turbo pascal, Delphi and now Kylix), does it have a
linux command line IDE like the DOS version
Thanks
Hello again,
I noticed that alot of people are displaying sip:[EMAIL PROTECTED]
Can I achieve this with asterisk or do i need something else?
I have a domain and spare IP's so the dns is not a problem.
Simon
___
Asterisk-Users mailing list
[EMAIL
Now that you say that I notice that switching to the sipura I no longer
have those callwaiting beeps anymore, not sure how to get them started
though.
Simon
Boater wrote:
Are any of you guys able to use the Sipura-spa2000 with call waiting on a zap call?
-Original Message-
From: Kevin
Someone asked about it on the list last year. See
http://www.marko.net/asterisk/archives/0301/0648.html
Gary
Notice: Spelling mistakes left in for people who need to correct others to
make their life fulfilled
http://www.garypigott.net
- Original Message -
From: Jeremy Bogan [EMAIL
Thanks for the clarification. Can you please tell me how to configure those text
messages
in *?
Thanks in advance
Reto
It's the standard LibIAX2, the nice features are implemented using text
messages. I'd recommend you use the standard LibIAX2 as it's more upto
date (Something I've been
Daniel,
Do you have a working firewall ruleset for HTB, optimized for voip
?
Joachim. (Zoa)
At 10:55 1/06/2004, you wrote:
Hi Carlos,
Try HTB. It is better than CBQ, requires less CPU and have a better
help:
http://luxik.cdi.cz/~devik/qos/htb/
Daniel
Carlos Arnt wrote:
Hi all,
Reading
That prob is solved here's latest problem,
== /var/log/messages ==
Jun 1 15:44:11 RAS pppd[1019]: Plugin zaptel.so loaded.
Jun 1 15:44:11 RAS pppd[1019]: Zaptel Plugin Initialized
Jun 1 15:44:11 RAS pppd[1019]: Using zaptel device 'stdin'
Jun 1 15:44:11 RAS pppd[1019]: pppd 2.4.1b2 started by
Hi Andy,
Once again thanks. This should make things a lot
easier for me. I am greatful.
btw what is the command line to execute the freepascal
ide, also do you have any other recomendations.
Thanks
Umar.
--- Andy Powell [EMAIL PROTECTED] wrote:
On 28/05/2004 at 19:58 usedcanon wrote:
At 09:23 01/06/2004 +0200, you wrote:
we have a connection to a leagacy pbx (Siemens Hicom 150E) via PRI (E1).
Everything works really fine, but the connection breaks sometimes (there is
not really a time scheme), so that you could not dial from the hicom to * or
from * to hicom.
I see from your
On Tuesday 01 June 2004 00:56, T. Chan wrote:
Dear All,
I have used Asterisk for a few months and I have been using a January CVS
version, it has been working but has been regularly crashing. I use
Asterisk mostly as a softswitch application receiving H323 calls from
customers and send to
joachim wrote:
Daniel,
Do you have a working firewall ruleset for HTB, optimized for voip
?
No but you can build your own following htb tutorial.
Daniel
Joachim. (Zoa)
At 10:55 1/06/2004, you wrote:
Hi Carlos,
Try HTB. It is better than CBQ, requires less CPU and
The R button can do a hook flash when configured correctly.
Jason
At 20:24 30/05/2004 +0100, you wrote:
thanks for the reply, i thought it may be a stupid question but if i hit
either hook buttons i do not get any result when in a call. if i press
the hangup button it hangs up, press the pick
Hi,
I'm installing E100P for isdn pri line.
My configuration are like this.
zaptel.conf
===span=1,0,0,ccs,hdb3,crc4
loadzone = usdefaultzone=us
bchan=1-15 dchan=16 bchan=17-31
zapata.conf
Hi,
We must have been tired last night when we were trying to get this
working, the problem has now been solved.
Just in case anyone has a similar problem in future and are searching the
archives, the problem was caused by the /etc/hosts file.
*DO NOT* have the servers name listed as
On 01/06/2004 at 11:00 Umar Sear wrote:
Hi Andy,
Once again thanks. This should make things a lot
easier for me. I am greatful.
btw what is the command line to execute the freepascal
ide, also do you have any other recomendations.
Thanks
Umar.
No problem, I hope it comes in handy :D
I
I think that's related to the 2.4 kernels, as they look at the HT CPU as
2 CPU's. I'm running Asterisk on Gentoo running kernel 2.6.5 and I'm
not having any problems.
Maron
Chris Bond wrote:
Are they any issues still with hyperthreading processors, I've read and been
told by a few people to
Hello Jason,
Everything works really fine, but the connection breaks sometimes
(there is not really a time scheme), so that you could not dial from
the hicom to * or from * to hicom.
I see from your config file you are using the hicom as the
second timing source make sure the hicom is not
At 19:49 01/06/2004 +0900, you wrote:
I'm installing E100P for isdn pri line.
My configuration are like this.
zaptel.conf
===
span=1,0,0,ccs,hdb3,crc4
loadzone = us
defaultzone=us
bchan=1-15
dchan=16
bchan=17-31
zapata.conf
On Tuesday 01 June 2004 06:33, Daniel Bichara wrote:
joachim wrote:
Do you have a working firewall ruleset for HTB, optimized for voip ?
No but you can build your own following htb tutorial.
The tutorials frankly suck ass. I am no newbie to Linux or firewalling and
it's thorougly
On Tuesday 01 June 2004 07:14, Maron Kristófersson wrote:
I think that's related to the 2.4 kernels, as they look at the HT CPU as
2 CPU's. I'm running Asterisk on Gentoo running kernel 2.6.5 and I'm
not having any problems.
I'm using 2.4.25 with a Xeon 2.4 with HT turned on without issue.
#
On Tue, 2004-06-01 at 10:11, Simon Chappell wrote:
I noticed that alot of people are displaying sip:[EMAIL PROTECTED]
Can I achieve this with asterisk or do i need something else?
Sure :)
I have a domain and spare IP's so the dns is not a problem.
Just create SRV records in your DNS to
What cards you using currently I've just got one FXO card that I need to use
with it.
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: 01 June 2004 12:40 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: Hyperthreading?
Thanks for your answer!
It's a pity ... it would be great to fallback to another (free) codec.
Mike
Am 01.06.2004 um 09:43 schrieb Isamar Maia:
AFAIK.. it shows up a crazy error...
The G.729 crying for more licenses...
Isamar
On Tue, 1 Jun 2004, Mike Heininger wrote:
Hi,
if the G.729 codec runs
My tests with all shapers on adsl so far give me too much jitter to use
without jitter buffer as soon as i do an upload.
Zoa.
At 13:38 1/06/2004, you wrote:
On Tuesday 01 June 2004 06:33, Daniel Bichara wrote:
joachim wrote:
Do you have a working firewall ruleset for HTB, optimized for voip
Not using any cards at the moment here, However I will have an E100 card
installed later this week.
Chris Bond wrote:
What cards you using currently I've just got one FXO card that I need to use
with it.
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: 01 June
Ah - can any other user confirm that the FXO card works with hyperthreading
enabled?
-Original Message-
From: Maron Kristófersson [mailto:[EMAIL PROTECTED]
Sent: 01 June 2004 1:12 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Hyperthreading?
Not using any cards at the moment
Hi,
I have used the channel.c.diff patch from the email
http://lists.digium.com/pipermail/asterisk-users/2004-March/039683.html
to correct the problem.
It seems that this fix doesn´t work anymore with the current head
version.
Is this true?
TIA,
Mike
I'm using * on 3Ghz P4 with HT enabled with a TE405P card with no problem.
I'm using fedora 2 but made to change the kernel to 2.6.6.
- Original Message -
From: Chris Bond [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 01, 2004 4:50 PM
Subject: RE: [Asterisk-Users] Re:
I'm running asterisk on a 2.8Ghz w/HT and 2.4.25 kernel. I wasn't aware
that I needed to disable HT, but all seems to be running ok for now. The
2.4.x kernel seems to be completely ignorant of hyper threading, which IMO,
is quite frustrating. HTT has been around for years now, and 2.4 kernels
Well, i think i've solved the problem by myself :-)
I had to change a line in chan_capi_pvt.h:
/* was : 130 bytes Alaw = 16.25 ms audio not suitable for VoIP */
/* now : 160 bytes Alaw = 20 ms audio */
/* you can tune this to your need. higher value == more latency */
#define
I have been using Cisco ATA's for analog connections and decided to give
a Sipura SPA-2000 a try. I noticed there is a fair amount of background
white noise that is noticeable, especially after breaking the dial tone.
After pressing a '1' to break the dial tone, there is a fair amount of
noise
Am Di, 2004-06-01 um 14.42 schrieb Stefano Finetti:
Well, i think i've solved the problem by myself :-)
I had to change a line in chan_capi_pvt.h:
/* was : 130 bytes Alaw = 16.25 ms audio not suitable for VoIP */
/* now : 160 bytes Alaw = 20 ms audio */
/* you can tune this to your need.
By the way I forgot to mention that I connect to * over ISDN. I read
somewhere that ISDN v.110 connection or v.120 connections are not supported
with zapras. Is it true ? Did the above LCP problem occur due to that ?
That prob is solved here's latest problem,
== /var/log/messages ==
Jun 1
I am just about to load asterisk onto a Compaq (now HP) ML350 with 2 Xeon processors (HT enabled), 2 gig ram, 5 76Gig SCSI hard drives with hardware RAID 5. System is running Fedora Core 1 2.4.x kernel. Ill let you know how it goes.
BTW, it seems the OS thinks there are 4 processors installed.
I'm working on a setup for a small office. I'd like to use SIP/iconnecthere
most of the time, because they're cheap. But they only allow a single call.
When the single iconnecthere line is in use, I'd like to use IAX2/voicepulse
instead:
exten = _1NXXNXX,1,Dial,SIP/[EMAIL PROTECTED]
I assume thta i need to open port 5060 also?
Simon
Fran Boon wrote:
On Tue, 2004-06-01 at 10:11, Simon Chappell wrote:
I noticed that alot of people are displaying sip:[EMAIL PROTECTED]
Can I achieve this with asterisk or do i need something else?
Sure :)
I have a domain and spare
Yes, it is something to worry about, because you might run out of RTP
ports or open fd's, depending on your port range in rtp.conf.
Which cvs version are you running?
This behavior was observed by several people for a short period of time
and then seemed to have disappeared with a cvs versions
Not I.
-Original Message-
From: Kevin [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 01, 2004 7:44 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sipura-SPA2000 background noise
I have been using Cisco ATA's for analog connections and decided to give
a Sipura SPA-2000 a try. I
What cards you using currently I've just got one FXO card that I need to
use with it.
T100P and TE405P (two identical machines with different Zap hardware)
I have another system with a TDM30P (3 FXS interfaces) but it's a single-proc
P3.
Regards,
Andrew
I've tried building the 2.4.21 and the 2.4.20 kernels with the appropriate
hdlc patch and I continue to have the same results. I'm thinking this is a
problem with the routing table rather than getting hdlc compiled correctly,
but I'm pretty much at a loss at this point. I have tried the one
I'm running asterisk on a 2.8Ghz w/HT and 2.4.25 kernel. I wasn't aware
that I needed to disable HT, but all seems to be running ok for now. The
2.4.x kernel seems to be completely ignorant of hyper threading, which IMO,
is quite frustrating. HTT has been around for years now, and 2.4
Klaus-Peter Junghanns wrote:
I forgot to mention that i'm using Snom105 phones. It seems that with GS
BT101 with Ilbc firmware the value 160 works fine, but with snom it
introduces an ugly distortion and choppy audio.
This is really surprising. What codec are you using on the Snom?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Can someone please show me what the proper syntax is for Variable: in an
Originate.Basically, I'm looking to pass a string into my dialplan as
${MYSTRING} so it's available from whatever Exten = I originate to.
Thanks
- -jwb
-BEGIN PGP
My tests with all shapers on adsl so far give me too much jitter to use
without jitter buffer as soon as i do an upload.
I always use a jitter buffer with IAX2. (that's all I use, no SIP)
8 iax.conf 8
pingtime=1
lagrqtime=1
jitterbuffer=yes
;dropcount=3
On Tue, 1 Jun 2004, Terry Goodwin wrote:
BTW, it seems the OS thinks there are 4 processors installed. Even
core 2 (2.6.5 kernel) when briefly installed (because it sucks) reported
4 CPU's.
Trust me... this is what you want :)
Stefan
___
On Tuesday 01 June 2004 05:44, joachim wrote:
Do you have a working firewall ruleset for HTB, optimized for voip ?
Here, for your viewing pleasure, is my htb script. I am *positive* it can be
improved upon. I found I had to put the bulk traffic in a separate HTB
branch or otherwise it would
I see the same thing:
marconi*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Format
192.168.1.100(None) 000f9054-3a 00101/02439 UNKN
192.168.1.103(None) 000f9057-96 00101/01079 UNKN
192.168.1.101(None) 000f9048-5d 00101/00113 UNKN
Dear all -
I am looking for some information about ISDN in Venezuela. I need a small Asterisk
system with 2 ISDN channels at our offices in Venezuela (Caracas).
Can anyone advise me on the best option? I am getting mixed reports - I am told that
a BRI will cost me 1000 US dollars per month!
On Tuesday 01 June 2004 05:44, joachim wrote:
Do you have a working firewall ruleset for HTB, optimized for voip ?
The other side of my SDSL link is our provider, which I happen to help out.
They have a ptp T1 to their upstream (MCI) and I've placed the following
configuraiton in the router
I also had some dual xeon machines not able to use ht with 2.4 kernels 2.4.22
It all depends on the hardware...
Joachim (zoa)
At 15:16 1/06/2004, you wrote:
I'm running asterisk on a 2.8Ghz w/HT and 2.4.25 kernel. I wasn't aware
that I needed to disable HT, but all seems to be running ok for
All,
I went thru kernel and zaptel code and see that zaptel driver prepared
for kernels = 2.4.20 and should be able to handle cisco hdlc without
problem... but
it has been commented out from the code. What is the reason? Does anyone
have idea?
Like this:
#ifdef NEW_HDLC_INTERFACE
# uname -a
Linux roanoke-voip01 2.4.25-gentoo-r2 #6 SMP Mon May 31 07:08:41 EDT 2004
i686 Intel(R) Pentium(R) 4 CPU 2.80GHz GenuineIntel GNU/Linux
# cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model : 2
model name : Intel(R)
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
[...]
They can't? HT is detected in /proc/cpuinfo (flags) and I see two
processors with 2.4.25 SMP kernels... What exactly isn't it using?
Linux doesn't realise that scheduling a process onto one virtual CPU
reduces the performance available on the
That's the problem we had with Asterisk and HT on a 2.4 Kernel: whenever Asterisk was
staying in the RTP stream, and HT was enabled (on a Dell Dual Xeon system), we had
choppy audio. After disabling HT, everything was fine again. Nothing measurable,
indeed, but you could definitely hear it. So
What cards was it FXO - cos is it card related this HT problem?
-Original Message-
From: Manuel Wenger [mailto:[EMAIL PROTECTED]
Sent: 01 June 2004 3:18 PM
To: [EMAIL PROTECTED]
Subject: R: [Asterisk-Users] Hyperthreading?
That's the problem we had with Asterisk and HT on a 2.4 Kernel:
Hi Tim,
The ISDN isn´t in widespread use here in Venezuela, we use full or partial
E1´s, and that´s the price, $ 1100 plus a three year contract, yet some
digital PBX use ISDN channel to comunicate between them.
How many voice channels you will need?.
Take advice about the signaling, here it´s
We didn't have any card: we had choppy audio on SIP-to-SIP streams. There were no
FXO/FXS cards in the system. I don't know what problem the other poster had.
-Messaggio originale-
Da: Chris Bond [mailto:[EMAIL PROTECTED]
Inviato: martedì, 1. giugno 2004 16:22
A: [EMAIL PROTECTED]
On Tue, 2004-06-01 at 14:03, Simon Chappell wrote:
I assume thta i need to open port 5060 also?
Yes also the appropriate RTP ports (unless your Firewall/NAT is
SIP-aware can open RTP ports based on SIP messages...)
F
Fran Boon wrote:
On Tue, 2004-06-01 at 10:11, Simon Chappell wrote:
Hi!
I have the rates that I currently pay my telco, and would like to
extract my CDR's and add an additional field displaying the actual price
paid for the call. I would like to do this based on destination phone
number, and outgoing channel.
Please look at this page for CDRtool
Pro's advise me on Installation and Configuration of Asterisk to support
Megaco (RFC 3015)
Clients.
Java
_
Is your PC infected? Get a FREE online computer virus scan from McAfee®
Security.
Hello everybody!
After checking the complete wiki and the mailinglist archives I still
haven't really found out why the following constellation does not work.
We have an asterisk-System with some SIP-Phones and an old ISA
Fritz-ISDN-Card used with i4l. The whole system is integrated in out
Thanks Frank,
last question ;-)
would the number be the extension? or the number i have at fwd..
ie sip:[EMAIL PROTECTED] or sip:[EMAIL PROTECTED]
Thanks
Simon
Fran Boon wrote:
On Tue, 2004-06-01 at 14:03, Simon Chappell wrote:
I assume thta i need to open port 5060 also?
Title: Unsupported Media error from iConnectHere
I can't talk through iConnectHere. The connection gets made but as soon as any sound is transmitted the call ends and the Asterisk console shows an Unsupported Media error as follow:
Got SIP response 415 Unsupported Media back from
Since I only seem to get questions, and no feedback, from the Wiki page,
I'll ask here. There seems to be no lack of opinions here...
I have a working wakeup call system on my home * system. The architecture
is something I'm not perfectly happy with, though. There are two AGI
scripts, written
Hello,
I have installed Asterisk and if I start the administration console with the
command line asterisk -c,and after 10 sec, my system is completely block and
I have to reboot each time.
warnings appear like:
- chan_iax2.c: 6835 load_module: unable to open IAX timing interface
- chan_skinny.c:
On Mon, May 31, 2004 at 10:53:43PM -0700, John Todd wrote:
At 1:18 AM +0200 on 6/1/04, Julian Pawlowski wrote:
Hi there!
I just try to play with die wake-up function described in
http://www.voip-info.org/wiki-Asterisk+tips+wake-up
Everything looks fine but there seem to be missing some
Kevin [EMAIL PROTECTED] wrote:
I have been using Cisco ATA's for analog connections and decided to give
a Sipura SPA-2000 a try. I noticed there is a fair amount of background
white noise that is noticeable, especially after breaking the dial tone.
After pressing a '1' to break the dial tone,
At 11:27 AM 6/1/2004, you wrote:
Rob
I would be very interested
Since I only seem to get questions, and no feedback, from the Wiki page,
I'll ask here. There seems to be no lack of opinions here...
I have a working wakeup call system on my home * system. The architecture
is something I'm not
To all that are interested Im hoping to have a beta (or alpha)
available for download today. I will email the list as to where it can
be downloaded.
Kyle
Brian D'Arcy wrote:
Kyle,
I also would be very interested. It may negate the purchase of a much
more expensive phone in the future. =)
looking to move an asterisk pbx server to a different vlan and as such
looking to check the impact of this change on the asterisk application
obviously we have the linux interface reconfiguration to complete
are there any application level settings that need to be changed to reflect
the changed
On Tue, 2004-06-01 at 16:00, Simon Chappell wrote:
would the number be the extension? or the number i have at fwd..
ie sip:[EMAIL PROTECTED] or sip:[EMAIL PROTECTED]
Your extension ;)
or nicer if you can set up an alias.
e.g.
exten = schappell,1,Goto(lan,2000)
F
Mike Heininger [EMAIL PROTECTED] wrote:
It's a pity ... it would be great to fallback to another (free) codec.
Just use a relatively-free codec (iLBC or GSM etc.) in the first place,
and avoid G.729. That strategy works for me. :-)
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/
Why all the time the firefly show me the message: Sip registration failed
for the network Home (407).
The server, username and password are correct. I'm using the default RTP
port 5000 in the SIP tab.
Using the SJPhone I can register; using the firefly I can call any
registered number, but I
Rob Fugina wrote:
It has occurred to me that the two AGI scripts could be rewritten as real
compiled asterisk applications, but then it always hits me that without
some kind of cron-line built-in scheduler, or changes to the outgoing
call queueing that would allow a call to be scheduled for the
On Tue, 2004-06-01 at 10:27, Rob Fugina wrote:
It has occurred to me that the two AGI scripts could be rewritten as real
compiled asterisk applications, but then it always hits me that without
some kind of cron-line built-in scheduler, or changes to the outgoing
call queueing that would allow
Hi,
I'm trying to compile h323 channel driver on cvs Asterisk 1.0 but no success
(I get a lot of errors - related to pwlib library).
I read in docs that there is also 3rd party h323 channel driver (somehow
both even share protion of code?).
I wonder what are pros and cons of both drivers ?
Rob Fugina [EMAIL PROTECTED] wrote:
So the Perl code creates call files in
a wakeup queue directory, and a cron job (a shell script) runs every
minute looking for wakeup calls in the queue that need to be handled,
and moves them to the outgoing call queue.
You may want to consider using at
Hello everybody,
we've implemented a new Channel Driver for *. It uses the new mISDN
isdn4linux architecture and supports bri te and nt mode for now.
I assume, there are lots of bugs we didn't found yet, and even mISDN is
rarely stable. So we search brave volunteers to test the driver.
Get
Not really a comfort noise. I say anything and it doesent go away. It
sounds like a shielding issue. I have tried to relocate the unit but it
doesn't seem to help.
-Original Message-
From: Kevin Walsh [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 01, 2004 11:46 AM
To: [EMAIL
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