Chris Bagnall wrote:
Greetings list,
Wondering if anyone's come across this before.
I've configured a couple of our servers with a privatedundi context to
allow calls to still flow between extensions even if they're registered to
different servers . The DUNDi lookups seem to work fine,
Forrest Beck wrote:
I setup sip realtime. Is it possible to use a type of friend? User
and Peer seem to work fine.
have you tried? If so, what went wrong? (*hint* ;-) )
--
Met vriendelijke groeten,
Remco Post
SARA - Reken- en Netwerkdiensten http://www.sara.nl
High
On Sun, Apr 29, 2007 at 09:06:53PM +0200, Clayton Milos wrote:
Since a PRI is a physical connection as well as a logical one, if you can
get the server to shut down when it has a problem you could put a 4-pole
relay to change the PRI over to the other box.
The ISDN Guard is an excellant
On Sun, Apr 29, 2007 at 09:06:53PM +0200, Clayton Milos wrote:
Since a PRI is a physical connection as well as a logical one, if you can
get the server to shut down when it has a problem you could put a 4-pole
relay to change the PRI over to the other box.
The ISDN Guard is an excellant
Steve Edwards wrote:
Should I be concerned that cpu1 is servicing only 700,000 interrupts
from my te410p while cpu3 is servicing almost 90,000,000?
I thought this is what irqbalance was for...
Steve,
It was my experience that irqbalance used smp affinity to bind the
interrupts from each
Voxy - the only way to integrate VoiceXML applications in Asterisk.
Configure your dial plan with the URL of your VoiceXML application and it's
done. Is something the free and open source Voxy what you are looking for?
http://sourceforge.net/projects/voxy
On 5/3/07, wendell hamilton [EMAIL
Hello list,
I have always though codec translation table is dircetly connected to system
speed, utill i came across this:
in my lab, i have 2 boxes,
First box is an Intel Celeron 1.7 GHZ with 256M RAM:
show translation
Translation times between formats (in milliseconds) for one second
Gordon Henderson wrote:
(aren't you guys getting rid of ISDN anyway? :-)
H... Some people would like to think so, but it's going to be here
for a long time yet! BT have/are dumping the consumer versions of
ISDN2 - home highway which went a while back, but business highway
is going soon
Per Jessen wrote:
Yeah, that's cheap - I've just ordered two M175s at USD40/each.
Just in case anyone's interested - I got the M175s this morning and they
work just fine with the Sipura/Linksys SPA-921/-941s.
/Per Jessen, Zürich
--
ENIDAN Technologies GmbH - managed email security.
Am Freitag, den 04.05.2007, 00:48 -0400 schrieb Doug Crompton:
Can anyone suggest a source for a free robot dialer or examples? I need to
do some non-commercial auto dialing using Asterisk. Simple phone numbers
in a file, line by line format.
I found one called AstAutoDiaker but I was not
On Fri, 4 May 2007, Per Jessen wrote:
Gordon Henderson wrote:
(aren't you guys getting rid of ISDN anyway? :-)
H... Some people would like to think so, but it's going to be here
for a long time yet! BT have/are dumping the consumer versions of
ISDN2 - home highway which went a while
Hi,
a block of my extensions.conf no longer works after upgrading from
1.2.17 to 1.4.4. I have:
[macro-dialout]
exten = s,1,Gosub(s-${ARG1},1)
exten = s,n,Congestion
;; default
exten = _s-!,1,Gosub(s-NET,1)
When calling that macro whith no argument
On Thu, 3 May 2007, Kyle Gordon wrote:
Hi all,
With the gamut of FXO cards out there, I'm looking for a recommendation for
home use. I have a nicely working Asterisk 1.4 system that just requires an
FXO card to connect my NTL PSTN to it. My previous X101P clone seems to have
kicked the bucket.
Steve Edwards [EMAIL PROTECTED] writes:
I see the following on one of my new servers:
-ts10::sedwards:~$ cat /proc/interrupts
CPU0 CPU1 CPU2 CPU3
0:29790452988620 87780075 87779501IO-APIC-edge timer
[...]
225:4611916 681023
Do any one knows the formula to calculate memory and cpu usuage for channel
on g729 codec,to know the hardware required for 100 concurrent call.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
On 5/3/07, Ken Leland III [EMAIL PROTECTED] wrote:
When dtmfmode is set to inband for SIP, and i originate a call from sip
out to the PSTN, I can hear the DTMF digit twice in the audio stream.
Once very briefly and once for normal duration.
Our Theory: While Asterisk is parsing the DTMF, for a
On Thu, May 03, 2007 at 08:17:25PM +0100, Kyle Gordon wrote:
Hi all,
With the gamut of FXO cards out there, I'm looking for a recommendation for
home use. I have a nicely working Asterisk 1.4 system that just requires an
FXO card to connect my NTL PSTN to it. My previous X101P clone seems
On Thu, May 03, 2007 at 11:03:59PM -0400, Nitesh Divecha wrote:
Hello All,
Can anyone please post their working T1/E1 configuration...
Both '/etc/zaptel.conf' and '/etc/asterisk/zapata.conf'. I believe if
you run 'genzaptelconf' it created '/etc/asterisk/zapata-channels.conf',
so please
AUC is Friday at 12:30 PM EDT. See http://x2z.eu for how to join.
http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622
We'll be talking about SIP/IAX providers and I'd like to hear more
about asterisk appliances like the Digium and the new D-Link.
If the Digium guys are around as they usually
Maybe I missed something here. In my understanding, the only parties in the
call at DTMF stage are the originator and Asterisk. The destination is not
in the picture yet. Is this correct? What is the purpose of the said DTMF
sequence? Do you have a sample dial plan?
No, the problem is to
From: Andre Wangler [EMAIL PROTECTED]
Date: Fri, 4 May 2007 07:35:38 +0200
Hello all
Could someone tell me what happens with running calls when reloading the
whole asterisk config files? I think SIP-calls are not
Nothing. All calls are maintained according to documentation.
Yuan Liu
Hi all,
I am working wit a Mandriva 2007 and have installed the asterisk svn 1.4
with success. I also used the patch for cellphones and it works perfectly. I
was that happy that I decided to buy a TDM11B and it works.
Now, I want to study a bit the code used by this people. Does anybody know
Nops. Its not working. i have restored to original chan_local file. Im also
having another problem now (in asterisk 1.4.4).
The call originates fine, ringing is done, call is accepted, channels
bridged fine. but when either of the channels hangup, asterisk dies and
displays the following msg:
Nops. removing res_features doesnt work.
On 5/4/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Nops. Its not working. i have restored to original chan_local file. Im
also having another problem now (in asterisk 1.4.4).
The call originates fine, ringing is done, call is accepted, channels
bridged
Friday, May 4, 2007, 10:42:13 AM, Phil wrote:
On Thu, May 03, 2007 at 08:17:25PM +0100, Kyle Gordon wrote:
With the gamut of FXO cards out there, I'm looking for a recommendation for
home use. I have a nicely working Asterisk 1.4 system that just requires an
FXO card to connect my NTL PSTN
-- Called private:password@ip/[EMAIL PROTECTED]
shouldn't that be 'private:password@ip/minotaur-201'? I guess you
have a mistake in your dundi mapping
I've tried both. Sticking @privatedundi on the end was a 4am test because I
couldn't think of anything else to try. Normally that
Which book are you talking about. and what are its contents. Is it based on
understanding the code used in Asterisk. If it is then plz tell me the name
of the book. I'll be happy to buy it.
On 5/4/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote:
Hi all,
I am working wit a Mandriva 2007
Hi Rizwan,
You can find the book in the next web page,
http://www.oreilly.com/catalog/asterisk/
Iban
From: Rizwan Hisham [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Hi,
Do you know how see the peers statuses like: sip show peers but when sip
peers are configured by Relatime method.
Thanks
0xception escribió:
yes you can use the type friend
On 5/3/07, *Forrest Beck* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I setup sip realtime. Is it
Oooh. i already have this book (Asterisk The future of Telephony). its not
about the code.
On 5/4/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote:
Hi Rizwan,
You can find the book in the next web page,
http://www.oreilly.com/catalog/asterisk/
Iban
From: Rizwan Hisham [EMAIL PROTECTED]
Hi All,
I'd like to thank everyone that answer my question about IAX Trunk. Now
I have a working IAX trunking, I just need to tune it.
Thank you.
Ronaldo.
Salvatore Giudice wrote:
Yes of course. If you want to limit it, I think you have to set
'incominglimit' and/or 'outgoinglimit'.
hi everybody,
i'm tryint to install a asterisk system which acts as a dialin server
using a Digium Wildcard 205P.
acording to http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS i
need a patched version of pppd, but it does not compile on my system.
Linux box 2.6.17-gentoo-r8 #1 SMP Tue Sep
Hi all,
Could someone please tell me how to make Asterisk start at boot on Ubuntu
Feisty 7.04?
Many thanks,
Christian
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In the UK CLID is sent before the 1st ring.
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, April 24, 2007 11:15 PM
Subject: Re: [asterisk-users] UK zaptel and zapata.conf for TDM400P
On Tue, Apr 24, 2007 at 09:35:07PM
It's the magical Celeron chip.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906
From: [EMAIL
Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 8:12 AM:
Friday, May 4, 2007, 1:56:09 PM, Joe wrote:
Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 6:34 AM:
Well this is a digium list, so here will be digium cards
recommendation. But You can use a linksys spa3102, that costs about
half
Hello,
I'm trying to compile asterisk from source (1.2.18). Faxing is fairly
critical for us, so in the past we've used spandsps app_rxfax and
app_txfax to support faxing in asterisk. Unfortunately I can't find
these applications on soft-switch.org anymore and even so I have a
feeling they
mail-lists wrote:
Does anyone know how to best handle faxing in 1.2.18?
http://iaxmodem.sourceforge.net
http://hylafax.sourceforge.net
Small foot print, works great with Asterisk and supports error correction.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to
Steve,
Am 04.05.2007 um 14:44 schrieb mail-lists:
I'm trying to compile asterisk from source (1.2.18). Faxing is
fairly critical for us, so in the past we've used spandsps
app_rxfax and app_txfax to support faxing in asterisk.
Unfortunately I can't find these applications on
On Fri, 2007-05-04 at 08:35 -0400, Joe acquisto wrote:
Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 8:12 AM:
Friday, May 4, 2007, 1:56:09 PM, Joe wrote:
Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 6:34 AM:
Well this is a digium list, so here will be digium cards
recommendation.
Hi there,
One of our Asterisk 1.2 machine is experiencing problems with MeetMe.
Whenever meetme runs, the console is flooded with warning messages:
The messages started as No such file or directory and becomes
Resource temporarily unavailable. I couldn't figure out what file
MeetMe might be
Let me check my table Voicemail and CDR in the MySQL database
works fine. sip show peers isn't giving me anything. Only the one
peer I left setup in sip.conf
Here Is what I get from a Dial Command:
[May 4 09:14:28] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit
of 0x9e42d38 (len
try enabling rtcachefriends
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck
Sent: Friday, May 04, 2007 9:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP RealTime Friends
Let me check my
Nevermind. Friday and my mind has gone home! :)
I forgot the ipaddr and port setting in the table.
On 5/4/07, Forrest Beck [EMAIL PROTECTED] wrote:
Let me check my table Voicemail and CDR in the MySQL database
works fine. sip show peers isn't giving me anything. Only the one
peer I
mail-lists schrieb:
I'm trying to compile asterisk from source (1.2.18). Faxing is fairly
critical for us, so in the past we've used spandsps app_rxfax and
app_txfax to support faxing in asterisk. Unfortunately I can't find
these applications on soft-switch.org anymore and even so I have a
On Fri, May 04, 2007 at 01:07:37AM -0600, Al wrote:
Here is the fun part, box1 is faster in converting ulaw to gsm!
Is this table accurate?
Yes. The task of transcoding a single call is done by a single thread
and hence a single CPU.
Does it mean asterisk is not handeling multiple cpus very
Hello all,nbsp;nbsp; I am using HALF DUPLEX modem for TAPI call.the following
message is displayed while i am starting the AsteriskNOTICE[1416] chan_tapi.c:
Channel format set to ULAW\' ERROR[1416] win32_tapi.c: TAPI Error: 8023
(HCALL 0x0) on lineGetID . If i will receive an Inbound call
- Original Message -
From: Alex Balashov [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, May 02, 2007 2:35 AM
Subject: [asterisk-users] Runaway MOH/mp3123 process?
Has anyone noticed a problem with runaway mpg123 processes for
music-on-hold eating up ~100%
Stefan Wintermeyer wrote:
Steve,
Am 04.05.2007 um 14:44 schrieb mail-lists:
I'm trying to compile asterisk from source (1.2.18). Faxing is fairly
critical for us, so in the past we've used spandsps app_rxfax and
app_txfax to support faxing in asterisk. Unfortunately I can't find
these
Thomas Göttgens wrote:
IF you don't want to reinvent the wheel and switch to
iaxmodem/hylafax, use them instead :-)
I wouldn't consider that re-inventing the wheel. If faxes are
'critical', then I wouldn't use anything but iaxmodem and HylaFAX+ :-)
Doug
--
Ben Franklin quote:
Those
On Fri, May 04, 2007 at 09:46:08AM +, Iban Lopetegi Zinkunegi wrote:
Hi all,
I am working wit a Mandriva 2007 and have installed the asterisk svn 1.4
with success. I also used the patch for cellphones and it works perfectly.
I was that happy that I decided to buy a TDM11B and it works.
On Fri, May 04, 2007 at 02:04:55PM +0200, Christian wrote:
Hi all,
Could someone please tell me how to make Asterisk start at boot on Ubuntu
Feisty 7.04?
Many thanks,
Christian
apt-get install asterisk
Look at the init.d scripts.
Note that in Ubuntu, subdirectories under /var/run are
Tom Rymes wrote:
On May 3, 2007, at 12:20 PM, Stephen Bosch wrote:
Mats Karlsson wrote:
Take a look here:
http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html
Ugh. This is a Win32 app, isn't it?
Wow,
The guy makes a useful application and provides it to the
Well, basically, I'm looking for something that has the possiblity to
use the Nuance licenses, and that can do Text to Speech, as well as
Voice Recognition.
So far it doesn't seem possible to have a single product that does all
this within Asterisk...
Rob Townley a écrit :
Voxy - the only
Thanks Guys...
Got the T1/E1 Card working... Digium Engineers helped... According to
them TE405P card must load first and then the analog TDM400P.
Other thing which I messed up was that I changed the configuration to T1
but forgot to remove the Jumpers from the TE405P card. So that was
Stephen Bosch wrote:
Tom Rymes wrote:
On May 3, 2007, at 12:20 PM, Stephen Bosch wrote:
Mats Karlsson wrote:
Take a look here:
http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html
Ugh. This is a Win32 app, isn't it?
Wow,
The guy makes a
Hi,
I've been asked for a headset recommandation for Polycom SoundPoint IP
phones. Since I believe they use a pretty standard headset jack (correct me
if I am wrong) it's really a general recommandation on headset.
Regards,
Mike
___
Yes, they uses a standard headset jack.
On Fri, 2007-05-04 at 11:15 -0400, Mike wrote:
Hi,
I've been asked for a headset recommandation for Polycom SoundPoint IP
phones. Since I believe they use a pretty standard headset jack
(correct me if I am wrong) it's really a general recommandation
Anyone can help me with this error?
May 4 10:41:10 WARNING[5171]: chan_zap.c:9134 pri_dchannel: PRI Error:
We think we're the CPE, but they think they're the CPE too.
Both ends of the T1 can't be running in CPE (USER) mode. Typically, the
Telco is NETWORK and you are USER (CPE). If you
Hi Mike,
Yes, they use a standard headset jack. In our implementations so far we've
just had the customers continue to use their existing headsets. We take one
of them from the customer ahead of time and test it out... So long as it
works well, we replace their phones and keep the headsets. I
Thanks John,
How can I change my conf to NETWORK? Where can I find this information?
Regards,
Nitesh
John Treble wrote:
Anyone can help me with this error?
May 4 10:41:10 WARNING[5171]: chan_zap.c:9134 pri_dchannel: PRI Error:
We think we're the CPE, but they think they're the CPE too.
On Fri, 2007-05-04 at 11:15 -0400, Mike wrote:
Hi,
I've been asked for a headset recommandation for Polycom SoundPoint IP
phones. Since I believe they use a pretty standard headset jack
(correct me if I am wrong) it's really a general recommandation on
headset.
Regards,
Mike
We
A. Yes, I have the cojones. He never mentioned what platform it was for.
We need something like this for Linux. I got all excited about it only
to be terribly disappointed when I unpacked it.
From the original announcement : It runs on any modern flavor of Windows.
It is not like if he said
From: Wilson Pickett [EMAIL PROTECTED]
Date: Fri, 4 May 2007 11:37:41 +0200
Maybe I missed something here. In my understanding, the only parties in
the
call at DTMF stage are the originator and Asterisk. The destination is
not
in the picture yet. Is this correct? What is the purpose of the
Asterisk Project Security Advisory - ASA-2007-013
+--+
| Product| Asterisk
|
On Fri, 4 May 2007, mail-lists wrote:
Stefan Wintermeyer wrote:
Steve,
Am 04.05.2007 um 14:44 schrieb mail-lists:
I'm trying to compile asterisk from source (1.2.18). Faxing is fairly
critical for us, so in the past we've used spandsps app_rxfax and
app_txfax to support faxing in asterisk.
Hi,
I have already done:
apt-get build-dep asterisk and then installed libpri, zaptel and asterisk from
the latest sources.
So what should i do then? New to Ubuntu.
many thanks,
Christian
On 2007-05-04 at 17:00 Tzafrir Cohen wrote:
On Fri, May 04, 2007 at 02:04:55PM +0200, Christian wrote:
Hi
As far as a call center is concerned it really depends on the type of
employees you have working for you. Most call center reps tend to be...
umm... well less than trust worth to put it nicely. I have seen reps
pour coffee into computers, sit on their headsets, get up and walk away
with their
Friday, May 4, 2007, 3:06:02 PM, Dave wrote:
On Fri, 2007-05-04 at 08:35 -0400, Joe acquisto wrote:
Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 8:12 AM:
Yes it is an ATA with an FXS and an FXO port, and you can use as many
as you want instead of one TDM400/TDM800/TDM2400.
It has two
Gordon Henderson wrote:
The downside of iaxmodem is that (to my knowledge) you can't easilly
implement an auto-answer/detect fax/voice/ auto attendant/voicemail
system. The channel must be dedicated to faxing, and that's that. This
may or may not be an issue for you though.
If you needed
Christian,
You can follow this procedure
http://www.aussievoip.com/wiki/freePBX-Ubuntu
Regards,
Nitesh
Christian wrote:
Hi,
I have already done:
apt-get build-dep asterisk and then installed libpri, zaptel and asterisk from
the latest sources.
So what should i do then? New to
If you do make config when compiling zaptel and asterisk, it should
put the script in /etc/init.d, and add the relevant entries to the
various start levels.
Thanks,
James Texter
On Fri, 2007-05-04 at 18:44 +0200, Christian wrote:
Hi,
I have already done:
apt-get build-dep asterisk and then
I've deployed Iaxmodem as part of a Unified messaging platform for a Fortune
100 company and it works great.
* detects fax tones and vectors to fax extension which iaxmodem terminates.
Kevin Collins
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gordon Henderson wrote:
iaxmodem is a softare modem. It's a program which takes an IAX channel
and gives you a serial-line like interface. You can send AT commands
to it and get/send digital data directly. This is what people connect
HylaFax to. HylaFax is a suit pf programs that have been
I get the following error when trying to compile zaptel on CentOS 5
kernel 2.6.18-8.1.3.el5
CC [M] /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o
/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c: In function â
/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c:171: error: â has no
Does anyone know how to gain access directly to the configuration files in
AsteriskNow? I have dual NICs and need to change the binding in the config
file. I believe they blocked ssh2 access by default.
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On Fri, 4 May 2007, Lee Howard wrote:
Gordon Henderson wrote:
The downside of iaxmodem is that (to my knowledge) you can't easilly
implement an auto-answer/detect fax/voice/ auto attendant/voicemail system.
The channel must be dedicated to faxing, and that's that. This may or may
not be an
On 5/4/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
Thanks John,
How can I change my conf to NETWORK? Where can I find this information?
#signalling = pri_cpe
signalling = pri_net
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On Fri, May 04, 2007 at 01:31:19PM -0400, Nitesh Divecha wrote:
Christian,
You can follow this procedure
http://www.aussievoip.com/wiki/freePBX-Ubuntu
If you like hard work, that is.
I wonder how our frePBX debs fare on Ubuntu (deb
http://updates.xorcom.com etch main ). Theretically they
Gotoiftime()
core show application gotoiftime
Thats the best bet it sounds like, but your question was
kind of hard to understand exactly, or why you would want to
do this.
-bkruse
Goke Aruna wrote:
Hello all,
I have a set up that answer my customer. and its working well,
however, the
hello there all,
if i have a script that writes drop files into /var/spool/asterisk/outgoing
asterisk picks up the file and initiates the call just fine.
however, what is supposed to happen if more than one gets dropped in there
within like a second. Will it wait till the first is complete to
I am having an issue with the autologoff fuction in agents.conf
I am running Asterisk BE and I am testing with agent 1656. I log in, and then
make a call into the queue. The agent's phone rings, and if I answer it, all's
fine/ I am trying to have this agent automatically be logged off
On May 4, 2007, at 10:08 AM, Stephen Bosch wrote:
Tom Rymes wrote:
On May 3, 2007, at 12:20 PM, Stephen Bosch wrote:
Mats Karlsson wrote:
Take a look here:
http://www.voip.com.sg/voip_products/
voip_ip_phone_provisioning_tool.html
Ugh. This is a Win32 app, isn't it?
Wow,
The guy
Doh! That was it. It was a permissions issue.
Thanks for your help!
Victor
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On Fri, May 04, 2007 at 02:28:32PM -0400, Andreas van dem Helge wrote:
On 5/4/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
Thanks John,
How can I change my conf to NETWORK? Where can I find this information?
#signalling = pri_cpe
signalling = pri_net
nitpicking:
;signalling = pri_cpe
Hello,
I try to configure a Patton smart node with a E1 for the Chile.
I need to know wich parameters I must set for Chile.
If someone have informations it's welcoms.
Laurent
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asterisk-users
On Fri, 2007-05-04 at 15:25 +0500, Rizwan Hisham wrote:
Nops. removing res_features doesnt work.
Rizwan--
This is strange; It would seem your main/cdr.c and res/res_features.c
are out of sync!
The code chunk I sent you does not contain any references to
ast_cdr_merge, and
does not have
Forget them! Use Hylafax and iaxmodem instead.
I wondering, how do you guys handle multiple calls? We frequently get
many concurrent faxes, sometimes even to the same number. As far as I
know, one instance of iaxmodem can only support one fax session at a
time. So essentially you need a pool of
Good question shawn,
The callfile does get deleted once the call has been finished (I believe
its FINISHED, not processed)
No, they are not executed sequentially..exactly. Well, from your
point of view, you can drop tons of them in there and all of the calls
will fire up. I have
We have an Asterisk v1.2.16 box registering with an ITSP using SIP. The
registration succeeds, and is confirmed with SIP SHOW REGISTER. However,
we frequently (every few minutes) see this on our console:
REGISTER attempt 1 to [EMAIL PROTECTED]
REGISTER attempt 2 to [EMAIL PROTECTED]
Any
Thanks Everyone for the help...
Got the T1 UP and insvc with Cisco AS5350, but I am failing to send the
call.
On the Cisco side I do not see any incoming call and on Asterisk side I
get message saying Channels unavailable, while all channels are available.
Can anyone post a working
Luki wrote:
Forget them! Use Hylafax and iaxmodem instead.
I wondering, how do you guys handle multiple calls? We frequently get
I have 23 iaxmodems running on my each of my Asterisk/HylaFAX+ servers.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a
Hi,
Many thanks got it working now.
All the best,
Christian
On 2007-05-04 at 13:31 Nitesh Divecha wrote:
Christian,
You can follow this procedure
http://www.aussievoip.com/wiki/freePBX-Ubuntu
Regards,
Nitesh
Christian wrote:
Hi,
I have already done:
apt-get build-dep asterisk
Just wondering if anyone has tried using Asterisk 1.2 on CentOS 5. Is it
worth considering for a Production install yet? Did they fix that
spinlock.h Kernel problem?
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asterisk-users mailing
Luki wrote:
So essentially you need a pool of iaxmodems running on different
ports, and then Dial() them until you find one that accepts your call.
Or did I get that wrong? That seems really like a drawback to me
The biggest drawback with
app_rxfax is that if it crashes for whatever reason
Requests
1. Directories linked to their databases
2 weather broadcasts
3 local traffic info.
4 local news headlines
5 sms send / receive
6 alarm on the phone of calendar events - not a call back, simply a beep and
notice without pulling out a pda or opening outlook
Couple of other one of's
So you'd rather have the entire PBX crash in order to avoid creating
sufficient iaxmodem instances to handle your fax call load?
No, but so far this occurred only once in about a year of service. Not
ideal, but acceptable considering Asterisk itself segfaults or
deadlocks every now for no
[EMAIL PROTECTED] wrote:
Why? There used to be a saying 'usb is for mice, firewire is for men',
though USB has grown a bit in bandwidth since then, it is still not very
well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I
suspect a lack of demand. Havng a E1 termintae in your
From: James Texter [EMAIL PROTECTED]
Date: Fri, 04 May 2007 12:28:39 -0500
If you do make config when compiling zaptel and asterisk, it should
put the script in /etc/init.d, and add the relevant entries to the
various start levels.
Not with 1.4 at least. makefile is not looking in the right
What I do is add an entry in the crontab file as such:
* * * * * if [ ! `/bin/pidof -s asterisk` ] ; then /usr/sbin/asterisk; fi
Its simple and it works. Additionally if asterisk crashes then cron
restarts the server in about a minute. Just be careful with your configs.
Mark Coccimiglio
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