Is Directed Call Pickup supported in asterisk?
(http://voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup)
The group call pickup works well, but I can't make directed call pickup:
example:
Someone is calling 202 (my phone, GS BT101), i stay near ext. 230 (Cisco
7905g), i want capture this
hi
would it be possible to add DTMF mode autodetection in asterisk? This
would allow clients running any kind of DTMF signalling. Why not just
set it at first occurence of one of them?
thanks
roy
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On Tue, 23 Nov 2004 11:11:02 +1100, Paul Hales [EMAIL PROTECTED] wrote:
Also - the mobile phone plans we are using get very expensive after approx
1500 minutes, so we have to make sure that none of the lines go over that!
But the zap/r1 options should be OK for a start at least.
show
Good day all
How do I dial from the cli
It says dial [number] but that doesnt do anything?
Thanks
altus
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On Tue, 2004-11-23 at 09:11 +0500, amna saleem wrote:
-- Forwarded message --
From: amna saleem [EMAIL PROTECTED]
Date: Thu, 18 Nov 2004 22:26:11 -0800
Subject: changing configuration file
To: [EMAIL PROTECTED]
From the above we see this has already been asked by you. Perhaps
Although I had no problems getting Asterisk up and running under RedHat 9
I'm running into problems under FC 3.
When running make linux26 I get:
make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/lib/modules/2.6.9-1.678_FC3/build'
CC [M]
Thomas Jagoditsch:
my general impression was that too, but most users seem to
work with the setup tim recommends.
Having wasted a load of money on Fritz! Cards from eBay, I too had the
problem with stutter and terrible audio with chan_capi. I moved over to
ZapHFC cards and haven't had a
Hi!.
I am testing firefly and I can say it's a great
program, but I have a problem.
When I use Sip and I activate the canreinvite option
in Asterisk, I can't hear anything.
My network is the following:
-Two Firefly clients with SIP. Each firefly is in
different networks behind NAT.
-One Asterisk
Anybody know of a UK supplier of Voip Telephony with Asterisk
by Paul Mahler ?
I've searched on the web, and the only suppliers I can find are US
based, and the postal charge is as much as the book.
cheers
--
Clive
Email : [EMAIL PROTECTED]
Alt : [EMAIL PROTECTED]
Tel : 0845 0043366
On Mon, 22 Nov 2004, Kevin Brennan wrote:
It happens servers come with twin GB NIC's, bonding is for redundancy not
capacity. /Kev/
If all you're after is redunancy then have a look at failover rather than
bonding - you'll have nothing to worry about with regard to frames
potentially arriving
Hello
In some SIP invite messages I see the below codec negotiation string, I
am wandering what the 101 telephone-event/8000 means Which codec
is that?
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
On say 2 out of 10 calls, when on a call, the Audio at our
end will drop for about 5 seconds, we can hear them, they cant hear us.
It doesnt happen every call, random, which is making
it very hard to trouble shoot, I am guessing it has something to do with RTP
stream?
Nothing has
[EMAIL PROTECTED] wrote:
In some SIP invite messages I see the below codec negotiation
string, I am wandering what the 101 telephone-event/8000
means Which codec is that?
RFC 2833 DTMF events
--
Andreas SikkemaRits tele.com
Scheepmakersstraat 11 3011 VH Rotterdam
t:
Hei!
I have a little problem with the subject. I use Asterisk
CVS-HEAD-09/06/04-12:42:56 as a production *, but I do tests with a
newer version
Asterisk CVS-HEAD-11/18/04-10:01:32. Ok the problem is:
in extension.conf I use macro for redirection, found on wiki pages:
[macro-stdexten]
;
;
I have problem with db creation - I always get
error creation.
I try to do it on many servers with many
configuration.
My mysql works ok and i have connection by sock and
by tcp for example from phpmyadmin. I configure
astcc with proper db user host and password.
I get new astcc from cvs
I have two Asterisk servers interconnected with IAX (non-trunk). I place a
call on Server B (using DIAX) which goes to an extension on Server A and
terminates with a Dial to a local SIP phone (Sipura SPA 2200).
The SIP phone rings immediately but when it is answered there is a delay of
about
Hi All,
I have been researching Asterisk for a few days now and have read
hundreds of web pages and other documents. While some things are getting
clearer, others are not.
I have managed to install Asterisk 1.0.1 on Debian testing (simple as
'apt-get install asterisk'). I understand FXS/FXO,
Tracy R Reed a écrit :
Has anyone here done this? I got it compiled just fine but when I make a
call I do not get any audio going either way. The * box is not behind any
sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I
have it set up properly to work through NAT and it
Hi everyone,
Could somebody enlighten me on this one? I have
configured my asterisk to run on oh323 using codec
g729. Incoming calls are working okay. But the thing I
want to work is say pressing some options, say dial 1
to go to voicemail or dial a certain number to dial a
specific extension.
I
On Tue, Nov 23, 2004 at 12:42:07PM +0100, administrator tootai spake thusly:
with my other regular x86 box running H323. One odd thing I note is that
when looking at the UDP traffic with tcpdump I see the * box receiving my
Same problem here. My * box is connected to GnuGK. CVS Head 11/02/04,
Brian West wrote:
http://www.obj-sys.com/open/index.shtml
Anyone up for trying to write a channel driver with this? I would love to
see something done with this... I had the h323ep demo app that's included
sending and taking calls between my box and netmeeting ... but it was just
signaling. :P
Found and fixed the problem. Did not have libssl-dev installed.
Thanks
--
Clive
Email : [EMAIL PROTECTED]
Alt : [EMAIL PROTECTED]
Tel : 0845 0043366
Alt : 01952 402032
SIP : [EMAIL PROTECTED]
Mobile : 07970 856261
___
Asterisk-Users
Hello,
Im using asterisk to relay sip to another SIP
provider, Ive setup a friend in sip.conf for my softphone and a user
peer section for the SIP provider, when my softphone calls out to the sip
provider and the sip provider returns an error (404 Not Found for example) the
sip message
Tracy R Reed wrote:
On Tue, Nov 23, 2004 at 12:42:07PM +0100, administrator tootai spake thusly:
with my other regular x86 box running H323. One odd thing I note is that
when looking at the UDP traffic with tcpdump I see the * box receiving my
Same problem here. My * box is connected to GnuGK. CVS
Hi,
But i just can't seem to
make it work using oh323/coded g729? Its like it does
not respond to DTMF signals? I have dig into many
mailing list and not any clear solutions. Could
What DTMF mode are you using? Do you have a g729 license installed on your
system? Remember that g729 only works
Al Escasa wrote:
Hi everyone,
Could somebody enlighten me on this one? I have
configured my asterisk to run on oh323 using codec
g729. Incoming calls are working okay. But the thing I
want to work is say pressing some options, say dial 1
to go to voicemail or dial a certain number to dial a
Hi all,
Is there any commercial g723 license for asterisk?
Where can it be purchased? Has somebody used it?
Thanks
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I
never used, but there are one "open" beta codec g723.1 from the makers of the
"open" g729 codec that uses Intel IPP library.
http://www.readytechnology.co.uk/open/g723.1/
Regards,
Vinicius
-Mensagem original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]Em nome de
Which is not for
commercial use
Van:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Namens Vinicius Viana
Verzonden: dinsdag 23 november
2004 14:57
Aan: Asterisk
Users Mailing List - Non-Commercial Discussion
Onderwerp: RES: [Asterisk-Users]
Commercial g723.1 license for asterisk
Michael Manousos a écrit :
Tracy R Reed wrote:
[...]
Same problem as in you ran tcpdump or something and saw the odd behavior
of receiving but not sending any packets? VERY interesting. Were you
on an
x86-64 bit box or regular x86? I was thinking this odd behavior was some
odd interation with
Hi
Just a shot in the dark, wouldnt this be coming into a CSU/DSU then into a
Digium PCI card of some sort, that is where asterisk would pick it up?
- Original Message -
From: Richard Howes [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 23, 2004 3:41 AM
Subject:
Please can someone look at my last two posts and try and shed some light
onto why my system is dropping calls..
If I don't get it right we will be forced to drop Asterisk which I
really don't want to do..
Thanks..
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Is there a way to log all PRI events to a logfile?
Cheers,
Ben Merrills
Griffin
Internet
T: 0870 8040862
F: 0870 8040805
W: www.griffin.com
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On Tue, 23 Nov 2004 13:17:57 +, WipeOut
[EMAIL PROTECTED] wrote:
Please can someone look at my last two posts and try and shed some light
onto why my system is dropping calls..
If I don't get it right we will be forced to drop Asterisk which I
really don't want to do..
I Have not read
Andrew Thompson wrote:
You should be able to set the inbound callerid from the switch/gateway
to a specific unknown in sip.conf file with just a callerid= line.
The place I looked on the wiki didn't show a specific description for
the callerid= line, but that's what I thought I read for
Received two new Polycom 500 phones. Dhcp and ftp configured properly to
load the various files including v2.5.0 bootrom.ld, etc. One of the phones
loaded all firmware and config files properly, registers with *, and is
usable.
The second phone loads bootrom.ld (from the same ftp server on the
Jason Williams wrote:
On Tue, 23 Nov 2004 13:17:57 +, WipeOut
[EMAIL PROTECTED] wrote:
Please can someone look at my last two posts and try and shed some light
onto why my system is dropping calls..
If I don't get it right we will be forced to drop Asterisk which I
really don't want to do..
On Tue, 23 Nov 2004, Ben Merrills wrote:
Is there a way to log all PRI events to a logfile?
Maybe pri intense debug span ??? is what you are after? If you set up a
logging file in /etc/asterisk/logger.conf that logs everyting you should
get all the pri events.
Peter
On Mon, 2004-11-22 at 17:03, Peter Svensson wrote:
Shouldn't the echo canceler cut out when it detects the 2.1kHz guard tone?
The comments in ecdis.h and zaptel.c seens to indicate that.
Yes it does. I see CLI output telling me so every time I receive a fax.
-Seth
--
Seth Remington
Does anyone have an update patch file to get Spandsp installed?
I'm running asterisk
CVS-HEAD-11/19/04-21:53:37 on redhat 9.0
I installed
spandsp-0.0.2
when runnig the
patch I get
patching file
MakefileHunk #1 FAILED at 41.Hunk #2 FAILED at 69.2 out of 2 hunks
FAILED -- saving rejects
Hi,
Is your asterisk server in DMZ environment?
Senad Jordanovic
Bicom Systems, The complete systems provider
www.bicomsystems.com
USA 1-212-400-7921
UK 0870 682 782
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: 23
But there's no way of just specifying PRI events? I'd prefer to have a
logfile that simple had all the PRI output (e.g. output of pri debug
span n)
Cheers for the suggestions though,
Ben
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Svensson
From their Ebay site, I see they (Signate) charge US$23 for Global Express
Mail (to be fair, only a little more than the actual cost to them). Maybe
you could ask them to ship it Global Priority Mail instead (about 4-5 days
to London), which has a cost of $9. If they won't do it, email me off
On Tue, 2004-11-23 at 09:00, Eric Hall wrote:
Does anyone have an update patch file to get Spandsp installed?
I'm running asterisk CVS-HEAD-11/19/04-21:53:37 on redhat 9.0
I installed spandsp-0.0.2
when runnig the patch I get
patching file Makefile
Hunk #1 FAILED at 41.
Hunk #2
Data doesn't currently work with linux 2.6.
Marcelo Pacheco
Em Seg 22 Nov 2004 20:37, [EMAIL PROTECTED] escreveu:
On Nov 22, 2004, at 6:19 PM, Ken D'Ambrosio wrote:
Hi, all. I'm thinking of provisioning my (non-PRI) T1 to be part data,
part voice (currently, it's only voice). With the
Senad wrote:
Hi,
Is your asterisk server in DMZ environment?
Hi Senad,
No, this server that is having the issue is behind a NAT firewall
connecting through IAX to a termination provider.. Being IAX I would
think NAT is irrelevant.. Also there is no patten to the dropping of
calls, a call can
Why
not? you only need to pay US$30.500 plus US$1.78 per port to use. Oh, you also
need to pay US$199.00 to license IPP library.
http://www.dspg.com/technology/LicensePricing.html
http://www.intel.com/software/products/ipp/pricelist.htm
-Mensagem original-De:
[EMAIL PROTECTED]
For some time (since pre 1.0), I've been seeing the following messages
fairly regularly from some, but not all, of my SIP devices:
Nov 23 06:37:59 NOTICE[2568]: chan_sip.c:7645 handle_request: Registration
from 'John Doe sip:[EMAIL PROTECTED]' failed for '200.100.50.25'
I have a mix of Sipuras,
I was finaly able to patch the Makefile in the apps dir. I used 2pre4
version.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Seth
Remington
Sent: Tuesday, November 23, 2004 8:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
E. Versaevel schrieb:
Which is not for commercial use
This mailinglist is called Non-Commercial Discussion ;-)
And according to: http://www.readytechnology.co.uk/open/g723.1/
... [If] you live in a country that has a more generous definitition of
'free', then you are welcome to compile and
Hi everyone could somebody help me on this problem ???
I have a tdm04b card with 4 fxo's connected to 4 POTS of a media
gateway. Suppose that I want to place the following calls:
Zap/1 dials Zap/2 (by placing in /spool/outgoing a call file which
dials a number corresponding to Zap/2)
Zap/3 dials
I know its on the wiki somewhere, but Ive been
searching for 2 days on the wiki and google. I just cant seem to find the web
interface for the meetme application. Does anyone have a link I could use. I found
ASGIs dynamic conferences which is neat. Thanks in advance.
Hi, all
I am trying to install E100P card, the 'modprobe zaptel' is ok, but
when I did 'modprobe wct1xxp', I got such error, so can not load the
driver for the card.
/lib/modules/2.4.20-8/misc/wct1xxp.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters,
I would like to see the answer to this also, I have experienced the same
problem but not had much time to look at it.
regards
Garry Taylor
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew Boehm
Sent: Tuesday, 23 November 2004 5:46 AM
To:
Yes, exactly! I think 100,000 phones all regeristing every 60 seconds would
put quite a load. And if 50% of them are all behind NAT/FW, asterisk
wouldn't play nice, would it?
So, is SER a better option for this?
-Matthew
- Original Message -
From: Matt Riddell [EMAIL PROTECTED]
To:
Hello,
With all the talk about Firefly, I decided to check it out, it seems to work
under wine (IAX only for some reason) so I'm thinking about using it on the
road. Now, my Asterisk box is behind a firewall, so I have set the firewall
to forward UDP port 4569 to my Asterisk box put I'm having
I did that
[EMAIL PROTECTED] apps]# patch Makefile.patch
patching file Makefile
Hunk #1 succeeded at 52 with fuzz 2 (offset 11 lines).
Hunk #2 succeeded at 88 with fuzz 2 (offset 19 lines).
When back to the top-level and did a make
I get this
make[1]: *** [app_rxfax.o] Error 1
make[1]:
George Burt wrote:
snip
[grandstream1]
host=10.0.0.26 ; we have a static but private IP address
canreinvite=yes; allow RTP voice traffic to bypass Asterisk
snip
IP Address:
statically configured as:
IP Address:10.0.0.26
Subnet Mask: 255.255.255.0
If you had 100,000 phones registering to the Asterisk server, I would think
you would have at least two or three more Asterisk servers for people to
point to their devices to. Who is to say that SER won't crash with 100,000
registrations either? You could always use a STUN server on each
There are 4 options in ASTCC routes, when I go to edit
a route. How does that work?
Thanks
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hello to all:
to help they can to me with this, which desire to do is that the voicemail
warns to me with a call extesion that corresponds whenever nuevo.he has a
message reviewed the Web but nonencounter like doing it.
Thanks
Rodrigo
___
Eric Hall wrote:
When back to the top-level and did a make
I get this
make[1]: *** [app_rxfax.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/apps'
make: *** [subdirs] Error 1
[EMAIL PROTECTED] asterisk]#
I just fought a battle with spandsp/rxfax and won.
My winning strategy can be
Please, could you send us cat /proc/pci?. Could you compile libpri?
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Ning Zhou
Enviado el: martes, 23 de noviembre de 2004 16:10
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Error when
Hi all,
I am having some problems getting my fax-machines to work properly. I can make
them dialup, and they cal also recive calls, but very often they disconnect
during the data-transfer. Anyone got some ideas?
The setup is like this:
Pri - E100P - Asterisk - TDM400 - Fax-Machine
I have
STUN requires 2 NIC interfaces on the machine running the server right? And
both interfaces need seperate public IP's right? 'And' the phones/ATA's need
to support STUN right? I don't think the Cisco phones support STUN.
-Matthew
- Original Message -
From: Brian Wilkins [EMAIL PROTECTED]
Anyone in need of some of these?
Garrett Smith
Sales Executive
[EMAIL PROTECTED]
B2 Technologies
454 Sonwil Drive
Buffalo, NY 14225
(716) 250-3408 Direct
(716) 630-1548 Fax
(716) 903-9495 Cell
AOL IM: B2sales
Specializing
in New and Used equipment from vendors
Hello all,
I want to setup Asterisk to forward a call if the dialed extension is
busy. I do not want to wait on the line until the extension timeout
expired. What I want is when I dial am extension currently Busy (Talking
with someone), asterisk inmediately forwards my call to an extension I
Hello everyone!
I tried to send a fax over SIP with an Asterisk Server in the middle (no
Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway
is external). Whenever I start sending a Fax to a PSTN destination, the
Call gets answered and asterisk tries to build a native bridging:
Yes...please contact me off list. Condition?
new / used what do they come with?in boxes? Cost?
I need about 20.
Jon Bebeau
- Original Message -
From:
Garrett
Smith
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Tuesday, November 23, 2004
Hi
I have a brand new ATA186 with the following firmware:
Version: v2.15.ms ata186 (Build 020919a)
I have been through the archives about how to configure it, but my colorful
configuration web page does not have the same fields that people say I need
to adjust. Even the examples on Cisco's
You need two public ip addresses. I am running STUN on one nic. I don't
know about the Cisco phones however.
I have a Gradnstream and a couple of soft phones bouncing off my STUN
server.
Lyle
- Original Message -
From: Matthew Boehm [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Are IVR and Auto Attendant interchangeable terms? They both do the Press
1 for thing. Sales is asking me how to word it and I've always used both
terms interchangeably.
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[EMAIL PROTECTED]
onto why my system is dropping calls..
If I don't get it right we will be forced to drop Asterisk which I
really don't want to do..
Thanks..
Did you know that you can obtain commercial support for Asterisk?
http://www.digium.com/index.php?menu=software_support
I am sure it will be
Matthew Boehm wrote:
STUN requires 2 NIC interfaces on the machine running the server right? And
both interfaces need seperate public IP's right? 'And' the phones/ATA's need
to support STUN right? I don't think the Cisco phones support STUN.
Why 2 NICS. There should be no reason that you can't
You probably aren't running SIP firmware. You'll have to upgrade it first.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rodney Acosta
Coya
Sent: Tuesday, November 23, 2004 11:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
app_rxfax uses an incorrect structure parameter. Change callerid on
line 83 (I think) to cid.
Greg
Eric Hall wrote:
I did that
[EMAIL PROTECTED] apps]# patch Makefile.patch
patching file Makefile
Hunk #1 succeeded at 52 with fuzz 2 (offset 11 lines).
Hunk #2 succeeded at 88 with fuzz 2 (offset
Sure they do. I have a bunch of Cisco phones that support STUN.
On Tuesday 23 November 2004 03:51 pm, Matthew Boehm wrote:
STUN requires 2 NIC interfaces on the machine running the server right? And
both interfaces need seperate public IP's right? 'And' the phones/ATA's
need to support STUN
Matthew Boehm wrote:
STUN requires 2 NIC interfaces on the machine running the server right?
And both interfaces need seperate public IP's right? '
Why ever for?
I realize that in order to set up a STUN server you need a public IP,
but why two of them and why two different interfaces?
Dazed and
No. Auto-attendant is a subset of a class of applications that fall
under IVR (interactive voice response).
Greg
Paul Rodan wrote:
Are IVR and Auto Attendant interchangeable terms? They both do the Press
1 for thing. Sales is asking me how to word it and I've always used both
terms
Still getting errors
make[1]: Entering directory `/usr/src/asterisk/apps'
Makefile:103: warning: overriding commands for target `app_rxfax.so'
Makefile:85: warning: ignoring old commands for target `app_rxfax.so'
Makefile:106: warning: overriding commands for target `app_rxfax.o'
Makefile:88:
Stun requires 2 separate public IP addresses? For comparison? Can they be on
the same subnet? I found a stun program but never got around to configuring
it. Is it as simple as it seems? Program it with both IP's, set it and
forget it?
-Original Message-
From: [EMAIL PROTECTED]
I would say that the Auto Attendant function, used to allow people to select
who they want to talk to, is a subset of IVR, which more generically allows
the person calling to retrieve information or to control the operation of a
computer or telephone switch.
Scott M. Stingel
President,
Emerging
Paul Rodan wrote:
Are IVR and Auto Attendant interchangeable terms? They both do the Press
1 for thing. Sales is asking me how to word it and I've always used both
terms interchangeably.
Yes, they are pretty much the same thing, although IVR can also mean
automated data update/retrieval systems.
That's correct. Go here to get instructions on setting up STUNd 0.94 :
http://www.voip-info.org/tiki-index.php?page=Vovida.org%20STUN%20server
On Tuesday 23 November 2004 04:42 pm, Paul Rodan wrote:
Stun requires 2 separate public IP addresses? For comparison? Can they be
on the same subnet?
And this is after you did a make clean, at least in the apps directory?
The part about overriding commands doesn't make sense to me...
Greg
Eric Hall wrote:
Still getting errors
make[1]: Entering directory `/usr/src/asterisk/apps'
Makefile:103: warning: overriding commands for target
Check what IOS ata have installed... Because by default it does not comes
with H.323 - SIP IOS...
If you want I can send you both ios...
Contact me at: [EMAIL PROTECTED]
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Rodney Acosta
Coya
Enviado el:
Hi Senad,
No, this server that is having the issue is behind a NAT firewall
connecting through IAX to a termination provider.. Being IAX I would
think NAT is irrelevant.. Also there is no patten to the dropping of
calls, a call can last less than 1 min or over 30 min or anything in
between..
My
I dont have a cisco acount yet
can some bady hel me with the
ata18x-v2-16-030401a-1.zip file.
thanks in advance
Rodney Acosta Coya.
Dpto. Tecnologia de la Informacion.
[EMAIL PROTECTED]
Tel:(53)(24) 62 611
-Mensaje original-
De: Paul Rodan [mailto:[EMAIL PROTECTED]
Enviado el:
Title: Message
Hi
all,
We are running
Asterisk 1.0.0 with a TE410P. Very often we exerience calls dropping in the
middle of the call. I enable the full logging and saw a couple of suspicious
messages right before the hangup. Thos could happen on a Zap-IAX2 bridge as well
as on a
On Tue, 2004-11-23 at 12:04 +, Clive Carter wrote:
Hi
Having my first go at compiling Asterisk from cvs source.
Compiled and installed zaptel ok
Running make asterisk returns the following error message
/usr/bin/ld cannot find -lssl
collect2: ld returned 1 exit status
The last part
Are you trying to send fax over T.38?
As far I understand * don't support T.38 event when passing packets
trouth.
I'm interested in T.38 support too, so if anybody could explain why *
can't just pass theese packets (as i undrstand there is no need foe
recoding etc.) I would be very appreciative.
- Original Message -
From: Eric Hall [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, November 23, 2004 9:18 AM
Subject: RE: [Asterisk-Users] Spandsp and Asterisk
I did that
[EMAIL PROTECTED] apps]# patch Makefile.patch
I tried to setup MySTUN (http://developer.berlios.de/projects/mystun/)
and it said STUN servers require 2 seperate public IPs on the same machine.
(and yes, I realize I made a mistake when I said 2 NIC cards)
-Matthew
- Original Message -
From: Gilad Ben-Yossef [EMAIL PROTECTED]
To:
I recently raised an issue which has just been fixed - this should clear
your problem as well:
http://bugs.digium.com/bug_view_page.php?bug_id=0002910
- Original Message -
From: Brian McCrary [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 22, 2004 3:41 PM
Subject:
Cristian Manoni wrote:
Is Directed Call Pickup supported in asterisk?
(http://voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup)
No.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi Rodney,
I dont have a cisco acount yet
can some bady hel me with the
ata18x-v2-16-030401a-1.zip file.
You will need a PC running Windows.
1. Unzip it.
2. Read the text file ata186us.txt
3. Follow instructions in it :)
This will convert your ATA from MGCP/SCCP to H.323/SIP .
--Stewart
Thank you, Mr. Stingel.
Mr. Carter, the book will be in Amazon UK inventory by year-end.
If you'd rather not wait until then, in a matter of days you will be able to
purchase an e-book version for the OSoft ThoutReaderT at
http://www.osoft.com/store/. There will be no shipping charge. We will
Hi Rodney,
I dont have a cisco acount yet
can some bady hel me with the
ata18x-v2-16-030401a-1.zip file.
ftp://ftp.rekom.ru/pub/ata18x/
You will need a PC running Windows.
1. Unzip it.
2. Read the text file ata186us.txt
3. Follow instructions in it :)
This will convert your ATA from
Senad wrote:
Hi Senad,
No, this server that is having the issue is behind a NAT firewall
connecting through IAX to a termination provider.. Being IAX I would
think NAT is irrelevant.. Also there is no patten to the dropping of
calls, a call can last less than 1 min or over 30 min or anything in
Geoff Nordli wrote:
onto why my system is dropping calls..
If I don't get it right we will be forced to drop Asterisk which I
really don't want to do..
Thanks..
Did you know that you can obtain commercial support for Asterisk?
http://www.digium.com/index.php?menu=software_support
I am
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