Displaying 20 results from an estimated 1000 matches similar to: "connect Sipura with Asterisk - both behind NAT"
2006 Oct 26
10
ECHO Cancellation in SIP Calls
Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update
of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no echo (for example if i call myself
via SIP->Asterisk->SIPProvider->TELEKOM->ISDN)
but if i call other people there occures Echo many times. The Routing is
always the
2006 Mar 29
5
Asterisk Between PBX and FXS
Hi guys,
I''m setting up asterisk to run with another pbx server. This pbx server
support a feature that allows 2 extensions connect to the same FXS. No I put
asterisk in the middle.
Asterisk receives the call and dial to a SIP/peer.
How the pbx installed support 2 extensions to one fxs... How can I figure out
in asterisk which extension was dialed before the call came to asterisk?
2006 Jan 09
1
how to adjust volume
how to adjust voice volume for sipura 2000 and cisco ata186?
2006 Nov 16
1
Asterisk 1.2.13 can't load module app_curl.so
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2006 Jan 12
1
No D-channels available! Using Primary on channel 16 anyway!
Hi!
I have a E1 PRI connected to my TE400P card on span 1, and two channelbanks
on span 3 and 4 and * 1.2.
Every few hours I get this message and asterisk dies just after that:
Warning: No D-channels available! Using Primary on channel 16 anyway!
When this happens restarting zaptel and asterisk services, generally puts
the system back online
my zaptel.con reads:
span=1,1,0,ccs,hdb3
2006 Oct 30
3
Live creation of trunk groups
Hi,
Is there a way to create trunk groups while asterisk is running.
For exemple let's say that zapata.conf defines g0 as channels 1-23
I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23
Any hints appreciated.
Andre Courchesne
2007 Feb 08
3
Automatic Dial, Play message
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?
For example, you have a database of
FirstName LastName PhoneNumber
Jon
--
***
Forrest Beck
IAXTEL: 17002871718
jonforrest.beck@gmail.com
2006 Jan 12
1
Problem with an automatic responder
Hi,
I have this setup:
(PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones
and a few of VoIP phones directly connected to Asterisk.
Calling a number (only one until now!) - an automatic responder (IVR) - from
VoIP phones works, from analog phones doesn't work: NOANSWER after a few
seconds. I'm using no 'r' in dial options (this caused a problem with an IVR
2006 Dec 20
5
Sangoma A101 with Unicall
I am having a problem trying to get a Sangoma A101 to work with
Unicall. I have installed the sangoma drivers and everything seems to
be well but when I try to run ztcfg I get the following error:
CAS signalling on span 1 conflicts with HDLC with FCS check on channel
16.
Here is my /etc/zaptel.conf
# MFC/R2 normalmente no usa CRC4
span=1,0,0,cas,hdb3
cas=1-15:1101
dchan=16
cas=17-31:1101
2006 Oct 17
2
duplicate "ghost" calls with long duration
Hello everybody,
I am running 1.2.10-BRIstuffed-0.3.0-PRE-1s with florz-patches on Linux
2.6.16 with 4 HFC-Cards in TE-mode connected to 4 PtP
ISDN-"Anlagenanschluesse". There are about 40 SIP-clients connected
(mostly Sipura/Linksys PAP2, and some SNOMs and softclients) to this server.
Everything works fine, except that my CDR reports some very long
_concurrent_ calls from one sip
2007 Feb 08
4
error when compiling zaptel-1.4
when i compile zaptel
make linux26
make install
i got these errors:
make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No
2006 Mar 23
7
[OT] Polycom provisioning
Does anyone have the polycom soundpoint ip's successfully remotely
provisioning? I've got the phone pulling default configs, and it's
downloading phone specific information, but it's not actually using that
information. Any help would be appreciated :)
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
amdtech@shsu.edu
(936) 294-4198
2005 Mar 13
2
Sipura 841 issues
Hi
Just 2 issues I have with SPA841.
1. I autodial extension 600 then inside an AGI wait for more digits.
The digits are transmitted correctly to * but they do not show up on the
SPA841 display, only the 600. How do I set the 841 is show the digits
after the 600#
2. Is the SPA841 pixel display backlit?
Master
2005 Dec 15
2
Outbound Routing
Hello,
I have a 4 port FXO digium card with 3 PSTNs attached to it and
AsteriskAtHome setup. Everything is working fine except outbound calls.
When I dial a outside number, it works fine, but when another employee trys
to dial out while I am on a line, it will not go.
I have a outgoing route setup in the AMP interface.
Dial Pattern:
1NXXNXXXXXX
NXXNXXXXXX
NXXXXXX
Trunk
2006 Feb 23
1
sipura 841 mass provisioning
Hi there,
I have bought 70 sipura 841 phones for a customer of mine.
When following the mass provisioning guide in the admin manual for the
sipura, I see it download the spa841.cfg file from my tftp server
Sometimes the phone also downloads is phone specific file via tftp, and
it works okay then.
But, after a reboot of the phone, it is very very likely that it won't
startup
2005 Aug 16
8
Asterisk and LCR
Hello,
How do you guys implement LCR in Asterisk?
Thanks,
2006 Apr 01
2
Problem: ringtones stop unexpectedly
I should've mentioned that before. I've tried doing that and it has no
effect. I've tried both upper and lower-case 'r's.
I've also tried a workaround that I thought would work, but it doesn't:
Answering the call and then using the playtones(ringing) command before
connecting to my cellphone.
-----Original Message-----
Date: Sat, 1 Apr 2006 19:59:46 +0100
From:
2006 Mar 23
1
kernel recompilation on a asterisk server
i've got a asterisk server running on slackware 10.2 sice 3 weeks, i
will upgrade the kernel to 2.6.x from 2.4.x , i will upgrade to the
last stable release of kernel...
i've got asterisk 1.2.4, zaptel 1.2.4 and libpri 1.2.2, correctly
compiled and configured, obviusly if i try to load the new kernel i
can't load wtcxx modules, should i recompile zaptel 1.2.4 with make &&
make
2007 Feb 08
2
Asterisk outbound calling does not wait for answer before playback
Hello Asteriskers, :-)
We're trying to set up an outbound notification calling for system
alerts with Asterisk 1.4.0. We generate a call file in
/var/spool/asterisk/outgoing and the outbound call is originated through
Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that
Asterisk does not wait for the other side to answer before it starts
playing the message. So the
2006 Oct 20
1
#Transfer - Timeout is configurable?
Hi guys,
This should be has an easy answer for you, my users are complaining
that when they press # and then ear gorgeous Allison "Transfer" the
timeout is very small, they must enter immediatly the extension to
transfer the call.
Is it possible to change this?
;transferdigittimeout => 3 ; Number of seconds to wait between
digits when transfering a call
This is timeout