Displaying 20 results from an estimated 8000 matches similar to: "* as pri_net?"
2004 May 23
5
PRI problem???
I have just finished installing a new asterisk box at my work. The box is
quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4
port Digium T1 card for channel bank and PRI access.
I activated a PRI from a local CLEC (DMS-500 based, National protocol).
This PRI is on slot 2 of the card and is set as the primary timing source.
It is ESF/B8ZS.
All the software is latest
2005 Sep 28
2
Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unkn own signalling method 'pri_net'
Did you compile and install libpri *before* Asterisk? I had same problem
(among others) b/c I didn't install in the correct order. Try the awesome
asterisk_update.sh shell script.
Are you trying to emulate CPE or NET? Try signalling=pri_cpe
Check for whitespace behind the statement, zapata.conf seems bitchy about
whitespace.
hth
-----Original Message-----
From: Steve Totaro
2005 Sep 28
1
Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unknown signalling method 'pri_net'
Any ideas?
51] logger.c: [chan_zap.so] => (Zapata Telephony)
Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found
Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata-auto.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing
2009 Feb 23
1
Compiling asterisk-addons-1.6.0 under Debian 2.6.18?
Is there some magic to compiling asterisk-addons-1.6.0 under Debian 2.6.18
and mysql 5.0? I am unable to get configure to recognize the existance of
mysqlclient. Imparticular, when it gets to:
checking for mysql_init in -lmysqlclient... it returns "no".
For the past several releases, I've had to hack or otherwise coerce this
to work, but this time, no amount of fiddling with
2004 May 23
1
Serious NAT problems: can't call between lines on sipura
I have a problem that is almost certainly nat-related, but I can't figure
out what's happening.
Since moving the Sipura behind a NAT server (Linksys), I am no longer able
to call between the two lines on the same Sipura. When I dial one
extension from the other, it rings, but immediately after I pick up the
ringing phone, the call is uncerimoniously dumped. I can tell the call
2004 Apr 12
4
Invalid module format in 2.6.5 after running make linux26
[root@asterisk zaptel]# modprobe ztdummy
WARNING: Error inserting zaptel
(/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format
WARNING: Error inserting zaptel
(/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format FATAL:
Error inserting ztdummy (/lib/modules/2.6.5-1.315/misc/ztdummy.ko):
Invalid module format FATAL: Error running install command for ztdummy
[root@asterisk
2004 May 07
1
cannot play sound files
Greetings, I have a new * system installed and everything works as it
should except for one annoying little problem: I can't play any sound
files. What this means is that when an extension script gets to the point
where it should play a sound file (voicemail greeting, auto-attendant,
whatever), the caller hears a click and then silence. According to the *
log, the sound file is being
2004 May 22
2
rejected NOTIFY requests
When I enable NOTIFY messages in my SIP device (Sipura), Asterisk reports:
handle_request: Unknown SIP command 'NOTIFY' from 'xxx.xxx.xxx.xxx'
When I disable NOTIFY messages, * reports the device UNREACHABLE, followed
by REACHABLE every couple of minutes.
I think I want NOTIFY on, because the Sipura is behind a NAT server, but
the constant stream of warnings from * make me think
2006 May 31
1
Connect 2 Asterisk Servers via PRI
I am trying to test out my PRI configuration on a server by conection
another asterisk server to it. I have the T-1 cards up and both set to
bchan=1-23 and dchan=24 in zaptel.conf I am getting lost on the signalling
in zapata.conf. using pri_cpe makes sense for my target box, but I am
getting the following errors on my source box if I use pri_net or pri_cpe
signalling.
May 31 17:15:06
2004 May 24
5
2 Sip phones behind un-natted Asterisk
I have 2 SIP phones (Cisco 7960 & XTen) behind a NAT provided by a
Linksys firewall that supports UPnP. The Asterisk server has a public
IP. Here are the problems that I am having with this configuration...
1. The 2 SIP phones can call MeetMe and have a conference but
cannot call each other. (Yes, they connect but no audio either
direction)
2. I have verify=yes in the sip.conf for both
2005 Sep 08
2
Distinctive ringing on Cisco 79xx
Greetings, I am trying to implement distinctive ringing on a Cisco 7960.
I have tried setting alert_info to chirp1 or chirp2 before dialing the
phone, but it has no affect. If you have successfully implemented
distinctive ringing on a 7960, I would really appreciate seeing the snipit
of code that works.
Thanks in advance
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775)
2004 Aug 08
6
Voicepulse problems?
Is any one else having problems with Voicepulse today? Suddenly, I can't
register and calls to my Voicepulse numbers get a fast busy.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
2006 Feb 07
3
No sound on 10% of incoming calls
Hello,
I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring
but I don't hear the caller and the caller doesn't hear me (all IP Phones
have the same problem).
This problem appear also if the call is directly send to the second E1 of
the digium card who is connected to an IVR.
It does not depand on the charge of the server (I have the problem with only
one call).
2007 Aug 01
3
TE120P in Canada
Hi All,
I'm having problems trying to get a TE120P operational in Canada.
I keep getting a congestion error when I try to make a call. I'm not
sure if my switching, parity, etc is correct. I'm hoping that someone
will be able to verify my config.
The Telco is SaskTel, with a 10 channel 50 DDI service.
Zap show channels show and ztcfg -vv looks ok and the zttool show
2004 Oct 04
3
echo cancellation: the never-ending quest for truth
Asterisk apparently has five echo cancellation algorithms: STEVE, STEVE2,
MARK, MARK2 and MARK3. The current default appears to be MARK2.
My question is, has anyone had any experience with any of the others
(other than MARK2), and is there some conventional wisdom as to when to
use one over another?
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
2004 Sep 24
1
help with skinny
Hi all,
I bought a couple phones for really cheap just for a simple solution. I'm
trying to get a few 7910 to work with *. I'm just not sure how to get them
to work. The 7910 just sits there "configuring IP" Here is a copy of my
skinny.conf. the extensions.conf is default. I just want to bring the
system up in default before a start making changes. Do I need to make
2004 Oct 06
2
Cisco router for PRI termination?
If you have a PRI terminated in a Cisco router talking SIP to * and would
be willing to share your Cisco config, please respond. Also, I would be
interested in knowing what version of IOS you are using. We are using an
NM-HDV in a 3640.
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
2005 Sep 06
5
PRI in and out
I am wanting to front-end a legacy PBX with an asterisk box. I have done plenty
of asterisk work over the last 6 months to PRI circuits, but not with a PBX
being involved.
I know I can use asterisk and digium cards in this manner, but do I need
separate cards for the PRI -> Asterisk side to the Asterisk -> PBX side, or will
a 4-port PRI card do the job? (I already have a spare one of
2004 Jun 25
2
forced ring on dial?
I am routing outgoing calls through a sip gateway. The calls go through
no problem, however the ringing in the callers ear begins as soon as the
last digit is dialed. This has two nasty side effects. First, the caller
hears 1-2 more rings than the callee. Second, and more importantly, if
the callee's line is busy, the caller continues to get hear ringing, even
though the gateway has
2004 Jun 04
2
Mystery PRI NOTICEs & WARNINGs
Since connecting a PRI to a Digium T100P, I have been seeing the
following messages in syslog every few minutes:
Jun 4 06:51:54 pbx asterisk[13435]: WARNING[1209214400]: chan_zap.c:6176 in zt_pri_error: PRI: Read on 56 failed: Unknown error 500
Jun 4 06:51:54 pbx asterisk[13435]: NOTICE[1209214400]: chan_zap.c:6913 in pri_dchannel: PRI got event: 8 on span 1
Sometimes, these messages come out