Displaying 20 results from an estimated 300 matches similar to: "visp multiaccount + firewall configuration problem"
2003 Nov 13
2
IAX trunk monitoring
I have an issue where * tries to route a call over IAX to another server
even if the server is down. I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output. If someone could
tell me what I have configured incorrectly, I would appreciate it.
Thanks,
Stephen
-----------iax.conf on voip2----------
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip
2008 Apr 02
1
show uptime and last reload
Hi,
I just upgraded from 1.2 to 1.4.
In 1.2, when I did a "show uptime" I used to see a
second line telling me the time since the last reload.
Has this been removed in 1.4?
The following is the output of my two test boxes:
Connected to Asterisk 1.4.18.1 currently running on
voip2 (pid = 10605)
Verbosity is at least 3
voip2*CLI> show uptime
System uptime: 15 hours, 55 seconds
2003 Dec 08
3
IAX error messages in log
I constantly get the following error messages in
/var/log/asterisk/messages:
Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324
(iax_ack_registry): Received unsolicited registry ack from '192.168.0.1'
Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181
(socket_read): Registration failure
Where 192.168.0.1 is another asterisk server. Below are the local and
2009 May 14
2
Problem with Asterisk + TDM410 FXO
Hi
I am in the middle of move a small business over from legacy PABX + PSTN
lines to VOIP infrastructure.
I borrowed a spa9000 to place between the PABX and the PSTN lines. I
have had this going for a while (>5 months) and it has been working fine
(some issues with echo and other minor things), which is why I am moving
to asterisk.
I bought a tdm410 with 3 fxo + fxs. The fxs is connected to
2005 May 10
1
Limiting outbound calls
My VoIP provider allows me to have more than one call outbound on the
same line simultaneously, for some reason. I am pretty sure that they
do not want this to happen, so I'd like instead to limit each line to
one call.
I do not want the users to have to dial another prefix to go out on
another line. Is there any way to add multiple accounts for my _9.
extension and have Asterisk
2006 Dec 13
0
Help with voicemail
I'm looking to use * for a HQ/branch office topology with fairly few calls
over the WAN. The questions I have all pertain to the following
architectural pic: http://www.45891.com/misc/arch.jpg
I'm looking at a distributed architecture so users are somewhat functional
when the link to HQ is down, with a centralized voicemail server to allow
for transfer of voicemail messages from user to
2006 Jan 20
5
Asterisk in SPA9000?
Did Linksys really use Asterisk for the SPA9000 software?
--
Andres
Technical Support
http://www.telesip.net
2005 Oct 05
0
Unwieldy outbound macro
I have the following pair of macros defined to handle outbound calls from *.
Rather than specifying full dialstrings in the main body of extensions.conf,
outbound dial commands are made using a macro call as follows:
Macro
(outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate
way4)
The final gateway defined is nearly always a fallback to PSTN if none of the
IAX or SIP
2006 Feb 25
1
Asterisk as a dedicated Analog PSTN gateway
Hi there,
I was wondering if anyone has successfully used Asterisk as a dedicated
Analog PSTN gateway to take the place of, for example, a Mediatrix 1204 or
an 8 port model?
Basically, I am thinking of using a Linksys SPA9000 as the PBX and just need
an Analog PSTN gateway for 4 to 8 FXO lines. It does not sound like the
Mediatrix 1204 does a very good job and I figure I can build a much more
2004 Apr 26
0
Help with connecting 2 servers via iax
I have successfully configured two servers and I am now trying to connect
via iax. When I attempt to call from one ext, 2006(server viop1) to
extension 3006 (server voip2) I receive a timeout or "call failed 403
forbidden.
The information I am receiving from the console is below.
Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type
registered for 'IAX'
2011 Jun 29
1
No audio format found to offer.
This *should* be something that's easy to fix, but apparently I'm not
doing something right.
Our SIP long distance provider is telling us to only use formats G.723
and G.729, so I've set up their trunk configuration in sip.conf as such:
[t564]
type=friend
host=XXX.XX.56.4
context=default
disallow=all
allow=g723
allow=g729
However, the Dial application gives the following error:
2006 Oct 27
1
Iax bug ?
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include "iax.voip1.conf"
#include "iax.renoir.conf"
The iax.voip1.conf file contains :
[VOIP1]
type=friend
2011 Mar 09
6
SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten => s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
/INVITE sip:0473 at sip.domain.be SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97
From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2
To: <sip:0473 at sip.domain.be>
2005 Aug 22
0
SPA3000 dial plan?
Hey, all... If this is too off-topic, I'd be grateful for directions to a
more appropriate mailing list.
I'm trying to set up Asterisk and some Sipura boxes. I've got an SPA-3000
which is registering twice with Asterisk - once for its FXS/Line1/VoIP1
and once for its FXO/PSTN/VoIP2.
My eventual goal is to have inbound calls on its FXO ring four times on
its FXS and then fail over to
2007 Feb 12
4
Zaptel install...
I am having trouble getting Asterisk to compile the zaptel stuff.
Here are the specifics:
Linux Kernel 2.5.9-42.0.8.EL
Asterisk 1.4.0
I compiled libpri, zaptel, asterisk and asterisk-addons (in that
order). This is a fresh install of CentOS. Following the CentOS
install, I did "yum -y update" until there were no updates left.
Here is my src directory:
drwxr-xr-x 24 root root
2004 Jul 15
1
zapras - and kernel ??
Hi,
I'm trying to get zapras do work, I had downloaded the pppd-source and the 2
patches.
I succefull compiled and install the patched version of pppd, but got this
error in message-log
Jul 15 11:43:32 voip1 pppd[9296]: In file /etc/ppp/filters: unrecognized
option 'active-filter'
Jul 15 11:43:57 voip1 pppd[9299]: Plugin zaptel.so loaded.
Jul 15 11:43:57 voip1 pppd[9299]: Zaptel
2016 Jan 26
2
Samba Hylafax PAM
O, try the following.
Test this first.
ldd /usr/sbin/hfaxd
if you getting libpam.so.. something, then hylafax is compiled with pam support.
Next,
apt-get install libpam-ldap ( just to be sure, i do believe you have installed it already )
create the file :
/etc/pam.d/hylafax
Add :
auth required pam_ldap.so
account required pam_ldap.so
2004 May 20
4
Mystery SIP channels
Has anyone seen this before? This channel is consistently present on
both of my asterisk servers. Sometimes they disappear for a few seconds
and then come back. It always has the same Call ID.
voip1*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter
Format
192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms
UNKN
2007 Nov 28
1
Polycom MWI's will not turn off
Hello,
I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that
the MWI indicators will never go off (The blinking red light and envelope in
the LCD).
I have tried to upgrade to 1.4.14 and all different SIP versions on the
Polycoms. I am now at 1.6.7
Here is the SIP Message that turns on the lights:
Scheduling destruction of SIP dialog '
2005 Jan 04
2
Asterisk stops - why ?
Hi,
Sometimes my asterisk server stops. (after a day or two)
Last output from CLI is:
--------------------------------
-- Registered SIP '000b82017eb7' at 213.237.12.125 port 11620 expires 120
-- Channel 0/26, span 1 got hangup
-- Hungup 'Zap/26-1'
voip1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk cleanly ending (0).