On Mon Nov 28 12:21:09 CST 2005 C F shmaltz at gmail.com wrote:
What does the TE406 leds indicate?
Both the ADIT 600 led and the TE406 led are green, the ADIT
has zeros in the error counters. Syslog has this as a final
message after running ztcfg:
Nov 28 02:31:08 xxx kernel: Registered tone
Looks like it's losing it's connection to the DNS server, make sure
you don't have any names that need to be resolved to IP address in any
of the config files for asterisk. Just use IP address.
There are other known ways of working around this problem (which I'm
sure others will mention), but for
Title: Accepting Inbound SIP Connections
Hello,
We have an * server setup and are trying to take inbound SIP calls from our provider. According to the asterisk log, our box sees the call come in, however, it never seems to route the call, rather gives a Congestion message and the calling
I heard of someone using the Zultys 4x5 phone for a
similar setup, as it has an FXO port built in
- Original Message -
From:
Joseph Rothstein
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 29, 2005 9:19
AM
Subject: [Asterisk-Users] small office
Thanks Michael - you got me on the right path. What you gave me didn't
work, but I figured out that the following does (on version 1.2):
exten = default,1,VoiceMailMain(${CALLERIDNUM})
(BTW, exten = Unknown,1,VoiceMailMain(${CALLERIDNUM}) used to work
for us in Asterisk 1.0.9 but obviously
Title: Accepting Inbound SIP Connections
Hi
Roger,
Can you please send a 'sip debug' output, so we can see the actual SIP trace of
the messages ?
Nir
S
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roger
JohnsenSent: Tuesday, November 29, 2005 12:30 AMTo:
Michael Welter wrote:
I'm getting the following messages when a call is answered by a SIP device:
Nov 28 13:03:01 NOTICE[22824]: rtp.c:1146 ast_rtp_raw_write: RTP
Transmission error to 192.168.1.254:19262: Operation not permitted
For a Cisco 7940 line, I have the following sip.conf entry:
I made an error in what I previously wrote. What actually works in v1.2 is:
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
Which is what Michael originally wrote. My bad!
Sascha wrote:
Thanks Michael - you got me on the right path. What you gave me didn't
work, but I figured out that the
Suggest checking your extensions.conf file under [from-sip-external].
By default it often has:
;give external sip users congestion and hangup
exten = _.,1,AbsoluteTimeout(15)
exten = _.,2,Congestion
exten = _.,3,Hangup
The above lines should be commented out and the following added:
Are there any example configs? Or does anybody have a default config
for this setup:
1 analog digium clone card for an analogue line (my home line)
Several sip phones (a few of them on the outside of my lan (NAT fw
between) and 2 insde my lan)
Or a simple way of configging through a
Hello Pinoy asteriskers,
This is also an annoucement of a new mailist list for Filipino Asterisk users.
please visit - http://groups.yahoo.com/group/asterisk-ph
Thanks,
Lito
___
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Asterisk-Users
On 11/28/05, Martin Joseph [EMAIL PROTECTED] wrote:
Hi,
I am a newbie, and I am setting up a simple system to share a PSTN
line with another location.
In the process of setting this up I am also testing the various codecs.
I am only able to get comedian voicemail (ie dialing 1234) to
I'm having a problem with Asterisk sending too many INVITEs to a peer for a
single call. I can't quite figure out why there are these rapid INVITEs sent
to the call proxy. The call completes correctly (to, in this example, an echo
test found via ENUM) but the number of INVITEs is really out
John Todd wrote:
2) The intervals between the INIVTEs after the 407 sequence are: 34ms, 30ms,
49ms, 91ms. This is _way_ too fast for response timers to be expiring for
reliable re-transmissions of INVITEs... isn't it? According to DEFAULT_RETRANS
in chan_sip.c, the proper delay should be
Hello,I have searched the site extensively and this is the only reference I find to my problem-the problem, but no answer.T Aksoy asterisk-users@lists.digium.com
Tue, 18 Feb 2003 23:36:39 -Hi,I have an X100P fxo card configured so that a user will ring on either line, and the call will
Michael Welter wrote:
Michael Welter wrote:
I'm getting the following messages when a call is answered by a SIP
device:
Nov 28 13:03:01 NOTICE[22824]: rtp.c:1146 ast_rtp_raw_write: RTP
Transmission error to 192.168.1.254:19262: Operation not permitted
For a Cisco 7940 line, I have the
Josheph:
I had have that problem, and it get solve when i take out the incominglimit from my sip.cfg
Also if you send you sip.cfg and extensions.cfg will be easier to help you
Tray it.
Alvaro Parres
On 11/28/05, BJ Weschke [EMAIL PROTECTED] wrote:
On 11/28/05, Kevin Hanson [EMAIL PROTECTED]
Hi list...
I have been testing the hint extension. And i detect
that when i have in the sip.cfg of the extension the
incominiglimit=X (any number) the hint doesn't work all the
time show the extesion as idle.
If this is a bug or not ??
Thanks.
All the channels are being seen (12total) before I run asterisk, when
running asterisk the channels are not being recognized. I am running 3
wildcards, and al which are installed correctly and are seen by linux,
and zaptel. What needs to be done to get these cards seen in Asterisk.
I followed the
I have that problem also when trying to use perl agi scripts.. although
not with php scripts. What would the output be if problem with perl?
Benoît Mérouze wrote:
Hello,
I've noticed my AGI, in Perl, was always returning 0 even if exit from
it with something else than 0.
On
Alvaro Parres wrote:
Hi list...
I have been testing the hint extension. And i detect
that when i have in the sip.cfg of the extension the
incominiglimit=X (any number) the hint doesn't work all the
time show the extesion as idle.
If this is a bug or not ??
Thanks.
Hello All,
I'm using an Avaya 4620SW with Asterisk, the phone when hooked up to
the network, works for sometime, (I have not actually monitored the
time) maybe 20-30 minutes, after which the phone will still have a dial
tone, but can't dial out or recieve calls. I scanned thru the logs and
found
Hi
i renew my question ;=)
i have 8 phone number provided by my VoIP supplier :
081037XX0
081037XX1
081037XX2
...
For each, i have a login/password
where in put the registrer into my config ?
it's a Trunk on incoming no ?
i have put one register= per number
Hi out there,
Im looking for support in regards to configuration of ZAP interface on a
TDM400P card (TDM04D).Im running Asterisk v 1.0.7 cvs on White Box 3.5 with
kernael 2.4.21
Ive got two problems with Asterisk:
1. When there is an incoming phone call on the zap interface, Asterisk holds
Hi Ren,
The optiPoint 400 does not support the multi-line function and hint.
The newer models i.e. the optiPoint 420 should support this features.
Ren Enskat [Teamware GmbH] wrote:
Hi stephen,
I have the latest SIPfirmware
form the siemens site!
I have an OptiPoint 400
On Mon, Nov 28, 2005 at 07:59:41PM -0500, Scott Geist wrote:
All the channels are being seen (12total) before I run asterisk, when
running asterisk the channels are not being recognized. I am running 3
wildcards, and al which are installed correctly and are seen by linux, and
zaptel. What
Pretty sure
zapata.conf
immediate=yes
should do it for incomming calls and you won't have CID.
Brett
On 11/29/2005, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi out there,
Im looking for support in regards to configuration of ZAP interface on a
TDM400P card (TDM04D).Im running
Hi John,
I know a good one and it's from Siemens. The new optiPoint 410 or 420
series are great phones with good quality speech and features. Contact
me off list for pricing etc..
With regards,
Stephen
John Fraser wrote:
Hi all,
Does anybody have any info on a decent quality sip hard
Im looking for support in regards to configuration of ZAP interface on a
TDM400P card (TDM04D).Im running Asterisk v 1.0.7 cvs on White Box 3.5 with
kernael 2.4.21
Ive got two problems with Asterisk:
1. When there is an incoming phone call on the zap interface, Asterisk holds
call for
Polycom make some surprisingly good and reasonably priced SIP handsets
too. Many of these support headsets and we've been quite impressed by
them.
FFF Managed Technology Ltd.
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz
I'm trying to implement some of the star services such as *61 for
weather or *71 for wakeup call, etc. I think I have asterisk setup
properly because I can get them to work fine using normal extension
numbers. However, if I try to use the 'star' numbers, my Polycom IP500
never sends the digits
Noc Phibee wrote:
Hi
i renew my question ;=)
i have 8 phone number provided by my VoIP supplier :
081037XX0
081037XX1
081037XX2
...
For each, i have a login/password
where in put the registrer into my config ?
it's a Trunk on incoming no ?
i
I am running the newest version of Asterisk, on Slackware10.2, and the
latest kernal, I believe it is 2.6. I am using 3 wildcards, the 400
series ones. THe zaptel.conf file is configured right, and the output
for the cat/proc/zaptel shows all 12 channels configured. As for zap
show channels, all I
The idea is that any number inside the [] is one checked for i.e.:
_123[456]78
will match:
123478
123578
123678
--
Cheers,
Matt Riddell
___
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http://freevoip.gedameurope.com (Free
Scott Geist wrote:
I am running the newest version of Asterisk, on Slackware10.2, and the
latest kernal, I believe it is 2.6. I am using 3 wildcards, the 400
series ones. THe zaptel.conf file is configured right, and the output
for the cat/proc/zaptel shows all 12 channels configured. As for
On Fri, 2005-11-25 at 13:08 -0500, Gary MacKay wrote:
After playing around with the CALLERID(number) and
CALLERID(name) variables and things, I find that asterisk is sending
the name to my phone and the name is unknown. I added a line
exten = _X.,Set(CALLERID(name)=${CALLERIDNUM}) and now it
On 08:48, Tue 29 Nov 05, [EMAIL PROTECTED] wrote:
From memory (at a previous installation) you will need a newer version
of
Asterisk than 1.09 for the lights to work.
on 1.0.9 the lights work.
In this way:
person is on the phone: light is on
Person is not on the phone: light is off
On Sat, 2005-11-26 at 10:43 -1000, Jean-Denis Girard wrote:
Hi list,
I installed iaxmodem and Hylafax to see how it compares to rx/txfax; so
far I had 0 failure in my limited testing with a Philips HFC21 fax
machine that failed very often with txfax (same test platform, with
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