After so many rings when the party does not answer, my SIP phone says
Call Failed. Why doesn't it just keep ringing?
Here's the dial plan rule:
exten = _NX,1,Dial(SIP/[EMAIL PROTECTED],,r)
exten = _NX,n,Hangup()
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Is there any plan to include the zaptel drivers into the main Linux
kernel? If not, there should be one.
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Please explain the relationship between modules from the driver
(wctdm), the /etc/zaptel.conf file and zapata.conf. Specifically, if
I have a FXS module 0 and FXO module 1, what should be used in
zaptel.conf and what should be used in zapata.conf? Then finally, in
extensions.conf, what
On Apr 11, 2007, at 2:38 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
Try to update your zaptel to latest 1.4 svn. I just fixed a bug in a
patch that was committed not too long ago. It should fix it.
Thanks. I will try that. I did start using the 1.4.2 tar release to
get things
Using the latest SVN of zaptel and asterisk, I can no longer receive
incoming analog calls. The caller just hears it ringing but Asterisk
doesn't pick up.
I am seeing these error messages:
[Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID
returned with error on
Neglected to mention the host operating system:
Linux myhost 2.6.20-1.2307.fc5 #1 Sun Mar 18 20:44:48 EDT 2007 i686
i686 i386 GNU/Linux
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On Apr 9, 2007, at 1:51 PM, Kevin P. Fleming [EMAIL PROTECTED]
wrote:
You also neglected to mention the version of Asterisk you are running;
'latest SVN' means nothing when there are 20+ branches of Asterisk on
our SVN server.
Sorry about that. It is the 1.4 trunk:
Asterisk
This compile error started happening about 2 weeks ago with zaptel.
/mysrc/asterisk/zaptel-1.4/xpp/xbus-core.c: In function ‘debugfs_open’:
/mysrc/asterisk/zaptel-1.4/xpp/xbus-core.c:171: error: ‘struct inode’
has no member named ‘u’
make[4]: *** [/mysrc/asterisk/zaptel-1.4/xpp/xbus-core.o]
Is there a zaptel mailing list?
Here's the error:
CC [M] zaptel-1.4/xpp/xbus-core.o
zaptel-1.4/xpp/xbus-core.c: In function ‘debugfs_open’:
zaptel-1.4/xpp/xbus-core.c:171: error: ‘struct inode’ has no member
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Ever since upgrading to 1.4 SVN, the advanced options on voicemail
have disappeared. When I press 3 for advanced options, it just
reviews the message. It used to present me with a menu to 1 = reply,
2 = call the person back, 3 = play message envelope. What gives?
If you own Aastra phones, here's a group dedicated to your specific
needs. BTW - The Asterisk-users mailing list is great but it has way
too much volume to be useful for specific problems. It needs to be
broken up into smaller more manageable lists.
Homepage:
I cannot access my voicemail and get the following warning in my
console:
[Nov 25 10:26:43] WARNING[5628]: app.c:935 ast_lock_path: Failed to
lock path '/var/spool/asterisk/voicemail/default/8900/Old': File exists
I have also noticed that Asterisk will crash several minutes later
after
I retested this with 1.4.0-beta3 and I still can't access my
voicemail. I dial the voicemail extension and I just get silence for
a few seconds and it hangs up. HELP! I have 295 messages in my old
mailbox and I want to retrieve my new messages.
There was a stale lock file in the mailbox directory. This is a bug
though. Asterisk should clean up all lock files on startup. Lastly,
I can't explain the intermittent crash and wasn't able to catch it
using gdb either.
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It seems like asterisk-addons in SVN has been broken for the last few
weeks:
gcc -DHAVE_CONFIG_H -I. -I. -I. -I./ooh323c/src -I./ooh323c/src/h323 -
DGNU -D_GNU_SOURCE -D_REENTRANT -D_COMPACT -c src/chan_h323.c -MT
chan_h323.lo -MD -MP -MF .deps/chan_h323.TPlo -fPIC -DPIC -o .libs/
I would like SayDigits to say a phone number faster. Is there a way
to control the speed?
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When you listen to old messages, it would be better if Asterisk
reversed the order so that it starts at the most recent message and
then forwarding goes to the next oldest message, etc... The last
message would be the oldest. This makes more sense for old messages.
Also, is there a way
On Oct 19, 2006, at 3:00 PM, [EMAIL PROTECTED] wrote:Date: Thu, 19 Oct 2006 09:30:38 -0500 From: "Eric \"ManxPower\" Wieling" [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk hangs up on incoming analog calls after a while To: Asterisk Users Mailing List - Non-Commercial Discussion
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Robert La Ferla wrote:
I have been experiencing a problem where after someone calls me
from an
analog line, the phone call is terminated after a period of time
(anywhere from 15 seconds to 15 minutes
I have been experiencing a problem where after someone calls me from
an analog line, the phone call is terminated after a period of time
(anywhere from 15 seconds to 15 minutes) The phone that I use to
answer the call is an Aastra 9133i SIP phone. There are several
other SIP extensions
I found an init.d script for asterisk BUT not for asterisk/zaptel
modules. I'm still looking for a good solution. It seems to me that
the correct solution would involve /etc/modprobe.d/modpobe.conf.
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If I boot my server and manually type modprobe wctdm, it correctly
loads both wctdm and zaptel. If I put the modprobe in /etc/rc.local
and reboot, it fails. Why? I am running the latest svn source of
zaptel on Fedora Core 5 (w/latest updates as of 8/15)
Here are the error messages from
Can someone send me their modprobe.conf file? I think that may be
the problem. A zaptel file is created during install in /etc/
modprobe.d but modprobe.conf must need to reference it...
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When I dial out, I can't hear any ringing. I am using the latest SVN
code (SVN-branch-1.2-r37458M). Is this a problem with Asterisk? Or
with my VOIP provider?
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Can someone send me a link to a GSM sound file (US-English) for the
words to and the?
BTW - These should be put in the standard asterisk-sounds
distribution. I couldn't find them in mine or in the SVN repository.
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I want to create an extension say 8000 that prompts the user to
enter a number and then dial that entered number according to a set
of rules. The rules for dialing out are in different context (dial-
out-rules).
[mymenu]
exten = 8000,1,Answer()
[dial-out-rules]
; toll-free numbers
build_tools/mkdep -fPIC -fPIC app_addon_sql_mysql.c app_saycountpl.c
cdr_addon_mysql.c res_config_mysql.c
app_addon_sql_mysql.c:15:22: error: asterisk.h: No such file or
directory
app_saycountpl.c:10:22: error: asterisk.h: No such file or directory
cdr_addon_mysql.c:22:22: error: asterisk.h:
I have encountered the following problem with the latest Asterisk source
(as of 4/23/2006):
Someone calls me on my PSTN line, it then dials my analog extension (I
have both SIP and analog phones where all analog phones are a shared
extension.) After a while, I get a busy signal. How can I
The volume/traffic on this list has been getting pretty heavy. I find
it hard to follow certain discussions and there are some that I am not
interested in. Perhaps, we could split the list into two: One for
discussing hardware (client phones and cards) and one for the software
I am looking for docs on how to diagnose and adjust the rx/tx gain in
zapata.conf. The wiki has a link to this article but it no longer
exists on the server.
http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html
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Steven [EMAIL PROTECTED] wrote:
Hi I have installed Festival on the same box as asterisk and followed the
instructions to integrate it with asterisk.
Festival seems to work fine on its own performing text to speech from the
command line or via a file.
Asterisk answers the call but there is no
If you have many old voicemail messages, to get to the most recent one,
you have to keep hitting 6 until you reach the last one. It would be
better if you could hit 4 from the first message to get to the last
message and/or have a digit that takes you the first and last messages
respectively.
I made a small change to apps/app_voicemail.c to permit circular
navigation when listening to messages. If you are at the first message,
and press 4, it takes you to the last message. If you are already at
the last message and press 6, it takes you to the first message. I
did a quick test
I have been trying to adjust the gain as per this document without any
success:
http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html
I have a PSTN and VoIP (SIP) connection via *. I disabled all echo
cancel/training in zapata.conf and set tx/rxgain to 0. I then changed
Anyone using the AlarmReceiver? Does it work? Mine doesn't seem to
communicate properly. How can I tweak the DTMF settings? Is it in the
zaptel.conf or somewhere else??
-- Executing AlarmReceiver(Zap/1-1, ) in new stack
AlarmReceiver: Setting read and write formats to ULAW
Kevin Bockman wrote:
If you are using 1.2, I would use native (codec, not MP3). There
should be an example in the sample config file in
/usr/src/asterisk/configs/musiconhold.conf.sample - I don't see it on
the Wiki. It should be there, somewhere. Must be buried. For this
option, you will
Chris Albertson wrote:
Second even if there were one the mpg123 process is not long lived. A new one
is started for each
MOH session.
I hate to say it but there is a problem where the mpg123 process never
terminates. This occurs with the latest SVN-branch-1.2-r7917 version
and has been
Douglas Garstang wrote:
I'm curious why the number of jobs out there requiring Asterisk seems to be
pretty low. After looking around dice, monster, careerbuilder etc, I was
surprised to find no more than 3-4 employment opportunities with Asterisk
throughout the US.
Is it really that low?
Jamie J. Begin wrote:
I've been pulling out my hair all day on this one. If anyone can help,
I'd really appreciate it. :-(
I've got an Aastra 9133i (with the latest firmware version) and a Cisco
7960 sitting behind a NAT device on my LAN. The Asterisk server is
hosted offsite and has a public
Is there some way * can trim the trailing silence in a voicemail
message? There's the maxsilence setting for silence detection which
is related to what I'm asking but not the same. Let's say I set the
maxsilence to 8 seconds. During the recording of a voicemail, if
someone doesn't say
Brett, Gary wrote:
My question is which OS would be preferred in this configuration Fedora Core
1 or Fedora Core 3, and are there any install guides out there that are
recent enough for asterisk 1.2
Try Fedora Core 4 (FC4). Works great.
___
Did you try running * under gdb? When it crashes, do a bt to get a
back trace and post it to the mailing list.
e.g.
% gdb /usr/sbin/asterisk
GNU gdb Red Hat Linux (6.3.0.0-1.84rh)
Copyright 2004 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License,
Michael Stearne wrote:
I am having trouble with FC3.
After doing a yum update (of 1264 packages) I still cannont compile
1.2.1 from source:
make[1]: `libedit.a' is up to date.
make[1]: Leaving directory `/usr/src/asterisk-1.2.1/editline'
make[1]: Entering directory
When I dial a festival extension (1222), all I hear is a series of fast
clicks and then it hangups. I do not have a sound card installed but I
would think I don't need one. Is a sound card necessary? Should I use
a script instead of the scheme code? Can someone who has this working
send me
Let me add that text2wave works fine. Something is wrong with the
Asterisk = Festival server communications. Ideas?
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You must have seen this page but maybe not because the site was down for
a while recently:
http://turnkey-solution.com/asterisk-sphinx.html
I am also interested in getting Sphinx to work with Asterisk. Please
report anything you find. I know that there are a few different
versions of
What does this warning mean?
WARNING[11065]: chan_sip.c:9596 handle_response_register: Got 200 OK on
REGISTER that isn't a register
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Moises Silva wrote:
Hello Ryan. Check out the file /etc/modules.conf,
/etc/modules.d/zaptel ... if for some reason you have empty the
modules.conf, modules-update force will fix it, tough. In order to
provide you with further help, please provide more clues.
What about systems that use
Alexander Lopez wrote:
I vote for 'a' as the auto-play option.
http://bugs.digium.com/view.php?id=6090
In thinking about this more, the auto-play option can be a quickie fix
but a more complete implementation is needed. Think about the scenario
when checking your voicemail from an
[EMAIL PROTECTED] wrote:
The seller refuses to tell me who the vendor is.
That should send up the big red flag to not buy anything from that seller.
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Tomislav Parcina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
I want to create an extension that goes directly to my new messages
without having to press 1. How do I do that? I can call
VoiceMailMain but then I have to choose 1 from the menu. I'd like it
to go there
Why would you use this? Can someone please elaborate on the below
description? I'm missing the intent of it.
localhost*CLI show application Page
[Synopsis]
Pages phones
[Description]
Page(Technology/ResourceTechnology2/Resource2[|options])
Places outbound calls to the given technology /
So I can set it up to call a bunch of extensions and broadcast a message
to them without the user picking up? Can I do this with Aastra phones?
This would be great for announcing incoming calls. You have a call
from XXX . Press 1 to pickup Press 2 to send them to voicemail.
Is it possible to dial with a silent ring? If so, is it configurable
with * or does the phone have to support it?
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Alexander Lopez wrote:
I vote for 'a' as the auto-play option.
http://bugs.digium.com/view.php?id=6090
I second the vote. I thought of using the same letter after reading
your reply.
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Andrew Latham wrote:
I think most all of the phones have a hack to get it working. Aastra
analog ADSI phones even work as I read some where...
It works with the 9133i. This is a great feature!
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William M. Sandiford wrote:
I just upgraded my system to the latest svn-trunk
I previously made extensive use of the SetAccount() function, but now
I'm getting the following error
Dec 29 20:54:08 WARNING[4925]: pbx.c:1679 pbx_extension_helper: No
application 'SetAccount' for extension
I want to create an extension that goes directly to my new messages
without having to press 1. How do I do that? I can call
VoiceMailMain but then I have to choose 1 from the menu. I'd like it
to go there and play the first message or say There are no new
messages and hangup. How can I do
-- Executing VoiceMail(SIP/999-e59b, 500|g4) in new stack
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1212703824 (LWP 4440)]
0x003d0110 in rawmemchr () from /lib/libc.so.6
(gdb) bt
#0 0x003d0110 in rawmemchr () from /lib/libc.so.6
#1 0x003c582b in
Jacques Leisy wrote:
Thanks Robert. I tried of course with time server disabled: 0 too.
Is it working for you? Which time server are you using, an external one?
Works for me and I'm using an internal one which is then synced to an
external one.
Try ONLY these entries. Remove the time
Franz Wu wrote:
I have one TE410P and want to know how to. Sending back to Digium
should be a good idea.
Is it possible to upgrade the firmware for a TDM400P? If so, where do
you download new versions and what's the upgrade procedure?
___
Can someone please send me your iptables rules for forwarding SIP/RTP
udp to your * server?
I tried this but I think I need more rules like DNAT or something...
iptables -A FORWARD -i $EXT_IF -o $INT_IF -p udp -m udp --sport 5060 -d
$ASTERISK_IP --dport 5060 -j ACCEPT
iptables -A FORWARD -i
The solution lies in sip.conf and extensions.conf. BroadVoice's
instructions are incomplete. You need to put your 10 digit phone number
as the extension in the register command in sip.conf and add entries
to extension.conf for your 10-digit extension under [from-broadvoice].
Neil wrote:
The problem appears to be with your settings. I have an identical
configuration with my * box running behind a NAT firewall with the same
firewall ports open.
I have experienced the same problem before. If port forwarding is switched
on then do NOT use the nat=yes and
I can't get festival to output any sound. Any ideas?
I have festival 1.95 installed on Fedora Core 4.
% rpm -qa | grep fest
festival-1.95-3
% cat festival.conf
[general]
host=localhost
port=1314
usecache=yes
cachedir=/var/lib/asterisk/festivalcache/
festivalcommand=(tts_textasterisk %s
Jacques Leisy wrote:
Since the release 1.3 the 480i displays the wrong date and time.
Something in 1947 !
I have followed the settings in the aastra.cfg.
time server disabled: 1
time server1: 192.168.0.10
time server2: 192.168.0.11
# time server3: 128.121.51.132
time format: 1
date format: 0
When someone calls me via BroadVoice, they get a busy signal. My * box
is behind a NAT firewall. I have enabled port forwarding of UDP 5060
and 1:2 to the * box. I added nat=yes externalip and localnet
to the sip.conf under [general]. It still doesn't work. I just want *
to be
trixter aka Bret McDanel wrote:
On Sat, 2005-12-24 at 20:17 -0500, Robert La Ferla wrote:
When someone calls me via BroadVoice, they get a busy signal. My * box
is behind a NAT firewall. I have enabled port forwarding of UDP 5060
and 1:2 to the * box. I added nat=yes externalip
Taco Scargo wrote:
Hello,
Just bought two 480i's which I updated to firmware 1.3
I experience the 'Far-End sound level issue' now.
I tried configuring the handset tx gain: value but can only make it
sound softer, not louder.
If there is someone that has managed to get decent Far-end sound
Looking for a low-cost but good quality (i.e. value) service provider
for US to Japan calls. Either a low monthly unlimited rate under $30 or
very low per minute rates. I'm using SIP and analog phones w/Asterisk.
The called party in Japan probably has PSTN phones with few exceptions.
How do you configure aastra.cfg to download directory list entries to
each phone? The Aastra documentation is very sketchy. Anyone have an
example???
You can use the Aastra Web UI (Operation-Directory) or the
configuration files
(aastra.cfg and mac.cfg) to download the Directory List.
You
Thanks. Can anyone explain what the three values for the ring pattern
signify? I assume it's a ring cadence pattern (in ms) but shouldn't it
be 4 values (ring on, ring off, ring on, ring off) So is Asterisk
ignoring the last ring off? And does Asterisk have some tolerance
value for the
Does anyone have distinctive ring working with Asterisk? Could you
share your zapata.conf and relevent extensions.conf?
Thanks.
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The audio volume of voicemail messages (msgNNN.wav) is rather low. Is
there a parameter/option to adjust gain?
In my voicemail.conf, I use these formats:
format=wav49|gsm|wav
Maybe I should use a different format?
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I am trying to configure zapata.conf to handle distinctive ring.
Everytime someone calls my main number, I get a ring pattern of 0,0,0
which works consistently. The problem is that every time someone calls
one of the other phone numbers (same number each time), I get a
different ring pattern
Anyone have an indications.conf entry for Japan?
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Carlos Chavez wrote:
On Fri, 2005-12-16 at 12:48 -0800, Dave wrote:
I had a lot of issues with 480i too and this is how I
resolved it:
1) Make sure that the file on the tftp server is
called firmware followed by the type they suggest (I
do not remember the type name)
2) Once this is done,
Rich Adamson wrote:
The traditional pbx vendors (back then) would always use the same words
that Kevin used, emphasizing the differences between key systems and
pbx's. However, many of the pbx manufacturers finally realized they
were loosing revenue due to those limitations, and began
Is there a way for another extension to join a call in progress? i.e.
If I can't share the line with all extensions, it would be nice to have
a single button (dial sequence) that allows any extension to join the
call. How can this be configured?
I'd like to configure Asterisk so that incoming calls from one POTS line
are shared amongst multiple extensions. i.e. If one SIP phone answers
the call, another SIP extension phone can pick up
and join the conversation. How do I configure this in extensions.conf?
Let me revise this a little:
I'd like to configure Asterisk so an incoming call from one POTS line is
shared amongst multiple extensions - both SIP and analog. i.e. If one
SIP phone answers the call, another SIP or analog extension phone can
pick up and join the conversation. How do I
Sean Cook wrote:
also you can ring multiple extensions:
Dial(SIP/101SIP/102SIP/103)
I have that but once one extension picks up, others can't join in.
Well, at least when I tried it with mixed SIP and Zap, it didn't work.
Maybe all SIP does but I need it to work for all phones SIP and
Kevin P. Fleming wrote:
Robert La Ferla wrote:
I'd like to configure Asterisk so an incoming call from one POTS line
is shared amongst multiple extensions - both SIP and analog. i.e.
If one SIP phone answers the call, another SIP or analog extension
phone can pick up and join
Kevin P. Fleming wrote:
Robert La Ferla wrote:
phones. When someone picks up (don't know how I can detect this), it
could transfer both parties to a meetme room. When additional
extensions pickup, they go to the meetme room. When everyone hangs
up, the call ends. Can this be done
Doug Lytle wrote:
I agree with Eric on this one. On my Polycom IP501s, I had to change
the digit map to allow for # and * matching. For testing, remove the
# and try again.
Remove it from the phone's dial plan or all together? Also, my phone
has a local dial plan that is set to this:
Is it possible to group all analog (regular phone) extensions so that
you can dial it from a SIP extension? i.e. for use as an intercom
I tried this:
[default]
exten = #3001,1,Dial(Zap/1,25,t,r)
exten = #3001,2,Hangup
but I just get a Call Failed and busy signal. I would think this is
Doug Lytle wrote:
Is it possible to group all analog (regular phone) extensions so
that you can dial it from a SIP extension? i.e. for use as an intercom
I tried this:
[default]
exten = #3001,1,Dial(Zap/1,25,t,r)
exten = #3001,2,Hangup
Change your dial to:
exten =
Eric ManxPower Wieling wrote:
The phone's built in dialplan is prolly blocking the call. Check the
docs for your SIP device. Remember SIP devices collect all digits,
then pass them on to Asterisk as one packet.
Also what Zap port is your analog phone connected to? What card are
you using?
How do I set up extensions.conf to wait for x rings (ringing all
extensions) before answering? I'm trying to mimic a regular answering
machine on an multiple analog phone system. Currently, Asterisk picks
up after 1 ring and just tries to dial one extension. I want all
extensions to ring.
Derek Whitten wrote:
[incoming]
exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()
Thanks. This will call/ring multiple extensions but what about waiting
for X rings before going to voicemail? How do I do that?
, microseconds or seconds?
Dave Cotton wrote:
On Fri, 2005-12-09 at 11:41 -0500, Robert La Ferla wrote:
Derek Whitten wrote:
[incoming]
exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()
Thanks. This will call/ring multiple
Dakota wrote:
Are there any cool free software I can use to create automated voice
message greetings for my PBX?
Take a look at Festival/Festvox. I'm not sure about the output format
but you could use sox to convert to gsm.
http://festvox.org
___
snacktime wrote:
The festival tts engine is free, but the the quality leaves a lot to
be desired. Definitly not something you would use in a business.
I'd say it depends on the use. Try it for yourself and see. Be sure to
try different voices because some sound better than others.
[EMAIL PROTECTED] wrote:
Thanks, I did that with upper and lower case, using 1.3. I have another
issue then because it is still not loading, it appears the phone is
loading but when I check the configs aren't there.
I looked at this last night. You need to have an aastra.cfg file in
your
I have one SIP phone (and soon a 2nd phone) and a Digium TDM11B (1 FXO +
1 FXS) card. I would like to be able to dial out the analog line via
Asterisk. How do I configure that?
i.e I'd like any extension to be able to dial 411, 911, 0, (617)
555-1212, 16175551212, etc... and have these
Lists wrote:
According to the wiki page
http://www.voip-info.org/tiki-index.php?page=Aastra+480i+Configuration it
shows lowercase file name and then there is a comment at the bottom that it
needs to be capitalized.
I have tried it both ways with no luck. Could someone comment on which way
the
Let me simplify my problem. I have a single Aastra 9133i SIP phone and
latest Asterisk from SVN source running on Fedora Core 4. The phone
currently says No Service I would like to be able to dial 1234 from
the phone and get Asterisk to play back an audio message or say some
digits. I
One more thing. I upgraded the firmware of the 9133i to 1.3.
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Pete Barnwell wrote:
I wasted a lot of time getting 9112is to work with almost identical
setup. The problem I eventually found was that the 9112is look for the
config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas
the documentation says they look for lower case, so they were
Dave Cotton wrote:
One thing is to do a factory reset to reinit everything, I did that with
my 9112i after upgrading the firmware.
I just did that. Now Asterisk is giving me the follow error: (0.99 is
my Asterisk server and 0.111 is the phone)
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