You could read the source code, but based on it's name I would say it is a
library responsible for zone specific tone generation. Many parts of the world
have different tone patterns than the U.S. and Asterisk is used worldwide. A
better question is, why are you concerned by it?
I was building
Trying to find out what the libtonezone shared object built with dahdi-tools is
for, the default dahdi package installation from the Digium repo's pull it in,
so when is it needed?
Thanks,
jlc
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I build them for the kernels I use (Fedora 12 x86_64 and i686.PAE) but
you can check http://messinet.com/trac/rpms/ and checkout the svn-build-rpm
tool to build from an svn checkout if you already have a build setup
configured.
Anthony,
I appreciate the pointer, and I do have a build environment
I build them for the kernels I use (Fedora 12 x86_64 and i686.PAE) but
you can check http://messinet.com/trac/rpms/ and checkout the svn-build-rpm
tool to build from an svn checkout if you already have a build setup
configured.
Anthony,
So this script builds them with the dahdi-tools-libs
git clone http://git.tzafrir.org.il/git/dahdi-extra.git
cd dahdi-extra
make gen-patch
And use the generated dahdi_linux_extra.diff . It includes OSLEC and
some other things. See the Makefile there for more information. The
patch should be applied with -p1 .
This repository includes the
Basically - yes. It's an extra patch to add to your source RPM. Are you
familiar with modifying them?
Tzafrir,
Vaguely, I would very graciously take any suggestions you could provide:)
The whole dahdi package routine has change since the last time I used it,
was shortly Jason Parker started
atrpms.net also provides packages for RHEL5, if those would work.
http://atrpms.net/dist/el5/
Just on my way to work on this server now, this would be great! That
way I don't have to work all night:) Does the atrpms ones finally do oslec?
Thanks!
jlc
I don't use them myself, but I was thinking that the RHEL5 spec files might be
another place to look for what you need to build with OSLEC included, more
specifically for CentOS. I just tried taking a look at ATrpms, but the site
is having some connection issues at the moment.
How about this
Looking at the source in the rpms from the asterisk package site
appears that oslec is not built and enabled for the kmod rpms.
Anyone know an existing repo or have direction on how to enable
this to built for those rpms?
Thanks,
jlc
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Not sure how to go about troubleshooting this, did a fresh install
of CentOS 5.4x86 with a netinstall iso off the base and update repo
followed by a install of dahdi-lniux/tools from the digium/asterisk
repo, ran genconf on my single fxo tdm410p and rebooted, ran fxotune,
rebooted and now this
After using the CentOS repo's at digium to install dahdi Linux tools, I got
this:
Installing : kmod-dahdi-linux-fwload-vpmadt032
WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs
unknown symbol voicebus_transmit
WARNING:
Even simpler:
exten = s,n,Set(Dialnum=${IF($[${ARG1:0:1}=1]?${ARG1:1}:${ARG1})})
Thanks Tilghman,
I am making a note of this as well!
jlc
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I have a dial cmd buried amongst a series of others in a macro
like so: exten = s,n,Dial(SIP/1${ar...@sip_peer,60,T)
Reason for adding a 1 is all the others in the macro don't
want the 1 so this was easiest at the time. Now I need to
send NA long distance through this macro. All the other dial
Is there some way to simply add some logic above it such that
if the EXTEN coming in starts with a 1, remove it so I don't
have to hack this extensions.conf all to heck?
Ok, a bit more searching and maybe I have it (I'm remote and cant
test this, so before I call in tomorrow I'd like to get it as
exten = s,n,ExecIf($[${ARG1} = 1${ARG1:1}
]?Set(Dialnum=${ARG1:1}):Set(Dialnum=${ARG1}))
Much simpler Dhaval, thanks!
jlc
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I then fire up twinkle on my desktop and dial sip:3...@pbx.mycompany.com.
The Asterisk console shows:
[Jun 17 22:58:55] NOTICE[32060]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension '36' rejected because extension not found.
If I use the same extensions.conf but change s to 36,
When I find the rc in the release name dahdi-linux-2.2.0-rc5.tar.gz, then
what does it mean the rc5?
Release Candidate.
Which is better, to select dahdi-linux-2.2.0-rc5.tar.gz or to select
dahdi-linux-2.1.0.tar.gz? I am afraid that rc means still not finally finished
and has bugs?
Any
A persistent local DNS cache such as pdnsd[1] or djbdns[2] could help.
[1] http://en.wikipedia.org/wiki/Pdnsd
[2] http://en.wikipedia.org/wiki/Djbdns
Philipp Kempgen
I am guessing it fails to reverse lookup your internal addresses (which
would fail anyway, even with the DNS up). If your
Sure. I might name it something like dhcp127 though.
That makes sense :)
This must be your dialplan. Can you post it?
You are right, never trust users :) They had erased it or something,
it actually wasn't there. So it does go straight to vm as it should.
My bad...
Close. The packets won't
I have a single server running asterisk 1.6.0.8 with a few sip voip providers
and a tdm card for redundancy. It has a caching name server and the sip
providers
are hard coded in the hosts file.
When the internet connection dies, it fails over to the dahdi channel as it
should, but slowly the sip
While I was in the console looking for something else, this appeared when I
called in on my cell.
[May 26 12:17:26] NOTICE[3364]: chan_sip.c:17229 handle_request_invite: Sending
fake auth rejection for user xxx xxx xx
sip:xxx...@xxx.xxx.xx.xxx;tag=as04e93fb9
What does this mean?
I have a caching name server setup on one of our units but after a prolonged net
outage the internal phones stopped working as well. In searching the bug tracker
I see the bug is still not fixed even though it was thought to be (using
1.6.0.8).
Some suggestions where to set srvlookup=yes but I
The only cleaner way is to define the group in [globals] as follows:-
[globals]
group1 = SIP/3615221401SIP/3615221402SIP/3615221407SIP/52260014
...and then refer to this variable in the dial statement...
exten = 5226001454,1,Dial(${group1},20)
That certainly makes life easier, is there a way
I am about to setup a new machine and based on a thread in the freetel-oslec
list, I came across the idea of compiling Intel optimizations in when using
oslec w/ dahdi. So I edit
dahdi-linux-complete-2.1.0.4+2.1.0.2/linux/drivers/dahdi/dahdi_config.h
to #define CONFIG_DAHDI_MMX which on its own
Why does enabling the mmx in dahdi_config.h break compilation?
I get the following:
{standard input}: Assembler messages:
{standard input}:86: Error: suffix or operands invalid for `mov'
{standard input}:87: Error: suffix or operands invalid for `mov'
make[3]: ***
What are the differences, or where do i find docs on the difference
between the 1.6.0.x and 1.6.1.x release?
Thanks!
jlc
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Occasionally, DIDs from different providers stop working for some reason.
I would like to be able to monitor situations like that and react before any
of my clients start going ballistic on me.
Any ideas? Scripts you know of or wrote and willing to share?
Any info would be greatly appreciated.
I
Any way to initiate a call and execute a playback of an audio file from the cli?
My only chance to debug or make changes is usually when no one's at the office
including me!
Thanks!
jlc
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Have a look at 'call files' on voip-info.org
That worked well.
Thanks!
jlc
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At the top of my /etc/dahdi/system.conf file is this line:
# Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009 --
do not hand edit
OK, so how do I adjust the timing source and LBO numbers, and echo cancellers
if I'm not supposed to edit this file?
Well, if you hand edit
Have you tried this?
'su - asterisk
'cd /var/lib/asterisk/moh
When this works, so will *.
Yup, I should have stated that more specifically:
# ll /var/lib/asterisk/moh
total 6604
-rw-r- 1 asterisk asterisk 1939794 Sep 20 2006 fpm-calm-river.wav
-rw-r- 1 asterisk asterisk 2582196 Sep 20
The specific error is that it cannot chdir to the music on hold
directory. Are you sure you have the right directory?
what do you get when you do:
CLI moh show files
and
CLI moh show classes
The specific error is that it cannot chdir to the music on hold
directory. Who owns the parent directory?
I am running Asterisk as non root and have set the required permissions for all
directories including the moh dir specified in musiconhold.conf yet asterisk
still complains it doesn't have access when starting? I get:
WARNING[3600]: res_musiconhold.c:987 moh_scan_files: chdir() failed: Permission
Have you tried your system stuff under su - asterisk? Once it works that
way, the system() command will work.
asterisk is running as root, I run the command at the terminal as root.
I am guessing he doesn't even have an asterisk user.
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I get the following error when I execute reload in the cli on one of my
boxes with a TDM400 card w/ one FXO port:
WARNING[26444]: chan_dahdi.c:14313 process_dahdi: Ignoring signalling at line
20.
I spent some time to understand what's missing in the OSLEC patch for
dahdi... I can confirm the same problem you reported some days ago and I
need OSLEC for home personal use.
Wow,
Appreciate the info! I will need a few days to get this done. Out of curiosity,
how do you find this ec's quality
It bombs out when compiling manager.c
On what platform is it?
Fails on CentOS 5x86 as well.
jlc
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I did build dahdi before building asterisk, but that`s it.
No problem. But what steps did you use? Did you edit *any* dahdi related
configs? See the voip-info url below.
I find it hard to find any documentation referring to dadhi instead of zaptel.
:) Yeah, it's not the most documented aspect
Yesterday I pulled in the latest svn of Dahdi and added the files
from a recent kernel in the drivers/staging/echo structure and modified
the Kbuild file so it would compile without error. I insmod'ed the module
in, and modified my system.conf has echocanceller=oslec.
cat /proc/dahdi/1 shows:
I have an issue with Dahdi trunk and Asterisk 1.6.0.1 where my analog line is
call
forwarded on no answer or busy to my sip provider.
When we call in on the analog line, I can see the call begin in the cli, and
after 15
seconds I see the call switch over to my sip provider, and after about 30
When we call in on the analog line, I can see the call begin in the cli, and
after 15
seconds I see the call switch over to my sip provider, and after about 30
seconds I get
the 3 raising tone signals and the call is hungup.
Sorry guys, been a long day staring at the tube:) Answer() followed by
Am I doing something wrong?
I just posted this exact issue on Wednesday:
http://lists.digium.com/pipermail/asterisk-users/2008-November/222063.html
I never got any response and Digium came through with keys for my HPEC
license in the nick of time. I am not pleased with the admin overhead
HPEC
Have you copied there the files from the directory drivers/staging/echo
in a recent (that is: = 2.6.28-rc1) kernel tree?
Tzafrir,
Thank you for following up on this. I don't have a quick command for only
the three files, I just grabbed the tar ball. But like the OP, the only
difference was that
I create my sip users using a common macro in 1.4:
[internal]
exten = 200,1,Macro(phones|200|SIP/200)
[macro-phones]
exten = s,1,Dial(${ARG2}|45|Tt)
etc...
But now in 1.6 this fails:
-- Executing [EMAIL PROTECTED]:1] Macro(SIP/201-0942b530,
phones|200|SIP/200) in new stack
[Nov 20 08:55:55]
AFAIR it was mentioned in UPGRADE.txt that argument separator was
changed from pipe to comma. Unless you read it, you might also
experience lot of other problems.
Whoops, missed that! I did see the suggestion on GoSub's but as it
stated Macros would still be supported I neglected to attempt to
What you might want to do it try OSLEC
Gordon,
Digium hasn't responded to me with my key to install HPEC after
waiting several days, and tonight I need to get the card installed
as my number port takes place and that location will be w/o phones.
I am using Asterisk 1.6 and DAHDI and from what I
Not trivial but not as voodoo as before:
http://docs.tzafrir.org.il/dahdi-linux/#_oslec
Tzafrir,
Appreciate this pointer, I am intending on setting this up on a CentOS 5 x86
box. The drastically different stock running kernel compared to the files I need
from your doc won't be an issue? Also,
Not trivial but not as voodoo as before:
http://docs.tzafrir.org.il/dahdi-linux/#_oslec
Tzafrir,
I pulled down linux-2.6.28-rc5.tar.bz2 and followed the doc, now
when compiling I get the following:
WARNING: oslec_create [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko]
undefined!
I have incoming analog and SIP DIDs that all ring multiple
sip extensions with a Dial command as the first exten. I
am curious to know if it's possible for the incoming caller
to transfer out of the Dial command while in progress and
dial a single extension?
Thanks!
jlc
lsmod | grep dahdi
dahdi_dummy38984 0
dahdi 231888 1 dahdi_dummy
crc_ccitt 35265 1 dahdi
How did you compile and install this? Did you simply make, make install,
make config and chkconfig dahdi on? I assume you edited your /etc/dahdi/modules
as your
I compiled dahdi 2.0 complete with:
make all; make install; linux/build_tools/genudevrules; make config
As per the readme, I did #make, make install, make config and then double
checked chkconfig
and although I think /etc/dahdi/modules is for controlling what loads.
I suspect as I also have
Does this make a significant improvement? The box in question I was going to
try this with has a 4 port TDM card w/ plenty of horsepower, but I do intend
to later migrate to a Soekris unit running Astlinux and therefore might not have
the power to run it after. If the difference is significant, I
I was playing with 1.6.0.1 and the latest gui and wondered how my sip
did was registered after creating it? How does this take place, normally
I made a register = command in sip.conf but don't see this in any files?
Thanks!
jlc
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I'm not the Sysadmin type so I don't want to have to labor over manual
upgrades once a
month or so - and that's the big argument against rolling my own * box
and doing everything from source. I'd rather be able to click
'upgrade', have it go do it's thing and trust that it's going to work.
Alternatively, you might fully unscrew and remove the front plate,
insert the card to fit properly and then either live without a
frontplate
That doesn't sound safe, a pull on a cable, or deploying the server
on its rails could unseat that card.
or mill the front plate to fit.
Funny, I
I have 2 HP Proliant 365 G5 servers with PCI-E risers. I bought a Digium TE121B
single port card. When installing the card, the slot on the card doesn't quite
line up with the tab in the PCI-E slot. If I loosen the front plate on the
card,
Ican sort of make it plug in, however, the card won't go
I define my sip users (phones) by using a macro. Is it possible to dump
these into an agent pool automatically w/o requiring a password either in
my extensions.conf macro so I could always have one dial syntax throughout
my dialplan instead of the array of SIP/{ext} I have currently in my dial
What is involved in provisioning Asterisk to use a multiline analog service
from our local telco?
I will only have one twisted pair entering in on a OpenVox card but am not sure
how Asterisk
interprets and deals with two incoming calls and/or two outgoing calls?
Thanks!
jlc
OpenVox.
Gordon
Appreciate that pointer, those are fairly cheap!
Thanks,
jlc
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I need to increase reliability at an office as SIP/Internet provider outages
are causing some issues.
What would be the least expensive analogue card that people are using reliably?
Thanks!
jlc
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X100P.
Yeah I saw these but they are single port and I need at least 2 ports. I only
have 1 free pci slot as well.
Thanks!
jlc
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Incoming calls ring SIP users who have |Ttr in their dial plan, but outgoing
calls are done through a macro as follows:
[macro-diallink2voip]
exten = s,1,Dial(SIP/[EMAIL PROTECTED],120)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Hangup
exten = s-CONGESTION,1,Dial(SIP/[EMAIL
I started this at 4pm yesterday, its 10am and the handsets still say they are
in progress?
Is that normal?
Thanks!
jlc
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The wiki says it should take about 20 minutes per handset.
yeah I just found that, and so I called tech support
and they said to reset the gateway, and if needed to pull
the battery out of the phones and power them on. I have done
this and they restarted the firmware download so I will wait
and
Does anyone have any perspective on how well Asterisk performs and
scales inside a Xen hypervisor environment?
I tried on many different pieces of hardware with various recent Xen
versions and it always had some level of unpredictability and was not
as reliable as running on bare hardware. I
core show application voicemail
So, in your voice mail context you'd have:
exten = a,1,VoiceMailMain(@sip)
exten = a,n,HangUP()
Thanks Doug,
Working great!
jlc
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AstriCon 2008 -
Now that we have voicemail working, people have asked to be able to
dial in externally and be able to access their voicemail. My dial plan is
simple, after ringing a few extensions for some time, it goes to voicemail.
What needs to happen to allow for someone to switch out of this into
Press *
Steven,
Appreciate the info but there must be something I missing as a
prerequisite to this feature. It has no effect at any point during the
call and message?
Thanks!
jlc
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I have an Asterisk server running iptables with a public interface
and an internal interface. I had to change the subnet of the internal
interface and now I see messages scrolling destroying.. 192.168.100.1
which is the old of the internal interface?
Sometimes outside calls are ringing busy and
I have a setup with a SIP DID inbound, and several SIP phones inside.
Obviously if the SIP phones are off/unplugged/otherwise not available,
incoming calls ring busy. My extensions.conf looks like this for inbound
calls:
exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr)
So what
exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr)
exten = _1xx,n,Voicemail([EMAIL PROTECTED])
Use whatever voice mailbox and voicemail context you want.
Well, its not advancing when *no* phones are online, just ringing busy.
It does however step through just fine when they
Check your bindaddr in sip.conf. Also check to ensure that you've restarted
Asterisk since changing the subnet. There are more than a few places that
we cache network information for speed purposes, and restarting the process
will fix that.
--
Tilghman
Got it, thanks!
jlc
exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr)
exten = _1xx,n,NoOP(Dial Status: ${DIALSTATUS})
exten = _1xx,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _1xx,n,Gosub(s-${DIALSTATUS},s,1)
[s-BUSY]
exten = s,1,Voicemail([EMAIL PROTECTED]|b)
Doug,
I am about to setup a new Asterisk box which only uses SIP.
I used to simply use menuselect with Zaptel and choose the tools
that Asterisk required to exist and ztdummy.
Now with Dahdi, I am reading
http://svn.digium.com/view/dahdi/tools/tags/2.0.0-rc2/UPGRADE.txt?view=co
and I understand I no
You can read more on my blog (in my sig below) by clicking on the
Asterisk tag for example.
Cheers
Al
Al,
What did you finally settle on as a firewall for this project?
jlc
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The question you haven't answered yet, Joseph, is how does your
Meridian connect to the PSTN?
Is it a T-1 now, or analog?
Sorry Jay,
I ended up in an offline conversation with someone regarding this.
Its on an analogue setup, it has an RJ-21 connector coming from a
punchdown block next to it.
I
The migration does not have to happen all at once, you can take it
slow, make it invisible to the end user, start using VoIP trunks and
all that Asterisk has to offer, and have a super flexible migration
path.
Steve,
Lots of good info! So if I put a T1 card in an Asterisk Server, and a T1 card
We have an older Meridian Norstar system and are thinking of using Asterisk
behind it
to use a SIP Voip Provider instead of our local telco.
Does anyone make an interface card that can integrate with the digital input of
the
Meridian. Not the optimal solution, but it allows for the current
By digital input do you mean a T1 interface? If so then yes several T1
interfaces are available. However I think you mean is there a gateway to use
the Meridian/Norstar phones with Asterisk. If so, yes there is a company
that makes a gateway to use the Nortel p-phones with a SIP based system.
Not odd at all as far as I'm concerned - I know a number of places that
segregate LAN traffic from VoIP traffic using multiple VLANs over the
one physical link. VLANs would be the best solution (short of running
multiples cables for PC and phone) to achieve this.
I would have about 30 phones I
This is almost standard with voip calls. The echo-cancellation has to
train up to the call parameters. Some hardware is better with it than
others and you can try tweaking the value for the echo canceler up and
down. What type hardware are you using - both phone and server?
Hi,
I have Astra
I am being told by the users on a purely sip based setup that when an
inbound sip call is first answered, they here an echo on their greeting
and then the conversation stabilizes and it works well.
Any ideas where to look to start curing this?
Thanks!
jlc
I can not seem to get AsteriskNow to register my SIP provider correctly?
I can do this manually when compiling Asterisk and installing it w/o a
GUI, but not with this. I just get the following message.
-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #22)
The register
I saw this shortly after ssh'ing into a box that was not answering sip inbound
calls:
--- SIP read from 192.168.100.253:5060 ---
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.5;rport=5060;branch=z9hG4bK7a87d233
Max-Forwards: 70
From: xx sip: xx @192.168.100.5;tag=as588c6a60
Still, that's kind of funny though :)
Hilarious :) This CentOS machine running asterisk is in a Xen vm and its not
behaving well.
I am moving it to physical hardware asap and thought that may have been part of
some
indication of the myriad of issues it has. That is a priceless coincidence!
Share the knowledge :P
jlc
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Koch
Sent: Monday, July 07, 2008 10:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED]
Subject: Re: [asterisk-users] rxfax not receiving
So how do we set it up if I'm out of the office, or on the mobile phone and
can't answer the call.
How does it know to go to voice mail?
You set it to ring for a certain duration then go to voicemail after n seconds.
You'll want an incoming call to go to a context at which point you can start
I'm not sure what province you're in, but maybe those clues will help
point you in the right direction.
Trevor
I'm in Alberta, thanks for the clarification. Did you guys get a Whitepages
listing by chance?
I am contacting Superpages now.
jlc
___
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So my SIP Provider states they do not offer the service to list my numbers w/
the Whitepages.
We phoned the Whitepages and they said we can't do it, the SIP Provider must?
Either one/both of them is/are useless or I must switch SIP providers to one
that can get this done.
Anyone familiar with
Asterisk gets very upset if it can't lookup the host name associated
with every IP on the system, normally it would use DNS to do this, but
since your Internet connection was down it could not do that.
So to clarify, it not only needs to resolve FQDN's, but do reverse lookups
on ip's as well? I
If I am not using any additional hardware and only need ztdummy,
would it be sufficient to run make menuconfig and remove all modules
except ztdummy or are there additional ones aside from the obvious ones
used for hardware I don't have?
Given I only have sip voip providers and all my phones are
We had an outage from our ISP this afternoon that cut prevented us from
connecting
to our SIP provider (someone physically cut a line downstream). All our phones
inside
the office stopped working as well? Why is that, and how can I set this up so
phones
can still dial each other inside the
The exact question pose I must leave for others to answer.
However, I recently completed a project that overcomes the situation
you describe. I installed a cellular gateway giving me a wireless
trunk. If I lose IP connectivity I can route calls out through my cell
carrier. Works really well.
What type of PBX hardware do you have on-site? Also what make/models of
phones?
Michael/Darryl,
I do have a local asterisk box, which is why I am baffled. I am new to Asterisk
and there is lots to learn, but my config is pretty basic, my sip.conf simply
has
the phones and single sip provider
in this whole thread are we missing a subtle difference? that being the
difference between inter vs. intra office. when your wan connectivity drops
I'd expect your INTERoffice (from one office to another) calls to fail.
INTRAoffice (within the same office) calls should work though.
Eric
I've seen this behaviour from Asterisk as well... while I can't say I have
tracked it down and verified this... I've seen other talks about how Asterisk
gets rather unhappy when it can't preform DNS queries. I suspect that may be
your problem. Might want to check the archives for other issues
They can now turn off their internet connection and everything works fine.
We left the internet down for 30mins.
I am worried that if the cache time on the DNS server runs out the problem
may come back, but this is set to 6 hours.
Hope this helps, and if anyone can shed some more light on this
I have my SIP provider and Astra 480i's set to ulaw, but unless my
Snom M3's aren't set to alaw they sound very bad as they pop and drop out?
Why is this?
Thanks!
jlc
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When I exit voicemail or an inbound caller hangs up I hear a busy signal for a
few seconds before Asterisk
terminates the call. I thought this behavior was handled in the dial plan with
a Hangup() command?
How can I correct this?
Thanks,
jlc
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--
I had my incoming call time set 120 seconds before going to voicemail,
apparently this
timeout is longer than some existing timeout of ~60 seconds and the call
terminates
before it reaches my voicemail command.
Is this an Asterisk default setting or could this be something on my SIP
providers
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