Follow this link for some more info. Maxim IC just released a couple of
chips that handle the
details of 802.3af for you.
http://www.maxim-ic.com/view_press_release.cfm/release_id/925
Matteo Brancaleoni wrote:
Hi.
The POEI simply connects the four ethernet signals on each of its inputs
I have tried to get my TDM400P card to automatically dial a number or run an
application when I pick up the phone without much luck. After reviewing the
email archives, config files and source to chan_zap.c it appeared that
all I had
to do was set immediate=yes in the zapata.conf file and have a
Lesher wrote:
On Sunday 25 January 2004 18:17, Clif Jones wrote:
I have tried to get my TDM400P card to automatically dial a number or
run an application when I pick up the phone without much luck. After
reviewing the email archives, config files and source to chan_zap.c
it appeared that all I had
to. Nice little
surprise. I
will try restarting asterisk. Wonder what else under /etc/asterisk
doesn't get reloaded
upon the reload command???
Tilghman Lesher wrote:
On Sunday 25 January 2004 18:17, Clif Jones wrote:
I have tried to get my TDM400P card to automatically dial a number or
run
I noticed this too and it is a pain to look at. I saw it because some
of my SIP phones were turned off and
the NOTIFY's for no voicemail reached maximum re-transmissions. Duh!
Nobody was there to answer it.
I didn't check to see what the log level was but if it only shows up on
-vvv console
I haven't taken the time to reverse engineer this on * but subscribe is
used in SIP for serveral things:
1. Message Waiting Indicator (MWI). Asterisk seems to send out a NOTIFY
even with no SUBSCRIBE
though. :)
2. SIMPLE (SIP Instant Message Presence Leverage Extensions). The
It was actually a good question. When I learned Unix internals, the
shared libs and executables
where busy when loaded because of swap-in/swap-out requirements. Swap
space was
used to store the core memory for the apps, and the app itself was
memory mapped when
needed. That is why you
Comments below.
Rich Adamson wrote:
I've been looking around for a small external sip fxo gateway, sending
emails to possible vendors, etc, and can not seem to come up with anything
that fits. Suggestions anyone? (No channel bank T1 card suggestions,
please. I've also just completed an eval of
Rich,
If the Mediatrix uses the Caller-ID field to select which channel to
use, then you really have no
choice but to do that. As you pointed out, the Caller-ID info is not
(and cannot) be passed to
the PSTN line.
Rich Adamson wrote:
Possible Mediatrix 1204 fxo sip gateway workaround
Need
Rich,
Try it again after executing: sip debug and give us the actual SIP
messages. The devil
is usually in the details.
Rich Adamson wrote:
Anyone have comments on this? Really could use some suggestions or ideas
why this is happening. Thanks.
Rich
Anyone
No they do not. I am managing an installation running 7960 SIP release
6.0 and the phones
are on about 4 different subnets. Half of these are on remote VPN
connections at people's homes.
Chris Clifton wrote:
So do the 7960's have to be on the same subnet as the * box ?
This seems like a
: Clif Jones [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, February 05, 2004 9:02 AM
Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
No they do not. I am managing an installation running 7960 SIP release
6.0 and the phones
are on about 4 different subnets. Half
Rich,
It is very important (at least to me) to have the whole SIP call flow.
That is, I must see the initial
INVITE come from the originating phone all the way to the last message.
I can only speculate at
this point but it appears that the second leg (destination) may never
have ACK'd the
I'm having some problems with a SIP FXO gateway working with Asterisk when a
call that involves the gateway is put on hold. This gateway was working
up to a firmware
upgrade but I believe it may have been working for the wrong reasons.
Here is what
happens:
1. User calls in from PSTN to SIP
A typical response from the SIP UAS if no intersecting media types are
found is:
415 Unsupported Media Type
Some user agents also add a warning header to tell you that it couldn't
find a
usable CODEC.
Maciek Kaminski wrote:
Regovich, Timothy wrote:
Not ACK'ing an invite can be problematic for
If anyone is familiar with the SIP SDP handling routines I would appreciate some
insight. The following problem that I found using Asterisk appears to be improper
handling of a call put on hold when there is no music on hold:
[FXO gateway] [Asterisk]
still be
considered active, regardless of the RTP that may or may not be happening.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Clif Jones
Sent: Tuesday, February 10, 2004 1:33 PM
To: [EMAIL PROTECTED]; asterisk users
Subject: [Fwd: [Asterisk-Users
I have 3 Pingtel phones and have tested them since they were
prototypes. I have had no lockups or
weird problems with them on Asterisk. I will says this about them though:
These phones are BIG on features and extensibility through Java at the
cost of quality. It doesn't
take a lot of work to
I have been struggling with several mediocre SIP FXO gateways on
Asterisk for the past
6 months and have found that each one has at least one major problem
with it. I am looking
for any success stories using 1 to 4 port SIP FXO gateways on Asterisk.
I need for them
to support RFC2833 DTMF
Gee, maybe I'm missing something, but the spec does not say that. The
RFC actually says that
when you send a final response, you are required to store that final
response for 64*T1 seconds
and retransmit the final response each time you receive the
retransmitted request. (T1 = 500ms)
I wasted a lot of time on this issue with an Audiocodes FXO gateway I am
currently exploring possible
workarounds. Of course, I would really like to see a patch on the
asterisk-cvs mail list. :)
Gerard O'Rourke wrote:
Hi,
We are using Asterisk for a h323 / SIP converter.
We are having
Ahhh, you must have upgraded to firmware version 4.2. I had the same
problem because
I didn't find the new parameter that they added in this release for
broken RTP connections.
Here is how I fixed it:
BROKENCONNECTIONEVENTTIMEOUT = 36
This makes the gateway drop the connection after an
I know a little history on the 3com SIP phones... We have about a dozen
of them
where I work. I'm not familiar with the NBX100 model number but the ones we
have are labeled: P/N: 655005001. The first ones didn't support SIP out
of the
box and had to be upgraded with a new flash image. I can't
The IR device is a 3rd-party piece of hardware from Extended System (now
owned by
iFoundry). The SIP phone looks like all of the other 3com IP phones
that I have seen
and turning it over with the front of the phone facing up the connectors
go from left to
right as follows:
1. Handset connector
and Asterisk attempts to use XML. I took
pictures of the
main board but forgot to bring them in to work. If anyone wants any
detailed info on the
unit, let me know in the next couple of days before I re-assemble the
device.
Clif Jones wrote:
The IR device is a 3rd-party piece of hardware from Extended
I am a fairly new user of Asterisk and I am generally impressed with its
features. I have some questions about the SIP channel support:
1. I have noticed that even when there are no active calls, there is a
list of
active SIP channels. This appears to be a bug. Has anyone seen this?
2. If
I have setup Asterisk to work with a SIP gateway, some SIP phones
and the Digium FXS/FXO development card combo on another *
box with pretty good results so far. Here are a couple of questions
that I have that wasn't obvious from the documentation:
Voicemail vs Voicemail2 - What is the major
I was looking at some fixes in the replies to the chan_sip.c problems and
I am wondering if I am seeing the same thing in the earlier file version. I
just checked to see that my chan_sip.c is version 1.179 when I did my
checkout so I never had the later versions. The problem that I am seeing
is
Ernest,
There is a beta load that you can get from the Audiocodes dealer which
is working for us.
We are using their 4-port MP-104 SIP gateway and the only problems we
have with it
are:
1. Outgoing calls go out to the lines in a round-robin fashion. You can
put any number of the
lines in
I am relatively new at Asterisk but have a 2-machine system running with
the developer kit (FXS/FXO cards), some Cisco SIP 7960's and a Audiocodes
MP-104 SIP gateway. I would like to fix a couple of the voicemail boxes
so someone can press some numbers such as 911 and get sent to a priority
I have searching the forums here on how to get Music On Hold working
and I have been able to get * to accept a command for MusicOnHold
and for Meetme after loading the ztdummy module. I used the default
config for /etc/zaptel.conf since I saw no guidance on this. My problem
now is that when I
I have 2 PBX linked together with IAX using the GSM codec. This link is
over a T1
that is shared with other traffic. I know that it is problematic using
ethernet to control
QOS so I would like to hear some practical solutions from other users.
___
I am trying to music on hold but I am having all sorts of problems with it.
I am running RH9 and the latest version of Asterisk as of yesterday.
Here is what I did to test it:
1. I manually deleted the mpg123 softlink to mpg321.
2. I downloaded mpg123-0.59r-1.n0i.src.rpm, compiled and installed
For anyone running RH9 with a recent version of *, if you are using
music on hold
I would be interested in what version you installed or compiled. The
version described
below is not working properly and leaves core files in the mohmp3
directory for me. :(
Clif Jones wrote:
I am trying
Here are some ideas for anyone with some extra time on there hands.
SIP phones on call pickup either use a special REGISTER or you can
place a call with the magic extension and have the switch hang up
on you and immediately call you back. With the second option, you
could dial *8, Asterisk could
I am currently using Asterisk with G.711 codecs and in-band DTMF for
several Cisco 7960's
and an Audiocodes GW. When allowing out-of-band DTMF, I could use
voicemail menus and
anything else on Asterisk that required DTMF but I could not get the
DTMF relayed out of the
GW. Has anyone verified
I am currently using Asterisk with G.711 codecs and in-band DTMF for
several Cisco 7960's
and an Audiocodes GW. When allowing out-of-band DTMF, I could use
voicemail menus and
anything else on Asterisk that required DTMF but I could not get the
DTMF relayed out of the
GW. Has anyone verified
Is there an option to configure Voicemail2 to NOT store the voicemail
messages on the disk once they have been emailed to one's mail server.
It can be a pain for some to receive voicemail via email and then go to
Asterisk just to clean out the voicemail you just heard.
This looks to me like the approach that Pingtel took for NAT. I think
it is a good option to
have but having STUN as an additional option is really what we want.
You can find an
implementation of a STUN library and apps at www.vovida.org. The
External IP approach
has some flaws and can be a
Best of luck on getting an answer, I have posted several times with the
same question.
Unfortunately my time to reverse engineer this problem right now is low
but my
temporary solution's cons are pushing me to jump into the code and fix
the problem.
As a workaround you can set your Cisco phones
This company seems to think pros outweigh the cons for Asterisk:
www.voicepulse.com
/. reported today that VoicePulse uses a variation of Asterisk to run
their Broadband Phone Service.
http://slashdot.org/article.pl?sid=03/11/05/1319251mode=threadtid=126
Steven Critchfield wrote:
On Wed,
Are you saying that the clicking only occurs when 128-bit WEP is
enabled? If this is the only
thing causing it (not network congestion), my guess would be that the
same processor doing
the encryption is also trying to drive the voice codec. We have this
same problem on our handheld
devices
I am currently helping a friend build an Asterisk PBX that spans
several cities using anything from T1s to DSL connections to
link remote SIP phones, IAX gateways, etc. to a central Asterisk
PBX server that serves up voicemail, features, etc. The biggest problem
that I have had with this system
Has anyone found any good online resources for performing transmission
tests for POTS lines? There is plenty of info on this list about
adjusting gains
on X100 cards, etc. but I am looking for test procedures using test sets.
I'm talking about tests for echo loss, distortion, etc. Thanks in
We still have a few of the 3com phones in use at our company but we do
not support them with our
SIP products. The 3com phone was meant to be a PBX feature phone as you
stated and as a result
the flash ROM and RAM was not beefy enough to support the SIP protocol
as it matured. The last
ROM
I have finally crashed Asterisk for the first time and I'm wondering if
anyone has seen this.
This is a configuration with SIP endpoints and an IAX2 channel to
another Asterisk PBX.
The main PBX dropped a core file after a SEGV (signal 11 ) with the
following trace:
#0 0x42079133 in strchr ()
Also I have found that safe_asterisk needs to have something like
sleep 5 following the
echo Restarting Asterisk If not, asterisk will immediately exit
with return code 1 after
restarting.
Clif Jones wrote:
I have finally crashed Asterisk for the first time and I'm wondering
if anyone
Thanks for the truly useful feedback. I'm having a real hard time with
the FAQ pages listing
RH 8 9 FIRST in the list of Linux distros that Asterisk compiles and
runs on and having
any bugs (oh I mean RH problems) discarded. It would be much more help
to have responses
such as yours or to
Thanks for the quick feedback! I don't have a lot of free time to play
with Asterisk
right now but a friend of mine wanted me to get it working on Red Hat
for him which
resulted in the RH problems/questions. Personally, I prefer Debian
which suites my
needs for embedded projects and hacked up
I have had several cases where the message waiting indicator was stuck
in the on state
with Cisco 7960 SIP phones. Here are the two cases:
1. Single extension that mapped to a single voice mailbox. Restarting
Asterisk or getting a
new voicemail then clearing it fixed the problem.
2. Three
No takers? Should I submit a bug report then? I didn't find any open
bugs on stuck
MWI.
Clif Jones wrote:
I have had several cases where the message waiting indicator was stuck
in the on state
with Cisco 7960 SIP phones. Here are the two cases:
1. Single extension that mapped to a single
I would stay away. I have evaluated these units and returned them. I
determined that this
unit or one from them that fits this description was actually a unit
that you put between
your phone and your phone line (1 FXS 1 FXO claim) and hook to the
ethernet. This
unit would connect the FXS
I have just added rules for working hours and have noticed some problems
with outgoing announcement using Voicemail2 and SIP. For normal working
hours, I basically use the example macro-stdext and outside of working hours
I have a macro that has the first command as Voicemail2. If someone calls
I am wanting to perform toll bypass using multiple gateways for outgoing
calls.
For example, if I call from location A to location B and I have a gateway in
location B I obviously want to use location B's gateway to make it a
local call.
I understand how to get the local prefixes from NANPA but my
that helps,
MATT---
-Original Message-
From: Clif Jones [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 11:21 PM
To: asterisk users
Subject: [Asterisk-Users] Dialing area question
I am wanting to perform toll bypass using multiple gateways for outgoing
calls.
For example, if I call from
I am thinking about using the G729 codecs on my endpoint devices and
purchasing some G729 licenses for Asterisk but I have several questions:
1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I?
2. If I have G729A on one end and G729B on the other, are they compatible?
I have
Found it. Anyone interested can look in RFC3551 RTP Profile for Audio
and Video Conferences with Minimal Control.
You can piece together that G.729, G.729a G.729b will play together
and the other annexes will not due to
bandwidth differences.
Clif Jones wrote:
I am thinking about using
-through feature then I guess you are fine. I have SIP users
going to h.323 g/w and I need g.729. So now I have it in pass-through mode,
I think that requires less CPU overhead and I do not have to mess with
licenses.
Cheers
SW
Message: 5
Date: Thu, 18 Dec 2003 13:59:06 -0500
From: Clif Jones
Interesting. For the record, the MultiTech MVP-130 comes with a default
setting
of 60ms packets on all of its supported codecs. I changed the packet
sizes to
20ms because I had never heard of anyone using such large sample sizes.
Andres wrote:
On Monday 22 December 2003 19:58, Rich Adamson
I have been trying to get automon working on Asterisk 1.2.12.1 and I am having
some problems. I have searched the list archives and have not found my answer
either. This system is setup for SIP to SIP calls with G.729 codecs. I
believe that I have the config files setup (*1 enabled in
Thanks for the response. Answers inline..
-Original Message-
From: Michael Neuhauser [EMAIL PROTECTED]
Sent: Oct 3, 2006 10:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com, Clif Jones [EMAIL PROTECTED]
Subject: Re: [asterisk-users
I have a friend with a PRI coming into a modem bank that is receiving
56K modem calls and some ISDN data calls. He wants to dump his analog
office phone lines and use some of the capacity on the PRI. I have been
digging through the mail archives and Wiki site on this subject but the
are able to do this because this would suggest that the
quad-span card would offer more advantages in this setup than just
expansion capabilities.
Peter Svensson wrote:
On Sat, 11 Sep 2004, Clif Jones wrote:
I have a friend with a PRI coming into a modem bank that is receiving
56K modem calls
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