Discussion
Cc: Bill Michaelson
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On Tue, Apr 5, 2011 at 1:31 PM, Bill Michaelson wrote:
>fail2ban might be good for this.
>
>
I think you missed the point, which is reducing the need for an external
applic
fail2ban might be good for this.
On 04/05/2011 01:00 PM, asterisk-users-requ...@lists.digium.com wrote:
Date: Tue, 5 Apr 2011 08:44:41 -0700 (PDT)
From: Steve Edwards
Subject: Re: [asterisk-users] Iptables configuration to handle brute
force registrations?
On Tue, 5 Apr 2011, Gilles wr
This is why I don't do this kind of work anymore. Impossible to
distinguish the phantom problems from the real ones - and I'm convinced
there ARE phantom problems when you install new telephones on people's
desks.
Suggestion: learn to use the facility in Wireshark that can log a
SIP/RTP strea
Polycom IP 430 or 330.
asterisk-users-requ...@lists.digium.com wrote:
Date: Mon, 16 Mar 2009 18:24:33 -0400
From: "David Ruggles"
I was looking at the aastra 9133i, however I was informed that this phone is
no longer supported. What are good phones around the $100 - $125 price
point? (Need POE
It's conceivable that the combined effort of these two responders
required less than ten minutes of time, yielding a theoretical pay rate
of $120/hour.
I wonder how much effort went into the other responses.
That will be $6 for my commentary, please.
Folks wrote:
Message: 1
Date: Tue, 3 Mar
Is there a convenient way to limit the number of call files (outgoing
directory) that are processed concurrently?
smime.p7s
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asterisk-
> > What "economic downturn"?
> >
> > I'm sick and tired of hearing this mantra.
I wish you the best of luck in maintaining your immunity.
>
> Same here (in the UK).
>
> As long as people need to make phone calls ...
>
> Gordon
The economy (
"David @ULC" wrote
One of my user was asking, can he use VPN to access asterisk ?
What does it mean ?
And its possible ?
How ?VPN
Sometimes what is called a VPN is not a VPN by everyone's definition, so
beware. By my definition, a (IP) VPN supports full layer 3 functionality
(and sometime
How are you getting these 80 or 120mm fans in a 1U chassis? Remember you
got barely 45mm to play
with at the back and front of the switch. How are you going to mount a
80mm or 120mm fan on there? Are you assuming that the units mounted
above (or below) your switch is a short 1U? You can't assu
Sometimes I do. It depends on my mood and purpose. And sometimes the
author prefers to write last things first, for whatever reason.
I'm kind of agnostic, too.
Mike Dent wrote:
>H, not sure about you but I often pick up a book and flick from
>the back to the front, does nobody else do that?
From: Doug <[EMAIL PROTECTED]>
"Net Neutrality" is great in principle. But ISP's need to
somehow control those few percentage of users who suck down
a huge majority of the bandwidth. It's dollars and cents.
There is a rational solution for the traffic management issue. It just needs
to be
On Sat, Nov 15, 2008 at 03:49:40PM -0700, Joseph wrote:
> What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux
software)?
>
> I'll eventually replace this crippled Linux with something better but I don't time to play around with it as most divers and modules are still too new
Alex Balashov wrote:
Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com
I'm pretty sure they meant two logical networks. At least, I hope they did.
Unfortunately, I was indeed referring to two physical networks. Cabling,
switches, everything, all the way
I'm wondering how prevalent the practice of physically segregating voice
and data networks is in the Real World.
What are the factors that typically lead to such a decision?
DIscussions of pros and cons are most welcome by me.
Experiences, anybody?
smime.p7s
Description: S/MIME Cryptogra
Kristian Kielhofner wrote:
On 10/23/08, Bruce Komito <[EMAIL PROTECTED]> wrote:
> We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP
> connections. I've seen the delay thing, as well as the Sonicwall throwing
> away entries from the ARP table because of inactivity.
Sorry for asking the obvious question, but are there other elements of
the slow path besides the Sonicwall? I mean, what is "in front" of the
Sonicwall? Also, might the Sonicwall be positioned as some kind of choke
point in the topology, thus leading to genuine sporadic congestion?
James Laman
I recently sent this email to a user in response to a problem report of
phone calls going to voicemail without the phone ringing. I'm wondering
if I've covered all bases, or whether there is some logical explanation
I haven't considered, and generally what others' opinions/experiences
are that
Steve Totaro wrote:
My only wish is that Linux had a facility like XP to bridge NICs without
running all sorts of commands for brctl. Just a GUI like XP. Last time I
setup a bridge in Linux, I had to change many kernel options and rebuild the
entire kernel to get bridging working properly. Wi
Jay R. Ashworth wrote:
>In our call center (250 agents), we use Panasonic KX-TCA60 headsets, which
>I think we get for 7 or 8 bucks apiece, and sell to the agents at cost.
Thanks for that - they look good, and I found several recommendations for them
after I got yours and started looking for them
Some users at a new Asterisk installation with Polycom IP330 phones are
complaining about echo with the amplified headsets they used to use with
their Nortel phones. I listened myself, and I here my own voice
annoyingly loudly, and no headset/phone combination of volume control
manipulation pr
>From: "Joseph L. Casale" <[EMAIL PROTECTED]>
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 n
My experience is very limited, but you asked for any perspective, so...
I put an Asterisk with freePBX on a linode server (linode.com), just to
play with it a few months ago. I can say that it worked to the point of
being able to dial out with my Polycom phone on a FiOS connection,
through th
Interesting to see it done. Vitelity claims it is impossible. The number
is 212-651-5632.
BTW, if you provide the originating number, the underlying carrier can
be determined, either by the pooling or NANPA block it is assigned to,
or its LRN if ported. If you want, you can privately e-mail
That is my position, and I appreciate the affirmation, as well as the
offer to determine the carrier. I might email you about that. But having
no business relationship with the other carrier, it is at best awkward
for me to initiate contact on this matter, and this should be obvious to
Vitelity
I have a Vitelity DID which generally works, but calls from a particular
caller do not reach it. Vitelity has thus far disavowed any
responsibility for working through this problem. I recognize that some
action might be required by another provider which is outside Vitelity's
control, but it
I am preparing to connect an asterisk box with a redfone fonebridge to a
T1 service provider. I am doing this by testing first with another
asterisk and a Sangoma card playing the role of telco.
I formerly had this test configuration operating flawlessly as a PRI
connection. But I discovered
Tzafrir Cohen wrote:
I usually configure the entire span of 24 channels (23 B + 1 D) and
only the turned up channels go into service. This is good for a
couple of reasons.
Also note that Zaptel will anyway reserve all the 24 (for T1) or 31 (for
E1) Zaptel channels for the span. So th
On Tue, Sep 9, 2008 at 7:17 AM, Bill Michaelson <[EMAIL PROTECTED]> wrote:
> I'm faced with an installation at a client site with supposed PRI service on
> a fractional T1.
Steve Totaro wrote:
I usually configure the entire span of 24 channels (23 B + 1 D) and
only the turn
I expected to find th module ztd-ethmf[.c...] in support of the redfone
TDMoE product in my zaptel distro (I have 1.4.11). But it's not there.
I am awaiting a response to a trouble ticket from redfone. Can anyone
give me a jumpstart? I can't seem to google this up.
smime.p7s
Description: S
t;
> I suppose the bit to check is the features ('show features') and then
> try to record a call (*1) and see what the terminal says...
>
>
> Bill Michaelson wrote:
>
>> > My client needs call recording features and would like to initiate the
>>
My client needs call recording features and would like to initiate the
process in-call (typically *1). I'm installing Asterisk 1.4.x and
FreePBX 2.4+. I'm using Polycom phones. I can't make it work. Would
somebody please give a checklist of items for me to compare my list
against - in the h
In case this is useful to others, a tip...
I moved one of my Polycom 501's off it's subnet to another one (I've got
an ether bridge glued to the back of the phone and a wireless card in
the * box acting as AP). Now it is still served by the same Asterisk
box, albeit through another ethernet p
I appreciate your report (below), but it's a strange and disturbing coincidence
for me. DTMF out through Vitelity was not working for me until 1-2 days ago
when I changed it from rfc2833 to inband!
Maybe I just missed the change date and I should change it back?
Date: Tue, 22 Jul 2008 1
Message: 1
Date: Fri, 18 Jul 2008 20:35:47 -0700
From: Dave Platt <[EMAIL PROTECTED]>
I'm preparing for a client install of * by doing a fresh one in-house.
Unlike my earlier installation that runs asterisk as superuser, my
current experimental box runs without such privilege. This is ca
I'm preparing for a client install of * by doing a fresh one in-house.
Unlike my earlier installation that runs asterisk as superuser, my
current experimental box runs without such privilege. This is causing
it to moan that it can't set TOS. I absolutely don't want to install it
on the clien
OK, I had broken the feature.conf fileset, but I just fixed it. Now I
can confirm:
pig*CLI> feature show
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# ##
Attended Transfer
A followup to my own inquiry...
pig*CLI> feature show
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #
Attended Transfer
One Touch Monitor
After much checking and puzzling, I cannot get my Polycom 601 to toggle
call recording with my Asterisk 1.4.21.1.
Via FreePBX, I can set a user to always record, and the recording will
show up in /var/spool/asterisk/monitor.
But if I try to start recording by toggling in-call, no luck.
I can
This appears to be a SIP gateway to four FXO ports for ~$250. Has
anybody used it with Asterisk? Comments?
http://www.ipphoneshack.com/products/D_Link_DVG_3104MS_VoiceCenter_4_Port_PSTN_Gateway-193-12.html
Any good reason to pay for a Mediatrix 1204 or some other box instead?
smime.p7s
Descr
I'm looking for reports of recent experience with redfone fonebridge2
(with echo can) TDMoE gizmos.
Anybody? Good? Bad?
smime.p7s
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I'm considering using a PoE switch like this...
http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=3023334&CatId=2800
...to power as many as 24 Polycom phones of varied kinds.
The sales lit indicates >190 watts available for PoE devices. But I'm
concerned about a probl
I've found OpenVPN to be easy to configure and very robust. It has a
zillion options, but they are just that - options. I haven't used it for
VoIP, but I've put it to good use doing layer 2 bridging which has
eliminated many problems with certain programs traversing NAT and
load-balancing route
Hooyoo kiddin? Exit 34, I-80.
And betta Inglish, myass...
Bill, Exit 8, NJTP
Date: Tue, 03 Jul 2007 18:13:47 -0400
From: Mark Phillips <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
Damn!!! Beat me to it ;-}
As an Englishman now living in New Jers
Yes, of course. What happens when you dial the number, Daryl?
Daryl Jones wrote:
> Bill Michaelson wrote:
>> Is it just me, or is the AGI interface at cnam.got-name.com failing
>> for others? Anyone know how to contact them without sending postal
>> mail or telegram
Is it just me, or is the AGI interface at cnam.got-name.com failing for
others? Anyone know how to contact them without sending postal mail or
telegram?
smime.p7s
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Would anyone care to recommend a T1 interface method for Asterisk that
would function as an (external) alternative to a PCI card like the
Digium TE120P? Like some sort of T1-SIP gateway?
Also, would anyone with experience using these products care to comment
on the practical value of the TE207
[EMAIL PROTECTED] wrote:
Date: Sun, 22 Apr 2007 19:38:04 +1000
From: Rob Hillis <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Softphone that supports central
provisioning?
To: Asterisk Users Mailing List - Non-Commercial Discussion
Message-ID: <[EMAIL PROTECTED]>
Content-Ty
l Discussion"
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed I have the
same setup as Florian, however I have dtmfmode set to rfc instead of
inband On 9/16/06, Florian Overkamp <[EMAIL PROTECTED]> wrote:
> Bill Michaelson wrote
Would anyone be kind enough to post a sip.conf fragment as a sample for
use with a Mediatrix 1204?
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I found a bogus binary in my (obviously) hacked system in /usr/sbin. I
am still investigating. FWIW, it was 608828 bytes big. It appears to
have arrived recently, but I haven't determined how. Here is some more
info...
sum /usr/sbin/asterisk.suspect
15139 595
I'm just posting this in c
My 501 admin manual refers to a precenence field, and in another place it
refers to a seven bit value. So I would guess it uses diffserv format. Does
that help?
Date: Mon, 10 Apr 2006 15:32:29 -0400
From: "Jonathan k. Creasy" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Polycom TOS
To: "Aste
From: Martin Joseph <[EMAIL PROTECTED]>
It sounds to me like you are suggesting that a QoS infrastructure can
be utilized over the internet at large? Is this only true for big guys
that have an SLA in place?
I would love to discover some QoS mechanism that is respected in
general, but tha
I suppose that in order to accomplish conferencing, Asterisk must produce a
broadcast audio stream (waveform) which is a numerically combined derivative of
all of the input audio streams. In order to do so, it almost cetainly will
work with uncompressed data. Therefore, encoding such as G.729
Date: Fri, 24 Feb 2006 14:56:54 +
From: Steve Kennedy <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] GPS-enabled cell phone/PDA
On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote:
> Its my understanding the cell phone coordinates are sent to the cell phone
> provider and
I'm configuring a box with a TDM400P with 2 FXS and 2 FXO. I configured
the FXO's first to try them, and they worked (I could talk to myself
thru the PSTN. But when I add the FXS's to zapata.conf and restart *, I
have a problem...
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/a
The answer is yes, I think, but I don't recall precisely how off the top of my
head, and I'm walking out the door in a moment. The phone will hold more than
a dozen distinct ring tones which you can create for yourself, and you can have
asterisk direct it to use a ring tone independently of li
Around 1978, when I was consulting to a multinational company in the
business of agriculture, I witnessed this configuration in their
communications center in NYC:
A paper tape punch attached to a teletype machine was busily punching
out a tape that was being spewed into a wastebasket. Someho
I can appreciate the desire to avoid reconfiguring existing hardware,
but that is part and parcel of what we are discussing: reconfiguring
hardware. Without further specification, it has no bearing on how to
preserve application behavior, which is what we are trying to accomplish
with this dis
Right - so I will assume this makes it slightly more efficient in that
respect. And of course, any solution that uses multiple hops brings in
a raft of considerations for limiting interference by other data streams
- the essential QoS question.
Date: Thu, 26 Jan 2006 15:16:25 -
From: "St
You've clarified your requirements for me. Please indulge me - I really
want to understand - what are the application implications of this? In
other words, what system behavioral changes will your users experience
in the various scenarios (pure circuit emulation vs. relay via IAX or
similar)?
This has been an interesting discussion for me (except for the
sniping). The last post led me, out of curiosity, to this wiki entry:
http://www.voip-info.org/wiki-Asterisk+TDMoE
I was unaware of this feature, and it looks pretty good. I've been
pondering replacing some T1's by leveraging IP
Actually, call groups are yet another layer of complexity. Let me try
another explanation.
With VoIP, the distinction between a call appearance capability and a
line is artificial to an extent. Think of a line as the analog for a
pair of copper wires. Think of a call appearance as call wait
In sip.cfg, add something like this:
...to correspond to something like this...
Please note that I just hacked this example out of my own cfg, modified it for
you, and possibly introduced an error, because it is untested now! But it was
lifted from working code, so it should get you on th
I use dbget to set callerid, but it's based on account code, and set
statically with the station, not the agent. Users can set callerid by
dialing a function coded in the dialplan for that purpose. Overhead is
not a problem.
In your case, perhaps you can set the desired callerid into a chann
valuating. However, if
all other things were equal (and they could become so with just some
firmware improvements), this nonsense will drive me into the arms of a
competing manufacturer in a heartbeat. It's needless grief to put your
customers through, and that's just stupid, in my opi
Anyone know how to obtain firmware and starter .cfg files for Polycom
phones? Despite registering at the Polycom web site, I can't locate
this stuff.
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Just got a Polycom 301 and I'm configuring. Examples given in wiki
recommend using dtmfmode=inband, so that's what I set in sip.conf for
this phone, as I have for various other IP phones on my network. But
the telephone does not seem to send DTMF tones up thru the network
(although I hear the
Running a fairly recent subversion release of Asterisk, I'm running into
a problem using labels (as opposed to priorities) with this application.
Here is the dialplan segment:
; isolate gotoiftime bug with labels
;exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4)
exten => 806,1,GotoIfTime(8:00-20
I am having trouble understanding how to use this. I want to transfer
certain incoming calls from an IAX ITSP based on caller ID. From what I
can make of the docs, I thought I need to do something like this...
exten => _NXXNXX,n(nocid),transfer(1000)
exten => _NXXNXX,n,noop(boo,${TRAN
Here is a work-in-progress that provides pop-up note-taking windows
based on caller-ID, outgoing call dialing from directory lookup
selection, and other stuff.
I hope it's useful to folks.
http://asteroid.from.net
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I'm probably mistaken and unaware of a feature, but I thought the
concept of dialing an agent does not exist. An agent is not a channel,
but rather, someone who associates themself with a station from which
they service a queue.
You "dial" the queue with queue()
Message: 8
Date: Fri
Just curious...
Is there anyone out there who has given this outfit money and actually
received any service from them?
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snacktime wrote:
On 11/23/05, Bill Michaelson <[EMAIL PROTECTED]> wrote:
> I'm working on a manager client that I designed to hold open TCP
> connection to asterisk while it is running for varoius purposes. After
> being puzzled by unexpected behavior, I realized th
I'm working on a manager client that I designed to hold open TCP
connection to asterisk while it is running for varoius purposes. After
being puzzled by unexpected behavior, I realized that the server closes
the connection after it completes an "originate" action - or at least it
does in the c
Due to some change I've been unable to identify, my Asterisk box is no
longer successfully passing caller ID to the called party with calls
placed through Voicepulse. This worked just fine until recently. Also,
identical code functions correctly (caller ID arrives) when the call is
sent via J
A nasty screech. That's what callers here sometimes when they dial into
my FXO port from the PSTN. But usually, it works OK.
Is this common?
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Asterisk-Users@lis
Is anyone using a Grandstream ATA-488 FXO port to connect a PSTN trunk
to their Asterisk box (via SIP, of course)?
Is it possible to have such a beast operate reasonably?
If so, is it also possible to use the FXS port concurrently and
independently?
_
Is there anyway to code for queuing for an available trunk. I thought
of this while reading about Erlang C.
Basically, the idea is that when a caller at an internal extension tries
to place a call via PSTN, but all available trunks are busy, the call is
placed in a FIFO queue for the first av
7:57 -0400
From: Andrew Kohlsmith <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] gotta be a dumb question...
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"
On Sunday 30 October 2005 09:44, Bill Michaelson wro
...but I'm gonna ask it anyway, because I can't figger it out...
Every call that is bridged in my * system begins with a console message
like this one...
-- Attempting native bridge of SIP/215-b09e and SIP/259412-5967
Now, I've got canreinvite=no in every sip definition, but it happens anyway
o-8859-1" I forwarded your note below to
[EMAIL PROTECTED] I found some docas on the FCT-11M at their site,
but it was in Chinese, so I sent them your problem. Hope they will
respond to this list and maybe to you directly. Murrah Boswel -
Original Message - From: "Bill Mic
I recently obtained a FCT-11M GSM-analog converter box. It arrived with
no documentation. So I popped in a SIM chip, and connected the the RJ11
port to an FXO port on my Asterisk box. It worked smoothly right away
for inbound and outbound calls in all respects. For about an hour.
Then eith
Has anyone used the GSM-SIP gateway product produced by a company at
sipcpe.com? Any comments?
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Is it possible to use the Hold/Transfer/Conference/Flash keys of the
Budgetone-101 (FW 1.0.5.22) with Asterisk?
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order in which you do a modprobe on them.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
Michaelson
Sent: Saturday, March 13, 2004 2:18 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dial via X100P
Just connected my X100P to Verizon. I stumbled acr
a 1 before it dials
the 609${EXTEN}
Unless I'm misinterpreting what is happening?
This obviously limits my possibilities. Can somebody explain to me why
it dials 1, or appears to?
--
Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED]
Thanks for putting up with
utgoing" call. And I'll get better quality for
the remaining duration.
But the call instead is dropped at this point instead - both sides disconnected
from the cloud.
Anybody know why and how this is controlled and what my options are?
--
Bill Michaelson - COS, Incorporated - Software
Thanks, Scott. I'm in a general exploration mode, but I do have a small
broadcast application in mind. My limited experimentation leads me to
suspect that there is no queue management at all. I was testing with
only a single call file just minutes ago, and the system tried to redial
the dest
es requests based on the capacity of resource combinations
required to satisfy the requests?
Or is it just a single thread/processing queue for all requests found in
the spool dir?
Also, is there any way to control the sequencing (priority) of the
"enqueued" requests? Or is it a
Anybody know how to implement a hotel wake-up call feature with *?
--
Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED]
Thanks for putting up with my spam filter!
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http
Can someone refer me to an example of an automated broadcasting
operation that sends a canned voice message to a list of phone #'s?
--
Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED]
Thanks for putting up with my spam f
It gets better (worse)...
I had been testing with console (-c) mode. When I allow * to run background,
it crashes after the system() call (which succeeds, by the way). The -vvv
option yields these final messages before *poof*...
== Spawn extension
(intern-post, 112, 1) exited non-zero on
>It is now indeed more discerning, but it has reported Fork failed. But
>the fork most certainly has not failed! The shell command invoked has
>run, and what's more, completed successfully, producing the expected files.
> Does anyone have any ideas?
[EMAIL PROTECTED] suggested:
Can you c
Attempting to correct the problem about which I earlier posted - wherein
a system() call which apparently succeeds is perceived to have failed by
the * process, I changed code in app_system.c so that it would be more discerning...
res = system((char
*)data);
/*
if (res
eply-To: [EMAIL PROTECTED]
I saw the same thing. I think I determined that it always failed at the same
point in the macro, no matter what command was being executed. I just put
the whole cleanup process in a shell script and I execute the shell script
from the macro.
Eric
Message: 2
Date:
Using John Todd's example for recording, from his cleanup/conversion macro...
; Turn the two in/out
.wav files into a single .wav file with both channels
exten => s,3,System(/usr/local/bin/wmix ${MONITORDIR}/${CALLFILENAME}-in.wav
${\
MONITORDIR}/${CALLFILENAME}-out.wav > ${MONITORDIR}/${CALLF
I've a SIP phone (GS 100) which dials out fine through a Voicepulse
Connect account via *.
And I've got a phone number which does DID in via IAX from Voicepulse.
I want it to ring the GS phone for now.
I have this in extensions.conf:
[voicepulse-incoming]
; This context tells Asterisk what to
I observed a packet routing endless loop at:
16 host-63-108-128-153.apid.com (63.108.128.153)
This happened with traceroute from two distinct origination points. Seems
to have been resolved.
Message: 3
Date: Fri, 13 Feb 2004 20:11:44 -0500
From: John Fraizer <[EMAIL PROTECTED]>
To: [EMAIL
> Regarding codec selection, I see a minor difference between the FWD
> and the local * box test cases, but I know nothing about the
> negotiation protocol...
>
> With FWD, the OK message lists 3 Media Formats:
>
>
Bingo...GS chokes with GSM...just disallow it in your sip.conf:
disallow=all
>I am trying to muddle my way tthrough getting something - actually
>anything to work - with Asterisk. I've acquired a Grandstream phone and
>I've got * on a Red Hat 9 box. I've gotten to a point where I can see
>(via ethereal) that the phone REGISTER's successfully
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