incoming call handling peer is at the very end.
Pretty wacky.
I am hopefully back on the road though with working caller ID as well.
Marty
On Nov 14, 2009, at 11:10 AM, Martin Joseph wrote:
Ok I am replying to myself, because I still don't have this figured
out,, but I think I have more
Ok I am replying to myself, because I still don't have this figured
out,, but I think I have more info.
On Nov 5, 2009, at 8:57 PM, Martin Joseph wrote:
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now
On Nov 6, 2009, at 5:14 AM, John A. Sullivan III wrote:
On Thu, 2009-11-05 at 20:57 -0800, Martin Joseph wrote:
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now, with pretty good results.
Recently I added
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now, with pretty good results.
Recently I added a callerid service to my landline (qwest).
I am using the audiocodes MP114 2fxo/2fxs gateway, which is an
outstanding
Hello again gurus.
I have been using Asterisk with great results going on a couple of
years now.
My primary box is running asterisk 1.42 built from a tar ball on Mac
OSX 10.4.9.
I have a very odd issue that I cannot seem to nail down, which is
related to my Nokia E60 SIP phone.
I use
On May 14, 2007, at 12:34 PM, Tim Panton wrote:
On 14 May 2007, at 17:50, Martin Joseph wrote:
Hello again gurus.
I have been using Asterisk with great results going on a couple of
years now.
My primary box is running asterisk 1.42 built from a tar ball on
Mac OSX 10.4.9.
I have
On 2007-03-26 01:46:40 -0700, Salvatore Giudice
[EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
I opened up a ticket with them, but I'm not holding my breath. I think it's
time to start moving my DID's before the inbound stops working.
That seems like it was probably
On 2007-04-17 00:53:56 -0700, Dinesh Nair [EMAIL PROTECTED] said:
On Mon, 16 Apr 2007 20:14:40 -0700, Martin Joseph wrote:
The phone no longer registers with asterisk, although it displays the
little icon as though it has, and it doesn't even seem to try to pass
calls to asterisk...
So, I
Just a warning for you all that are using Nokia series E phones for SIP
function.
I updated my phones firmware today using the Nokia Updater, and now
the SIP functionality, which previously worked pretty well is
completely broken.
The phone no longer registers with asterisk, although it
On 2007-03-24 01:53:16 -0700, Edoardo Serra
[EMAIL PROTECTED] said:
Hi Francois,
[EMAIL PROTECTED] ha scritto:
Hi men,
I have already encountered some issue like this with few switches (very
known great brand) which doesn't like VoIP traffic !
I also have switches of a very known
On 2007-03-23 14:37:18 -0700, Tom Lynn [EMAIL PROTECTED] said:
Now I know where they've been spending my remaining balance...
I still use Sellvoip as my primary terminator, and have found the call
quality to be superior to any other ITSP from my location (Seattle).
I agree completely
On 2007-02-22 04:22:20 -0800, Frederico Madeira [EMAIL PROTECTED] said:
Hi guys,
My asterisk is show me some errors on line registration.
This message appear on console: Request to schedule in the past?!?!
What it mean ?
Thanks.
I see this message all the time on my lowely powerPC mac
On 2007-02-14 22:12:23 -0800, jameson asterisk [EMAIL PROTECTED] said:
I'm currently looking to deploy an Asterisk server using an FXO media
gateway to connect to the PSTN and was looking for any user experiences that
may aid in selecting a gateway. Specifically i'm looking for a 4-port model
On 2007-01-28 08:37:43 -0800, Eric Germann [EMAIL PROTECTED] said:
We LOVE Teliax. We're on a Time Warner business class fiber connection and
avg 25ms latency from Ohio to Denver CO.
With that connection I would love Teliax also.
Marty
___
On 2007-01-17 10:29:43 -0800, Yelson Vivas [EMAIL PROTECTED] said:
Hi Guys
I'm conecting 2 astersk servers using this arquitecture
(Ext softphone)==sip==(asterisk 1)iax2 trunk(asterisk 2)
===alaw==(pstn)
If i call from the Ext to the asterisk 2 the sound is perfect, but if
i call
On 2007-01-14 22:01:44 -0800, Tomer Horn [EMAIL PROTECTED] said:
Hello,
I am looking to purchase a new quad-band cellphone and I'm looking for
one with WiFi and enough CPU power for stable SIP calls. I was
wondering if anyone here can share his experience and recommend on a
good cellphone.
On 2007-01-07 01:23:22 -0800, Mark Coccimiglio [EMAIL PROTECTED] said:
Marty,
Where are you paying $1000 for a 1600 series Cisco? I can get you
20% off that price on any quantity (note: Sarcasam). Its not the
1990's anymore. You can get them on eBay ($50-150) for only slightly
more
On 2007-01-04 09:56:58 -0800, Mike [EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
Hi,
I'm looking for opinions on the best value router to use for home offices.
It should work for a scenario in which there are 3 computers and 2 SIP
phones, handling QoS so that the
On 2007-01-05 09:40:18 -0800, Biju [EMAIL PROTECTED] said:
hi,
i am using nokia e61 . we have an asterisk server and i want to use my
nokia phone to register with asterisk server .
anybody can help me to do this.
Try Google, it works.
Marty
On 2007-01-06 00:48:11 -0800, Mark Coccimiglio [EMAIL PROTECTED] said:
Mike
I'm using a Cisco 1605R [running IOS 12.3(5a)] small office router with
Fair-Weight queueing enabled. Works great. The nice thing about
Fair-Weight queueing is that it dynamically adapts to lower the
priority of
On 2006-12-21 13:29:47 -0800, cb [EMAIL PROTECTED] said:
Has anyone used either the 8 port or 4 port FXO device from
Grandstream? (GXW-4108 or 4104).
They seem to be the lowest cost multi port FXO devices that I can
find, so I'm getting ready to buy the 8 port version. I just want to
see
On 2006-12-31 00:52:27 -0800, mitcheloc [EMAIL PROTECTED] said:
Those wifi phones are neat but I'd rather not carry around two
devices, does anyone know of any good dual-mode GSM/SIP phones?
I'm using a T-Mobile MDA right now and it is way too slow.
Apparently the Nokia e61 has a built in SIP
On 2006-12-24 00:35:06 -0800, Martin Joseph [EMAIL PROTECTED] said:
I have a spiffy new gateway which seems quite promising.
It's the Audiocodes MP114 FXS_FXO (2 of each).
I have got it configured and working reasonably well, but have a couple
of issues.
1) Asterisk 1.2.13 voicemail seems
On 2006-12-20 05:18:08 -0800, Chris Blunt [EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
Hi List
I need a quality US 800 DID over IAX for my Asterisk server, preferably one
that doesn't cost the earth.
Any suggestions please?
Depends a lot on your geographical
Hi,
I have a spiffy new gateway which seems quite promising.
It's the Audiocodes MP114 FXS_FXO (2 of each).
I have got it configured and working reasonably well, but have a couple
of issues.
1) Asterisk 1.2.13 voicemail seems to be hung up on by the gateway
after 10 seconds. This isn't
On 2006-12-10 04:13:02 -0800, RR [EMAIL PROTECTED] said:
Hello,
does anyone else have a problem with Asterisk crashing right after a
valid password/PIN is entered when trying to access voicemail in the
1.4b3 version? Not sure if this is anything to do with realtime per
se but I keep getting
On 2006-12-01 09:45:00 -0800, Martin Joseph [EMAIL PROTECTED] said:
snip
So, I would like to purchase another PSTN gateway which WORKS WELL
with asterisk. I need it to hook up via ethernet, since my platform of
choice (mac OSX) has no PCI card support. I only have one PSTN line,
and already
personally (in particular the FXO and
and asterisk)?
Thanks again,
Marty
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Martin Joseph
Envoyé : lundi 4 décembre 2006 20:47
À : asterisk-users@lists.digium.com
Objet
On 2006-12-04 04:10:36 -0800, Giedrius Augys [EMAIL PROTECTED] said:
Hi,
I am testing Nokia E60 with Asterisk. And I noticed that if another side
is busy, nokia is still calling (I hear alerting), it do not show that
another side is busy. Maybe somebody has noticed the same problem too adnd
Ok,
I am back from my thanksgiving holiday, and I find there was a big
snow storm here in Seattle. Apparently during the storm there where
multiple brown out/black outs.
I have struggled since day one to get a high quality PSTN gateway
configured with my very long loop and Mac based
On 2006-11-15 18:30:38 -0800, Lucas Barbuto [EMAIL PROTECTED] said:
Hi all,
Originally tried to post this without being subscribed, apologies if
the list gets this twice.
One of my users has a problem with many of his calls via my Asterisk™
server. He describes the problem as having the
[EMAIL PROTECTED] said:
I'm trying to set up the Music on Hold feature.
However, when I place a call the moh starts and stops
immediately and as a result I dont hear the audio.
On 2006-11-13 00:14:40 -0800, zen Perry [EMAIL PROTECTED] said:
Mac OS X, Asterisk 1.4 beta
Yeah, I am
On 2006-11-12 14:48:13 -0800, joe a. ([EMAIL PROTECTED])
[EMAIL PROTECTED] said:
Experiencing one way audio using IAX2.
I did see some other posts on this, and see there may be some internal
issues with asterisk and one way audio. Can this be a widespread
problem? So many seem to be
On 2006-11-12 23:08:05 -0800, zen Perry [EMAIL PROTECTED] said:
I'm trying to set up the Music on Hold feature.
However, when I place a call the moh starts and stops
immediately and as a result I dont hear the audio.
-- Started music on hold, class 'default', on
channel 'SIP/XXX'
--
On 2006-11-10 15:48:23 -0800, Andrew Joakimsen [EMAIL PROTECTED] said:
I am surprised that you have had good success perhaps you haven't done
proper testing?
I see you are skeptical...
I am using the Nokia e60, which also has no problems on the asterisk
side. The phone could use some
On 2006-11-10 09:10:30 -0800, Mario François Jauvin
[EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
I have had no success in getting the voicemail working on Asterisk
1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried
with or without ztdummy device,
On 2006-11-08 14:40:09 -0800, Ken Williams
[EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
After about one weeks time I've gone from no VoIP to a completely
configured system for two of our offices to be able to page/communicate
interoffice as well as handle existing
On 2006-11-08 06:26:45 -0800, [EMAIL PROTECTED] said:
Hello,
I need (some) help in configuring PAP2.
Try looking in sip.conf
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Tom Vile wrote:
That probably because you are using Webmin. Just change the port
Webmin listens on instead, I use 9000.
On 11/6/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
I'll keep that in mind for future. I read about using 10001 as start
port on Nerd Vittles website.
Is there some
On 2006-11-06 13:16:50 -0800, Christian [EMAIL PROTECTED] said:
Hi all,
DO my messages come through to the list? I have had some problems wiht
my email client here.
Looks like your spell checker has issues also...
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On 2006-11-02 07:51:15 -0800, Noc Phibee [EMAIL PROTECTED] said:
Hi
anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ?
Actually my HandyTone 488 are connected to:
wan port to my lan
line FXO port are connected to my local analogic line
i want that when a call in
On 2006-11-02 07:34:15 -0800, mail-lists [EMAIL PROTECTED] said:
snip
My question is this: How do huge voip companies like vonage handle
bandwidth. I'm pretty sure that they have to have sufficient bandwidth
available for X numbers of simultaneous calls, in other words ALL VOIP
traffic runs
On 2006-11-02 20:57:28 -0800, James Harper
[EMAIL PROTECTED] said:
We have a 100 number indial range and every so often get fax calls on
our voice numbers (our fax number isn't in the 100 number range). If you
just hang up the sending fax will often try a few times before finally
giving up.
On 2006-11-02 05:11:28 -0800, Florian Hars [EMAIL PROTECTED] said:
I've long since given up registering to bug trackers, there are far too
many of them, and I don't want to remember a username/password pair for
every program I use.
Don't complain about bugs then!
I am testing 1.4 branch on OSX (10.4.8) and although it's running and
passing calls ok, I am still not able to connect using asterisk -r.
When I do open a CLI using asterisk -r, it appears to start up
normally, but then is non responsive to commands (exit works though?).
I am currently
On 2006-11-01 08:28:28 -0800, Jason Walker [EMAIL PROTECTED] said:
Ok sorry for not being specific. I am having a problem when people
outside call in to my number which terminates at VoicePluse then The
send IAX to me and I do not get any tones. People press buttons but it
just goes to the
On 2006-11-01 10:42:12 -0800, Joshua Colp [EMAIL PROTECTED] said:
Martin Joseph wrote:
Good news!
I did an SVN update to my 1.4 branch again today, and 1.4-r46154 seems
to have resolved the asterisk hogging the whole CPU issue.
I still can't use the regular console though (asterisk -r
On 2006-11-01 09:09:26 -0800, Martin Joseph [EMAIL PROTECTED] said:
I am testing 1.4 branch on OSX (10.4.8) and although it's running and
passing calls ok, I am still not able to connect using asterisk -r.
When I do open a CLI using asterisk -r, it appears to start up
normally
On 2006-10-31 17:29:47 -0800, Brad Templeton [EMAIL PROTECTED] said:
I've been losing patience with my current provider, a small company
called Sellvoip. Their termination is good, and they are
asterisk based, but they are understaffed and have no concept
of customer service. So I'm
On 2006-10-29 01:35:46 -0800, Alberto Pastore [EMAIL PROTECTED] said:
Martin Joseph wrote:
I think it's cleary true that wiring WIFI infrastructure is easier and
more reliable then WDS.
On the other hand, I have been running my little network with WDS for
over three weeks now, and it has
On 2006-10-28 03:51:57 -0700, Alberto Pastore [EMAIL PROTECTED] said:
Andrew Joakimsen ha scritto:
Are you using WDS? While it won't totally fix every issue, I've found
in my trials that turning off WDS and making sure all the AP were
connected to the same wired network was way more reliable,
On 2006-10-27 11:55:14 -0700, Andrew Joakimsen [EMAIL PROTECTED] said:
Are you using WDS? While it won't totally fix every issue, I've found in my
trials that turning off WDS and making sure all the AP were connected to the
same wired network was way more reliable, no more random
On 2006-10-28 07:55:43 -0700, Dean Collins [EMAIL PROTECTED] said:
Alberto, you should have bought a dect solution, the dect technology is
far better at swapping between cells.
Wifi is still a little immature at this time.
Not if correctly configured. This is simply wrong.
Marty
On 2006-10-26 23:02:40 -0700, Stefan Agethen
[EMAIL PROTECTED] said:
Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update
of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no
On 2006-10-27 09:59:10 -0700, David Parcerisa [EMAIL PROTECTED] said:
Hello;
I have a problem with voicemail and my asterisk 1.2.1 on a OS X Mac
Pro intel box.
When I try to record a message from an incoming call or a greeting
message from internal phone using voicemail, It's like something
On 2006-10-27 08:49:44 -0700, Alberto Pastore [EMAIL PROTECTED] said:
Hello everyone.
I know it's a little bit off-topic, but I was just wondering...
Has anyone ever had any experience with asterisk,
a wi-fi meshed lan (with more than one access point)
and wi-fi sip phones?
I don't think I
On 2006-10-25 22:33:47 -0700, John Marvin [EMAIL PROTECTED] said:
Martin Joseph wrote:
Transcoding is a bigger hit then mixing as i understand it.
If all the conference members are using ulaw for example, then having
the playback material encoded in ulaw is the big winner
On 2006-10-26 09:21:20 -0700, Dave Cotton [EMAIL PROTECTED] said:
On Thu, 2006-10-26 at 17:43 +0200, Pavel Jezek wrote:
with SIP qualify, I can specify, what time in delay I will accept,
with sip and setting qualify=3000 I can circumvent this anoying
messages (bacause delay in reply is about
On 2006-10-26 03:18:20 -0700, Stefan Agethen
[EMAIL PROTECTED] said:
Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update
of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no echo
Good news!
I did an SVN update to my 1.4 branch again today, and 1.4-r46154 seems
to have resolved the asterisk hogging the whole CPU issue.
I still can't use the regular console though (asterisk -r) as that is
unresponsive.
Using asterisk -c to start it , works and gives me a color CLI
On 2006-10-24 17:25:37 -0700, Steve Underwood [EMAIL PROTECTED] said:
The development of Asterisk has now degraded to the point where I will
no longer contribute anything to it.
I am not interested in a flame war, but would love to here a more
explicit explanation for what is occurring
On 2006-10-24 13:04:02 -0700, Matt [EMAIL PROTECTED] said:
Hi,
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.
The customer is connected via IAX2 to our softswitch.
On the customer's end I
On 2006-10-24 06:44:01 -0700, Wildheart
[EMAIL PROTECTED] said:
Hi,
Does anyone know a what to use a different codec for calls which are
handset to handset (eg, G711) then when we have calls to the out side
world (via an asterisk server) to use a different codec(eg, G729)?
The idea
On 2006-10-25 08:14:43 -0700, Noah Miller [EMAIL PROTECTED] said:
Hi Matt -
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.
On the customer's end I have the following config in iax.conf:
On 2006-10-24 06:44:01 -0700, Wildheart
[EMAIL PROTECTED] said:
Hi,
Does anyone know a what to use a different codec for calls which a
re
handset to handset (eg, G711) then when we have calls to the out side
world (via an asterisk server) to use a different codec(eg, G729)?
snip
I
On 2006-10-25 08:14:56 -0700, Matthew Rubenstein [EMAIL PROTECTED] said:
What's the native soundfile format for SIP?
??? I think you might need to do some research (the above is a nonsense
question I think).
Any idea which soundfile
takes the least CPU for mixing together in
On 2006-10-25 15:00:52 -0700, Andrew Joakimsen [EMAIL PROTECTED] said:
Also the Nokia E60 and E61 are hybird GSM/WiFi phones, when you have WiFi
coverage your calls will go over that technology and when you aren't its
just a regular mobile. Works great if you only want to purchase one device,
On 2006-10-24 10:32:09 -0700, Henry.L.Coleman
[EMAIL PROTECTED] said:
Hi all, the lists seems to be littered with disconnect problems using
various equipment (TDM 400,Linksys etc etc.)
My question is very simple and could make for good solution to Asterisk
users.
Since * can detect various
On 2006-10-21 22:20:51 -0700, Joshua Colp [EMAIL PROTECTED] said:
Okay folks, give the latest 1.4 branch a try. I spent some time this
morning isolating the issue and think I have it.
OK! Thanks Josh, that builds and seems to work a bit, but it's
easting my whole CPU... Any ideas on how
On 2006-10-22 09:16:04 -0700, Tim Panton [EMAIL PROTECTED] said:
On 22 Oct 2006, at 07:02, Martin Joseph wrote:
On 2006-10-21 22:20:51 -0700, Joshua Colp [EMAIL PROTECTED] said:
Okay folks, give the latest 1.4 branch a try. I spent some time this
morning isolating the issue and think I
On 2006-10-22 07:14:46 -0700, Joel Lansden [EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
Greetings list,
=20
I have an older Dell Poweredge server running Asterisk 1.2.13. I have
installed 5 licenses for G.729 from Digium. I have 5 SIP trunks through
a US provider.
On 2006-10-22 20:58:46 -0700, Avi Miller [EMAIL PROTECTED] said:
On 23/10/2006, at 10:13 AM, Joseph wrote:
I'm trying to log-in externally (from PSTN line) to check my
voice-mail so I created context to authenticate log-in
Just create an inbound route to VoiceMailMain(). Then, press *
I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight.
I took the additional step of nuking my modules directory first...
When I used the command asterisk -v to start asterisk, it seemed to go
along and get to the point where asterisk is running(ie Asterisk Ready).
At that point
On 2006-10-21 05:09:33 -0700, Tim Panton [EMAIL PROTECTED] said:
On 21 Oct 2006, at 09:58, Martin Joseph wrote:
I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight.
I took the additional step of nuking my modules directory first...
When I used the command asterisk -v to start
On 2006-10-21 11:50:37 -0700, Joshua Colp [EMAIL PROTECTED] said:
Tim Panton wrote:
On 21 Oct 2006, at 09:58, Martin Joseph wrote:
I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight.
I took the additional step of nuking my modules directory first...
When I used the command
On 2006-10-19 20:30:03 -0700, Martin Joseph [EMAIL PROTECTED] said:
On 2006-10-18 12:34:43 -0700, [EMAIL PROTECTED] said:
Hello, well, I need to configure two asterisk box like SIP trunks to se
nd sip
calls from one asterisk to the other and visceversa. So How I setup con
fi g
files
On 2006-10-19 08:51:01 -0700, Dustin Wenz [EMAIL PROTECTED] said:
I just built 1.4.0 beta 3 on OS X 10.4.8, and it went pretty smoothly.
I didn't need to install wget.
Asterisk starts and runs with 0% CPU. The CLI also works, but hangs if
I try to tab-complete commands. However, that might
On 2006-10-19 09:30:14 -0700, Todd- Asterisk
[EMAIL PROTECTED] said:
I'm a Certified Apple Sys Admin - lots of experience with Macs and Mac
servers. However, when setting up an asterisk server, I'm still
thinking a Dell box with linux is the best direction - to get the full
reliability
On 2006-10-18 12:34:43 -0700, [EMAIL PROTECTED] said:
Hello, well, I need to configure two asterisk box like SIP trunks to send sip
calls from one asterisk to the other and visceversa. So How I setup confi g
files to get this working?.Thanks.
You can do it via IAX2, there was a recipe posted
On 2006-10-17 14:19:00 -0700, Daniel Salama [EMAIL PROTECTED] said:
You can get wget for OSX from DarwinPorts (http://wget.darwinports.com/)
Ok, I bit the bullet and build wget.
This allows me to build 1.4 branch, which does the same thing as 1.40b2.
It starts up, consumes as much CPU as
On 2006-10-16 20:54:09 -0700, Mike Lynchfield [EMAIL PROTECTED] said:
reboots are wise
No, they are foolish...
snip
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On 2006-10-16 17:10:49 -0700, Lacy Moore - Aspendora
[EMAIL PROTECTED] said:
So I was wondering is there a way to make this happen in asterisk??
Depending on where you are located, you might want to allow emergency calls
to go through. The bloodsuckers, I mean attorneys, here in the US
On 2006-10-16 03:22:47 -0700, Tim Panton [EMAIL PROTECTED] said:
On 16 Oct 2006, at 09:09, Martin Joseph wrote:
On 2006-10-15 23:50:34 -0700, Tim Panton [EMAIL PROTECTED] said:
On 16 Oct 2006, at 07:15, Martin Joseph wrote:
On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said
I am interested in using the Sipura 901 as a home phone.
Does anyone have experience with this unit? Positives, negatives,
opinions welcomed.
Thanks in advance,
Marty
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On 2006-10-17 01:06:25 -0700, Tzafrir Cohen [EMAIL PROTECTED] said:
On Tue, Oct 17, 2006 at 12:57:46AM -0700, Martin Joseph wrote:
SVN Trunk doesn't currently build on OSX (10.4.8).
If you're in for stability now, try branches/1.4 and *not* trunk.
This will eventually become beta3, rc
On 2006-10-17 11:12:27 -0700, Jack Morgan [EMAIL PROTECTED] said:
All,
I'm not able to play background files since this morning. I'm seeing
this error message in the logs:
[Oct 17 10:23:56] WARNING[4572] file.c: File
custom/asterisk-prospectus_IVR-main-day does not exist in any format
[Oct
On 2006-10-17 09:00:51 -0700, Bjoern Metzdorf
[EMAIL PROTECTED] said:
I run into that from time to time for this business account we have
where channels were staying open for a long time so I made a script run
from cron to hang up any extension over X amount of time:
/usr/sbin/asterisk -rx
it cranking some calls
out on OSX soon.
Marty
On Oct 17, 2006, at 4:13 PM, Martin Joseph wrote:
On 2006-10-17 01:06:25 -0700, Tzafrir Cohen [EMAIL PROTECTED] said:
On Tue, Oct 17, 2006 at 12:57:46AM -0700, Martin Joseph wrote:
SVN Trunk doesn't currently build on OSX (10.4.8).
If you're
On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said:
On 11 Oct 2006, at 19:35, Dean Collins wrote:
Lol - use a real PC maybe :P
Nah, that would be dull.
In some ways the mac intel is nearer to a 'normal PC'
(whatever that is) than the systems I normally run asterisk on
- a
On 2006-10-15 23:50:34 -0700, Tim Panton [EMAIL PROTECTED] said:
On 16 Oct 2006, at 07:15, Martin Joseph wrote:
On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said:
On 11 Oct 2006, at 19:35, Dean Collins wrote:
Lol - use a real PC maybe :P
Nah, that would be dull.
In some
On 2006-10-14 20:00:30 -0700, Julian J. M. [EMAIL PROTECTED] said:
Hi,
I've finally given up on trying to fax over my Digium TDM400 card.
I've found that fax over VoIP is quite more reliable (at least I can
receive the faxes).
My ITSP supports G729 and alaw/ulaw. As I won't be receiving faxes
On 2006-10-14 13:15:55 -0700, Benny Amorsen [EMAIL PROTECTED] said:
MJ == Martin Joseph [EMAIL PROTECTED] writes:
MJ I added the rtptimeout=60 to my general section in sip.conf, and
MJ now when the e60 goes out of wifi range, 61 seconds later, my
MJ channels are clear! Sweet.
Does this work
On 2006-10-10 23:14:45 -0700, Martin Joseph [EMAIL PROTECTED] said:
On 2006-10-10 20:25:44 -0700, Nic Bellamy [EMAIL PROTECTED] said :
I am seeing occasional stuck SIP channels that seem to occur when the
fricking Nokia E60 drifts out of WIFI range in the midst of a call.
snipI wonder
On 2006-10-11 03:22:00 -0700, Thomas Kenyon [EMAIL PROTECTED] said:
I have been seeing this problem for a long time and it occurs in
1.4.0b2 (as well as 1.2.0-1.2.12.1).
If the internet connection is lost and I have SIP services that require
me to register, any SIP devices attached to the
On 2006-08-21 02:44:55 -0700, Benny Amorsen [EMAIL PROTECTED] said:
MR == Matt Riddell (NZ) [EMAIL PROTECTED] writes:
MR And so you're thinking it would be better to run several hundred
MR Asterisk instances?!
Why not? As long as you stay away from the things that need zap
timing, asterisk
On 2006-10-14 07:36:51 -0700, Matt [EMAIL PROTECTED] said:
Contact them again... they have always been very good... I'm chocking
this up to the snow storm.
Yes, might still be too early, I see over 200K still without power in
there neck of the woods (Buffalo, NY).
Massive tree damage
On 2006-10-10 20:25:44 -0700, Nic Bellamy [EMAIL PROTECTED] said:
On 2006-10-08 21:28:08 -0700, Nic Bellamy [EMAIL PROTECTED] said:
I am seeing occasional stuck SIP channels that seem to occur when the
fricking Nokia E60 drifts out of WIFI range in the midst of a call.
snipI wonder if there
On 2006-10-10 18:12:23 -0700, hugolivude [EMAIL PROTECTED] said:
An Internet browser uses port 80. I might have two or more behind a
NAT both using port 80. Isn't that the same thing?
Remember that the browser INITIATES all activity on the port 80
transfers. There is no data coming in out
On 2006-10-09 22:05:06 -0700, Joseph [EMAIL PROTECTED] said:
On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote:
Anyone using the echo cancelation cards from digium? We are using the
single span T1 card with out echo cancel and I was curious if it was
worth the money.
I'm running
On 2006-10-09 15:53:36 -0700, Brandon Galbraith
[EMAIL PROTECTED] said:
Does anyone know of any ATA devices (Linksys, Dlink, Cisco, etc) that will
fail over to POTS for an emergency call? I'd like to route any call except a
911 call over SIP or IAX, but any 911 call should be routed out over
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