So I'm the only person that actually enjoys reading the RFC's?
On 10/7/07, Brian West [EMAIL PROTECTED] wrote:
Telling someone to read the RFC bah.. might as well give them a blanket and
pillow because they will fall asleep. chan_sip is just ugly in every way.
/b
On Oct 7, 2007, at 9:26
On 10/10/07, Ex Vito [EMAIL PROTECTED] wrote:
Hi list,
I'm evaluating a private telephony scenario of about 20
locations - 300 phones, 50 FAX machines.
More than 1 PRI?
All other locations, small by themselves, would get SIP
phones managed by asterisk, since there is good IP
http://en.wikipedia.org/wiki/Network_File_System_(protocol)
On 10/12/07, Pepo [EMAIL PROTECTED] wrote:
Using two Asterisk connected between they, How do I can check the voicemail in
a remote system but working like *97?
I mean dont want ask the voicemail box, just the password and go to the
On 10/15/07, Doug [EMAIL PROTECTED] wrote:
Case:
1 CodeGen 4U Server Case $80
http://tinyurl.com/bnobz
http://tinyurl.com/95s2b
http://www.newegg.com/Product/Product.aspx?Item=N82E16811182566
Or:
1 Eagle Tech ET-RMAL2025-SL Beige 2U Server Case 2 External 5.25
Drive Bays
On 10/11/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Yes, see the t1e1override module parameter in wct4xxp/base.c. IIRC, it's
0xff to hard code to E1 mode, and set it to 0 for T1 mode. -1 is to use
the jumper settings.
Seems like a bad design. Why not just make it a software choice??
Is there any trick to getting T.38 fax to work with SIP? I had it
working and one day with no changes *poof* it stopped working and
hasn't worked for months. The only common factor is Asterisk 1.4.x
(always try to use the latest version) and NAT.
I've tried:
-Linksys ATA
-Grandstream ATA
://80.86.84.71/kolmiwiki/index.php/Send_Receive_Fax-T38
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas van
dem Helge
Sent: Thursday
tail /var/log/mail or /var/log/maillog
On Thu, Mar 13, 2008 at 5:04 PM, Mike Hammett [EMAIL PROTECTED] wrote:
I need to setup a small mail server on a local network. It only needs SMTP
ability as it's just so Asterisk can send out emails. The machine has
sendmail installed. My primary
1) Use RFC2833
2) Make sure all devices are properly configured
3) Try another provider.
On Thu, Mar 13, 2008 at 4:54 PM, Jarga Jallow [EMAIL PROTECTED] wrote:
Hi,
I have polycom 301 IP phones most of them especially when I call a direct
line with extensions, I cannot dial an extension.
Linksys SPA2102 does. It even has the option to auto-detect so if it
is assigned an RFC1819 address it will act as a switch and otherwise
just as a NAT router.
So does Grandstream HT496 (and I'm sure others) but it must be
manually configured.
On Fri, Mar 14, 2008 at 1:59 AM, Thermal Wetland
On Tue, Mar 11, 2008 at 9:41 AM, Louwrens Benadé [EMAIL PROTECTED] wrote:
Better reporting through a new call event logging capability in Asterisk
1.6 will allow complete tracking of events that take place during a call.
The goal, according to Fleming, is to provide more detail than
MOR PRO - Advanced Billing Solution for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas van
dem Helge
Sent: Friday, March 14, 2008 3:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
option.
On Fri, Mar 14, 2008 at 11:47 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Fri, Mar 14, 2008 at 02:32:54AM -0400, Andreas van dem Helge wrote:
Linksys SPA2102 does. It even has the option to auto-detect so if it
is assigned an RFC1819 address it will act as a switch and otherwise
SPA1001 is 1FXS only. SPA3102 is 1FXS + 1FXO... if you don't need FXO
why not get the SPA2102? It should be cheaper and you have the extra
port for future use.
On Sat, Mar 15, 2008 at 12:08 AM, Thermal Wetland
[EMAIL PROTECTED] wrote:
That is good to know.
I will be using the the device to
It's not bad in the sense of stability (well the original ones are
claimed to have overheating issues..).
But its that it lacks ANY features. The IAXy has no features at all.
Also no security, it MUST be placed behind a firewall, as the
configuration doesn't have any sort of security whatsoever.
What is your extensions.conf setup? that has alot to do with it (I
strongly suggest you use macros.) What SIP NNN code does the phone
return when DND?
On Mon, Mar 17, 2008 at 2:00 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
I am using Polycom IP600 phone. If I call a phone which has DND (do
*CLI show application Transfer
-= Info about application 'Transfer' =-
[Synopsis]
Transfer caller to remote extension
[Description]
Transfer([Tech/]dest[|options]): Requests the remote caller be transferred
to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only
an incoming
I think that's still a better idea than using a dump the caller into
meetme hack and is actually what I was going to suggest.
If you want something simpler than a queue then inject the sounds into
the moh already.
On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis [EMAIL PROTECTED] wrote:
You may be
On Sun, Apr 6, 2008 at 1:52 PM, Alex Kauffmann [EMAIL PROTECTED] wrote:
Thank you for the replies. It was my understanding that rebuilding the
kernel was necessary in 2.4 but everything needed was already included
in 2.6 series.
*HEAVILY* dependent on the distribution!
I want to 3rd this. They admitted some of their hardware runs GPL code
(Linux, IPTables, wget and more) yet refuse to provide the source code
or evidence of an alternate license agreement with the authors of the
software (which I doubt they did I just like to give people that
benefit of the
On Fri, Apr 18, 2008 at 3:53 PM, Godwin Stewart Horwich IT Services
[EMAIL PROTECTED] wrote:
On Fri, 18 Apr 2008 08:37:32 -0800, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
If you care to use ping pong balls and the atlantic ocean as your medium,
you should be able to
Does anyone have a script for manual wardialer for asterisk? not sure
if wardialer is the correct term but basically I want to call X
number say 555- through 555-0050 and be able to listen to each
call and when I hang up or press a key it will dial the next number
for me. I guess sort of
So I can't dial my own number blocks for auditing? I do this manually
right now dial 1 number, dial another on and on it gets very
tedious and sometimes you loose your place. Approx every 2 months per
number. The companies using these numbers have very specific reasons
for requiring these
Anyone have a download link for 3.0 SIP firmware?
If you are going to say ask polycom or ask your vendor don't even
waste your time posting. I've asked the Nazis and they'll probably
take 1 week.
Thanks,
Andy
___
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AFAIK Siemens ceased distribution of their Gigaset line in North
America a few years ago either you find a wholesaler that is importing
grey market items or you buy it from a distributor overseas.
On Sun, Apr 27, 2008 at 11:16 AM, Michael Graves [EMAIL PROTECTED] wrote:
On Sun, 27 Apr 2008
: Polycom 3.0
Andreas van dem Helge wrote:
Anyone have a download link for 3.0 SIP firmware?
If you are going to say ask polycom or ask your vendor don't even
waste your time posting. I've asked the Nazis and they'll probably
take 1 week.
Suggest you get a different vendor then. I
vendor forum.
-Jon
- Original Message -
From: Andreas van dem Helge [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 29, 2008 3:21:30 AM GMT -06:00 US/Canada Central
Subject: Re
Could someone please add to the documentation that Zaptel is required
for SLA to work? It becomes sort of frustrating when you read the
documentation a few times, keep on trying to get the thing to work for
a few hours only to discover there is a minor omission in the
documentation.
Is Zaptel 1.4.10 compatible with RHEL 3 (2.4.21-53.ELsmp)? Because I
can compile 1.2.20.1 just fine but 1.4 says:
echo You do not appear to have the sources for the 2.4.21-53.ELsmp
kernel installed.
You do not appear to have the sources for the 2.4.21-53.ELsmp kernel installed.
exit 1
make[1]:
=
=
===
System without Zaptel:
unisoft*CLI sla show stations
No such command 'sla show' (type 'help' for help)
On Wed, Apr 30, 2008 at 12:53 PM, Patrick
[EMAIL PROTECTED] wrote:
On Wed, 2008-04-30 at 08:56 -0700, Andreas van dem Helge wrote:
Could someone please add to the documentation
On Wed, Apr 30, 2008 at 2:47 PM, Mik Cheez [EMAIL PROTECTED] wrote:
Have you tried kernel-smp-devel?
Andreas van dem Helge wrote:
Is Zaptel 1.4.10 compatible with RHEL 3 (2.4.21-53.ELsmp)? Because I
can compile 1.2.20.1 just fine but 1.4 says:
echo You do not appear to have
in the CLI you can issue the command
zap show status
e.g.:
pbxserver-doral*CLI zap show status
Description Alarms IRQbpviol CRC4
Wildcard X101P Board 1 RED0 0 0
In this case the phone line is unplugged and
/.build is missing and I get errors trying
to do 'make cloneconfig'
On Wed, Apr 30, 2008 at 4:57 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Apr 30, 2008 at 09:21:37PM +0300, Tzafrir Cohen wrote:
On Wed, Apr 30, 2008 at 02:00:57PM -0400, Andreas van dem Helge wrote:
Is Zaptel 1.4.10
.build file is missing in the kernel-source package. Solutions is:
Once you have the appropriate kernel sources installed you will
need to configure them. Execute the following commands:
cd /lib/modules/`uname -r`/build
make mrproper
Some of the polycom phones support this with a specific firmware and
Plantronics headset.
Read the polycom SIP release notes/changelog for details
On Mon, May 5, 2008 at 5:29 AM, Louis-David Mitterrand
[EMAIL PROTECTED] wrote:
Hello and sorry for the OT,
Is it possible for a wireless
I totally agree. Someone filed a bugreport for this? Also asterisk
init script should be installed by default too.
I am going to give Cesar's instructions a try (sans removing /bin/sh)
and hope it works!
On Tue, May 6, 2008 at 3:24 AM, Stelios Koroneos
[EMAIL PROTECTED] wrote:
In general, if
see voicemail.conf.sample all the options you need are documented there.
maxmsg delete
On Wed, May 7, 2008 at 12:49 PM, Steve Johnson [EMAIL PROTECTED] wrote:
Hi everyone,
We have a particular user on our Asterisk 1.4.x system who always
listens to his voicemail messages via email.
-
The call is still going to show up as the codec with which the voice
segment was established.
Have you viewed the SIP debug messages and confirmed that T.38 is not
being used?
FWIW the device that is receiving the T.38 fax (generally callee)
should be issuing the T.38 re-invite, so you might
In menuconfig did you select the g729 music on hold?
Or you can try this one:
http://app5.netjdn.com/~joako/sounds/SampleAudioSource.g729.wav
(remove the .wav extension)
On Thu, May 8, 2008 at 5:46 PM, Nitesh Divecha [EMAIL PROTECTED] wrote:
Hello All,
Recently, I build three Asterisk 1.4 box
I think it's all personal preference I'd never recommend anyone
use ubuntu for anything, honestly.
SLES is my #1 pick with CentOS / PNAELV being a close second...
problem with Cent is there's not central administration like there is
in SuSE (YaST2... it's so simple! gotta setup a network no
Oh, and FWIW a Cisco uses PNAELV as the basis for one of it's most
popular voice products.
http://www.bouncethem.com/5455
On Fri, May 9, 2008 at 11:19 AM, equis software [EMAIL PROTECTED] wrote:
Hi, I allways use Gentoo y my Asterisk servers and work well, but what do
you think about to use
Anyone have shared lines (sla.conf) working with Polycom phones? Also,
has anyone figured out if its possible to do 1 button call park with
the softkeys?
___
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asterisk-users mailing
I think there are quite a few aspects to the issue. I agree I've used
the X101p cards which really are a Windmodem with a resistor removed
and I had nothing but echo problems but then again I could have tried
harder.
1) It was an early digium product. I think the Sangoma cards and the
newer
Anyone have shared lines (sla.conf) working with Polycom phones? Also,
has anyone figured out if its possible to do 1 button call park with
the softkeys?
___
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asterisk-users mailing
I've had the same issues with T-Mobile. Thanks for pointing out the
exact cause. I did speak to someone in their back-end offices and they
did resolve it for a short time, then it reverted back for a bit and
started working again. Now it seems to work 90% of the time.
On Sun, May 11, 2008 at
srv04*CLI show application Dial
srv04*CLI
-= Info about application 'Dial' =-
[Synopsis]
Place a call and connect to the current channel
*SNIP*
p- This option enables screening mode. This is basically Privacy mode
without memory.
P([x]) - Enable privacy mode. Use 'x' as
A quality 3U chassis will mount the cards parallel to the mainboard
with the use of a riser card, just as a 1U chassis does.
If you are intent on sourcing the components yourself may I suggest a
Tyan or Supermicro barebones server? I think that is the best
solution for integration in these sort
This will work:
http://www.newegg.com/Product/Product.aspx?Item=N82E16899705001
I assume you have devised a way to power the USB to serial adapters
from the PC power supply.
FWIW I think your system is inefficient but maybe you do need 750gb
per each installation. Each to his own.
On Tue,
On PRI SetCallingPres works fine it should work with ISDN because its
the same signaling.
-= Info about application 'SetCallerPres' =-
[Synopsis]
Set CallerID Presentation
[Description]
SetCallerPres(presentation): Set Caller*ID presentation on a call.
Valid presentations are:
A posting to the correct mailing list?
Or at least a post with the details of the issue? What OS? Can you
play these same .gsm files in any media player your OS might have?
On Thu, May 15, 2008 at 7:26 PM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I have a lot of recordings from asterisk in
The docs as far as I can tell are not correct. E.g. Zaptel is required
(because it seems that it uses MeetMe) but none of that is documented.
So yes please do see if you can make the feature work and please post
a working example config for a Polycom phone.
On Fri, Nov 30, 2007 at 8:10 PM,
Yea... tf.voipmich.com went down ~ 1 week ago when you call through it
the calls just ring forever.
Some issue seems to happen with tf.voipmich.com at least once a year
and it always takes a long time to fix.
tollfreegateway.com seems to be working..
On Sat, May 17, 2008 at 1:38 PM, Adrian
So why don't you just disable reinvite?
Using 1.4.15 here with no issues with MixMonitor. Then again I've
*ALWAYS* disabled reinvite because it never works for me.
On Mon, May 19, 2008 at 4:33 PM, Trevor Peirce [EMAIL PROTECTED] wrote:
Hello,
I'm wondering if anyone else has been observing
Cisco gateway with T.38 support. That's the only real way to do faxing
through asterisk. I think a VG200 with newer firmware will support SIP
+ T.38 but don't buy on my suggestion because I've never used that
device outside call manager configuration.
Or see if your VoIP provider supports T.38
On Tue, May 20, 2008 at 12:23 AM, Lee Howard [EMAIL PROTECTED] wrote:
Andreas van dem Helge wrote:
Cisco gateway with T.38 support. That's the only real way to do faxing
through asterisk.
Although this statement has marginally more truth to it given the
SIP-only context that the original
Here's something similar for Linux: http://sourceforge.net/projects/vgps/
Note I do not support nor endorse Voicepulse. Actually let's get it
straight, I detest Voicepulse.
On 5/3/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Mats Karlsson wrote:
Take a look here:
On 5/4/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
Thanks John,
How can I change my conf to NETWORK? Where can I find this information?
#signalling = pri_cpe
signalling = pri_net
___
--Bandwidth and Colocation provided by Easynews.com --
On 5/5/07, dave cantera [EMAIL PROTECTED] wrote:
nitesh,
you are correct. you need 1.4.x...
daveC
It is supposed to have H.263, which does work with 1.2.x:
[general]
...
videosupport=yes
..
[video-enabled-sip-phone]
...
canreinvite=no
disallow=all
allow=ulaw
allow=h263
...
, Andreas van dem Helge wrote:
On 5/4/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
Thanks John,
How can I change my conf to NETWORK? Where can I find this information?
#signalling = pri_cpe
signalling = pri_net
nitpicking:
;signalling = pri_cpe
signalling = pri_net
(The comment
On 4/27/07, Steve Murphy [EMAIL PROTECTED] wrote:
I'm the guilty party. I've been trying to fix several CDR bugs,
involving stuff like missing times, missing changes in state (like
NO_ANSWER when the call was ANSWERED), etc.
Now that we are talking about CDRs, I must ask: in 1.2.x if the CDR
On 5/13/07, Jon Pounder [EMAIL PROTECTED] wrote:
what exactly was the charge ?
- trespass - no its public land for the most part this stuff is on so
that doesn't apply
- vandalism/mischief - if no other customer was impacted I don't see
how this charge would stick since there is no measurable
On 5/16/07, Robert Lister [EMAIL PROTECTED] wrote:
I was wondering if it is possible (in 1.2.x) to get the SIP response code
back after doing Dial().
Dial() seems to treat most call-setup problems as dialstatus CONGESTION, and
some are NOANSWER, but I want to know the SIP response code, so I
I've had the opposite problem. Press mute while the call is still
ringing and it will say MUTE on the display but the microphone is
not muted. It was very embarrassing to discover this bug.
On Wed, Feb 13, 2008 at 2:03 AM, Thomas Kenyon
[EMAIL PROTECTED] wrote:
Lutgring, Sam wrote:
I take it
What are you trying to accomplish exactly? They sell SIP overhead
speakers or you can use a SIP phone with an adapter on the 2.5mm
headset jack.
On Wed, Feb 20, 2008 at 2:44 PM, Jerry Geis [EMAIL PROTECTED] wrote:
I am looking for an ATA like device but instead of VOIP to analog phone
I
Does anyone have a script that will emulate a normal numeric pager but
send the number to an email address? Also anyone happen to have the
traditional tones used in North America?
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