Hi,
Two things come to mind,
(1) being that you don't have the TE110P card jumped for an E1.
(2) UDEV isn't creating the devices fast enough for the driver load.
My guess is it's UDEV. You can test this theory by creating a startup
script that loads the modules, put a sleep statement
Should be able to edit the following lines in
/etc/asterisk/voicemail.conf
; Who the e-mail notification should appear to come from
serveremail=asterisk
;[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Suber
Sent: Wednesday, 6 June 2007 2:20
Hi,
Easy to check if the problem is udev:
ls -l /sys/class/zaptel
there are lots of subdirectories in there, at least after a rmmod
modprobe. I can reboot later to see how it looks like before that.
If there are files there and not under /dev/zap, udev is to blame.
in /dev/zap there
On Thu, Jun 14, 2007 at 08:10:33AM +0200, Sebastian Reitenbach wrote:
Hi,
Two things come to mind,
(1) being that you don't have the TE110P card jumped for an E1.
(2) UDEV isn't creating the devices fast enough for the driver load.
My guess is it's UDEV. You can test this
On 6/14/07, George Williams [EMAIL PROTECTED] wrote:
Thank you both for your expert responses to my question about asterisk
testing on the asterisk newsgroup.
I'm taking my follow up questions off-line...
1) SIPP looks like just what I need for SIP testing, thanx. You also
mentioned dialplan
Set your core debug level to greater than 2
SET DEBUG seems not to have any effect on my asterisk.
Let us know what you find.
The effect was caused by an misconfigured phone: The phone did nod
signal busy but ringing due to an call waiting indication.
Switching off call wating indication
Dear users.
My current setup uses a euroISDN E1 with 8 cahnnels for incoming and outgoing
calls from a digium te110p.
Currently all phones use SIP.
However, I need to add some faxes lines and some POS credit card machines.
These will require POTS lines with a fixed DDI.
I have purchased the
On 13 Jun 2007, at 22:48, Alvin Austin wrote:
Hello all,
The wiki has a fairly detailed description of the the issues
involved with encryption of Asterisk calls:
http://www.voip-info.org/wiki/view/Asterisk+encryption
I'm interested in hearing what is working for people today.
I think the
I totally puzzled by this situation. I have asterisk 1.4.4 behind NAT.
All SIP peers are working properly to place or receive calls.
Any SIP peer or friend whether NATted or not will become UNREACHABLE
if qualify=yes.
I have identical peers on the other asterisk 1.2.16 production server.
In
On Thu, Jun 14, 2007 at 09:45:01AM +0100, Matt Scott wrote:
Dear users.
My current setup uses a euroISDN E1 with 8 cahnnels for incoming and
outgoing calls from a digium te110p. Currently all phones use SIP.
However, I need to add some faxes lines and some POS credit card
machines.
Hi
If you use debian install the libmysqlclient-dev package
David a écrit :
Hello Asterisk-Users,
I'm trying to install addons 1.2.6 on Asterisk 1.2.16 (is that OK?),
but my MySQL server is installed on a different sever, so the MAKE of
the addons fails with the following (truncated) error
thanks for you response. Am using sip to access the sound files.The sound files
are recorded with higher sampling rate and 'soxed'to 8khz on the IVR
machine... could it be that resampling is responsible for the
degradation?Date: Wed, 13 Jun 2007 16:44:10 -0400From: [EMAIL PROTECTED]:
I purchased FXS modules so that I could terminate the machines or faxes (eg
just like a standard phone) the outgoing/incoming channel will be be
provided by my E1.
I hope I have the right modules for the job?
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To:
Hi Guy,. you should at least put a subject any way follow this link
http://nerdvittles.com/index.php?p=134 From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Date: Mon, 11 Jun 2007 18:36:54 +0530
Subject: [asterisk-users] (no subject) Hi, please help me in developing
and
Before I go and start coding is anyone aware of an auto-dialer plugin
for Sugar CRM that will allow me to click a button when I'm in someone's
account and have my phone ring and then connect me to them?
___
--Bandwidth and Colocation provided by
On 6/14/07, Matt Scott [EMAIL PROTECTED] wrote:
I purchased FXS modules so that I could terminate the machines or faxes (eg
just like a standard phone) the outgoing/incoming channel will be be
provided by my E1.
I hope I have the right modules for the job?
You do indeed have the right
Exist a module VoiceRD to do that.
JuntaDeAndalucia_es_sf_diphone
2007/6/14, Matt [EMAIL PROTECTED]:
Before I go and start coding is anyone aware of an auto-dialer plugin
for Sugar CRM that will allow me to click a button when I'm in someone's
account and have my phone ring and then
Greetings,
We have An Adtran 616 Total Access device talking to a colocated
Asterisk machine over MGCP. Calls placed to the phones connected to the
Adtran go through as do outgoing calls from the phone (prefixed by 9),
but feature access codes (*97 for voicemail, for example) and
I see that module, but it does not work with the current version of Sugar.
Does anyone have a solution that works with the current version of Sugar?
On 6/14/07, Nuria Fernandez [EMAIL PROTECTED] wrote:
Exist a module VoiceRD to do that.
JuntaDeAndalucia_es_sf_diphone
2007/6/14, Matt [EMAIL
Try vTiger
-E
On 6/14/07, Matt [EMAIL PROTECTED] wrote:
I see that module, but it does not work with the current version of
Sugar. Does anyone have a solution that works with the current version of
Sugar?
On 6/14/07, Nuria Fernandez [EMAIL PROTECTED] wrote:
Exist a module VoiceRD to do
On 6/13/07, Erik Anderson [EMAIL PROTECTED] wrote:
On 6/13/07, C F [EMAIL PROTECTED] wrote:
This is just weird I wrote it in caps so you can read it but you still
didn't read it so here it is again: its a T1 card that does NOT have a
CSU in it, and it is working fine and yes it is a T1
On 6/13/07, Jon Pounder [EMAIL PROTECTED] wrote:
Quoting C F [EMAIL PROTECTED]:
This is just weird I wrote it in caps so you can read it but you still
didn't read it so here it is again: its a T1 card that does NOT have a
CSU in it, and it is working fine and yes it is a T1 providing PRI.
On 6/13/07, Jon Pounder [EMAIL PROTECTED] wrote:
Quoting Erik Anderson [EMAIL PROTECTED]:
On 6/13/07, C F [EMAIL PROTECTED] wrote:
This is just weird I wrote it in caps so you can read it but you still
didn't read it so here it is again: its a T1 card that does NOT have a
CSU in it, and it
Dear Group,
I have just purchased two Linksys SPA941 and flashed these to the latest
firmware.
Everything works well except for the Hold button? Has anyone else
experienced the same issue? What was the solution?
Kind Regards
Shad Mortazavi
___
ESI Phone systems are supposed to support IP stations via SIP
integration(http://www.esi-estech.com/products/systems/ESICS/), has anyone ever
tried to link Asterisk with one of these?
I'm thinking my asterisk box could be an extension off that phone system, that
would then provide a Dial by
Lee Jenkins wrote:
Hi all,
My company has pretty much standardized on Polycom phones and I am in
the beginning phase of writing a GUI for administering/managing polycom
provisioning at multiple sites which we intend to release as OS. I've
started studying the docs and I'm having trouble
Thanks for the suggestion, unfortunately we are using SugarCRM.
On 6/14/07, EdPimentl [EMAIL PROTECTED] wrote:
Try vTiger
-E
On 6/14/07, Matt [EMAIL PROTECTED] wrote:
I see that module, but it does not work with the current version of
Sugar. Does anyone have a solution that works with the
Quoting C F [EMAIL PROTECTED]:
On 6/13/07, Jon Pounder [EMAIL PROTECTED] wrote:
Quoting C F [EMAIL PROTECTED]:
This is just weird I wrote it in caps so you can read it but you still
didn't read it so here it is again: its a T1 card that does NOT have a
CSU in it, and it is working fine and
Quoting C F [EMAIL PROTECTED]:
On 6/13/07, Jon Pounder [EMAIL PROTECTED] wrote:
Quoting Erik Anderson [EMAIL PROTECTED]:
On 6/13/07, C F [EMAIL PROTECTED] wrote:
This is just weird I wrote it in caps so you can read it but you still
didn't read it so here it is again: its a T1 card that
I probably shouldn't be hijacking this thread but it seems that there's
some people paying attention here that know what they're talking about.
We've recently acquired a cisco IAD 2400 router with 2MFT-T1 VWIC card
in it. Doing some cursory reading It seems that this card can be
interfaced
I am seeing this too on both Polycom and Linksys phones, as well as
external SIP peerns not behind NAT, such as FWD. I've posted a
couple of times about it, but I don't see the posts.
On 6/3/07, Ian Clough [EMAIL PROTECTED] wrote:
Hi
I have FC6 system in the office running SVN-trunk-r63567
I noticed that there is a function in the func_odbc.conf called PRESENCE exists.
I am assuming that this goes into dial plan but it is not clear how this might be used. Any ideas?
___
--Bandwidth and Colocation provided by Easynews.com --
Anyone know if it's possible to send a line of text to a phone that's not
currently in-use?
What I want is:
SendText(SIP/101, Hello World)
but that doesn't exist ...
I'm after an application where someone (say a receptionist) can send one
of a small set of pre-defined messages, so that
If they're polycom 501s or higher, you could have each phone use a
different homepage. Those pages could be loaded dynamically (say in php)
and then you could just store the message to display in a database.
Probably easiest to maintain and create an interface for.
Rob
Gordon Henderson wrote:
Before I head down the path of converting voicemail to an ODBC backend, I
have a couple questions that I was hoping someone would know.
1. Is the voicemail message stored in the datbase, or just it's
location/filename?
2. Does MWI propagate when using an ODBC backend?
3. If it does both of
On Thu, 14 Jun 2007, Rob Schall wrote:
If they're polycom 501s or higher, you could have each phone use a
different homepage. Those pages could be loaded dynamically (say in php)
and then you could just store the message to display in a database.
Probably easiest to maintain and create an
We had several of these when we were first playing around with Asterisk.
They are somewhat nice. The audio quality left some to be desired,
however, we did not have a hold button issue.
On 6/14/07, Shad Mortazavi [EMAIL PROTECTED] wrote:
Dear Group,
I have just purchased two Linksys SPA941
On Thursday June 14 2007 1:12 pm, Matt wrote:
We had several of these when we were first playing around with Asterisk.
They are somewhat nice. The audio quality left some to be desired,
however, we did not have a hold button issue.
On 6/14/07, Shad Mortazavi [EMAIL PROTECTED] wrote:
Dear
I think you mean 60x not 50x. The polycom 501s don't have the microbrowser.
Rob Schall wrote:
If they're polycom 501s or higher, you could have each phone use a
different homepage. Those pages could be loaded dynamically (say in php)
and then you could just store the message to display in a
On 6/13/07, Akpome Akpoguma [EMAIL PROTECTED] wrote:
I have been using wav files with sample rate of 8khz and 8 bits and I find
the sound quality really poor.
8khz is correct, if you are using 8 bits, you need to use 16 bits if
I'm not mistaken.
___
I believe the newer versions of firmware do implement the microbrowser on the
501.
- Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
I think you mean 60x not 50x. The polycom 501s don't have the
microbrowser.
Rob Schall wrote:
If they're polycom 501s or higher, you could have
Actually, sorry to not research this first:
14759: Added microbrowser support to the SoundPoint IP 501 platform
from
http://www.voip-info.org/wiki/index.php?page=Polycom+Microbrowser
But I'm not sure which SIP firmware this is talking about being present in.
Mojo with Horan Company, LLC
Actually they do, but only if you're running SIP firmware 2.1 or higher.
Mojo with Horan Company, LLC wrote:
I think you mean 60x not 50x. The polycom 501s don't have the
microbrowser.
Rob Schall wrote:
If they're polycom 501s or higher, you could have each phone use a
different homepage.
Awesome, thanks for this tip!
Moj
Dave Fullerton wrote:
Actually they do, but only if you're running SIP firmware 2.1 or higher.
Mojo with Horan Company, LLC wrote:
I think you mean 60x not 50x. The polycom 501s don't have the
microbrowser.
Rob Schall wrote:
If they're polycom 501s or
Ahh I didn't see that in the first post. Yes Mr. SpamSucks is correct.
You should use 8khz @ 16bits. Using 8khz @ 8bits will sound like a drowning
goat under water.
On 6/14/07, randulo [EMAIL PROTECTED] wrote:
On 6/13/07, Akpome Akpoguma [EMAIL PROTECTED] wrote:
I have been using wav files
Hi List;
Where I can download Asterisk GUI and what I can have
benifit from it?
Regards
Bilal
Be a better Globetrotter. Get better travel answers from someone who knows.
Yahoo! Answers - Check it out.
What does sip show peers output ? Also set a timeout in millisec like
qualify=200 instead of qualify=yes
On 14/06/07, randulo [EMAIL PROTECTED] wrote:
I totally puzzled by this situation. I have asterisk 1.4.4 behind NAT.
All SIP peers are working properly to place or receive calls.
Any SIP
Hello All,
I got one quick question on A2Billing.
Specs: -
- A2Billing v1.3
- OS CentOS 4.5
- Asterisk 1.2
- Zaptel 1.2
Did the installation and everything is working as it suppose to...
Using the A2Billing documentation, I created the RateCard, SIP Trunks,
and SIP Customers. I was also able
On Thu, 14 Jun 2007, C F wrote:
On 6/13/07, Erik Anderson [EMAIL PROTECTED] wrote:
On 6/13/07, C F [EMAIL PROTECTED] wrote:
This is just weird I wrote it in caps so you can read it but you still
didn't read it so here it is again: its a T1 card that does NOT have a
CSU in it, and it
Hello, I'm trying to set up a b410p rdsi card, and I'm having problems
getting it up.
I followed the instruction on asteriskguru and everything seem to be fine
but all leds on the card are in red.
[EMAIL PROTECTED] ~]# uname -a
Linux rdsipbx 2.6.15.7 #2 Tue Jun 5 16:37:07 CEST 2007 i686 i686
On 6/14/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
Where I can download Asterisk GUI and what I can have
benifit from it?
Whaddya know - there's a whole page on the wiki dedicated to such things:
http://www.voip-info.org/wiki-Asterisk+GUI
;-)
I'm a CLI-only guy myself, so I can't
On 6/14/07, Jaswinder Singh [EMAIL PROTECTED] wrote:
What does sip show peers output ? Also set a timeout in millisec like
qualify=200 instead of qualify=yes
Doesn't matter. I've used qualify=2000
There is another thread about this now, OPTIONS response from the
phone is ignored.
Yes they do. If you download the newest boot rom, they sure do have it
now. I was surprised myself when I saw the feature, but we use it on all
of our 501s here. The resolution isn't pretty, but it works. :)
Rob
Mojo with Horan Company, LLC wrote:
I think you mean 60x not 50x. The polycom
On Thu, 14 Jun 2007, Nick Seraphin wrote:
Bottom line is, no matter what the FCC says... and if somehow you
managed to get it to work without a CSU... I believe the phone company
would have a fit if they knew you connected equipment to their network
without a CSU on it. They're very big on
Voip-info has some different links to packages out there for a gui based
asterisk. In my experience, I've found it much easier to tweak a
dialplan and user accounts by hand. We are using realtime/mysql for all
our voicemail/sip/extensions, and I have a small gui I made that creates
those initial
On Thu, 2007-06-14 at 14:46 -0400, Nitesh Divecha wrote:
Hello All,
I got one quick question on A2Billing.
Specs: -
- A2Billing v1.3
- OS CentOS 4.5
- Asterisk 1.2
- Zaptel 1.2
Did the installation and everything is working as it suppose to...
Using the A2Billing documentation, I
Thanks, everybody, for bringing this to my attention! I can't wait to
play around with it!
Moj
Rob Schall wrote:
Yes they do. If you download the newest boot rom, they sure do have it
now. I was surprised myself when I saw the feature, but we use it on all
of our 501s here. The resolution
Can one server (like AMD 6000+ X2) with 2 GIG ram
running asterisk 1.4 handle having 2100 wireless phones connected.
All phones will not be talking at the same time only a couple will be.
There may be 1 T1 card in the box.
Will this work? If not how does one handle this situation.
Thanks,
Jerry Geis wrote:
Can one server (like AMD 6000+ X2) with 2 GIG ram
running asterisk 1.4 handle having 2100 wireless phones connected.
All phones will not be talking at the same time only a couple will be.
There may be 1 T1 card in the box.
Will this work? If not how does one handle this
Jerry Geis wrote:
Can one server (like AMD 6000+ X2) with 2 GIG ram
running asterisk 1.4 handle having 2100 wireless phones connected.
All phones will not be talking at the same time only a couple will be.
There may be 1 T1 card in the box.
Will this work? If not how does one handle this
Hi,
I am trying to set up an E1 line with CAS signaling using available
unicall patches with libmfcr2 implementation. Inbound calls works well,
I am able to get DNIS and ANI from incoming call, but I am still not
able to make an outbound call with our local carrier.
After tweaking of
Hi List;
I did yum install kernel and yum install kernel-devel,
now when I type 'uname' -a I have the following:
[EMAIL PROTECTED] /]# 'uname' -a
Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1
SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386
GNU/Linux
And when I type rpm -q kernel, then I
bilal ghayyad wrote:
Hi List;
I did yum install kernel and yum install kernel-devel,
now when I type 'uname' -a I have the following:
[EMAIL PROTECTED] /]# 'uname' -a
Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1
SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386
GNU/Linux
And
On Mon, Jun 11, 2007 at 11:21:49PM -0500, Rob Ristroph wrote:
Hi everybody,
I have a Fedora Core 4 x86 32 bit install, which I recently
upgraded from asterisk 1.2 to the office 1.4.4 tarball.
In the process of doing that I had to upgrade some
autoconf/automake stuff, but it
On Thu, Jun 14, 2007 at 03:02:20PM -0700, bilal ghayyad wrote:
Hi List;
I did yum install kernel and yum install kernel-devel,
now when I type 'uname' -a I have the following:
[EMAIL PROTECTED] /]# 'uname' -a
Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1
SMP Tue Mar 14 16:05:46 EST
On 6/14/07, Remco Post [EMAIL PROTECTED] wrote:
bilal ghayyad wrote:
Hi List;
I did yum install kernel and yum install kernel-devel,
now when I type 'uname' -a I have the following:
[EMAIL PROTECTED] /]# 'uname' -a
Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1
SMP Tue Mar 14
Hi,
Clearback signal due to billing pulses normally drops calls after a fixed
amount of time 2 minutes or so, Can you stablish an outbound call and after
a while it drops? Or it never succeds?
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Peter Gubis
On 6/14/07, Kyle Sexton [EMAIL PROTECTED] wrote:
Hey Kyle!
1. Is the voicemail message stored in the datbase, or just it's
location/filename?
Yes, the voicemail message itself is stored in the database, as a BLOB
or large object file.
2. Does MWI propagate when using an ODBC backend?
Hi!
Anyone know if it's possible to send a line of text to a phone that's
not currently in-use?
What I want is:
SendText(SIP/101, Hello World)
but that doesn't exist ...
Snom's or Grandstream GXP2000's I'm afraid... Sending text to them while
in a call works fine (although
That was easy... Thanks a million man...
Dunno what I was thinking and went too far writing custom scripts...
Cheers,
Nitesh
Guillermo Salas M. wrote:
On Thu, 2007-06-14 at 14:46 -0400, Nitesh Divecha wrote:
Hello All,
I got one quick question on A2Billing.
Specs: -
- A2Billing v1.3
-
Please disregard.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
David Gomillion wrote:
That's true. But isn't it easier to tell him to check his
/boot/grub/grub.conf file? And only one line...
Easier, but not smarter.
If you'll excuse me I have some wild animals to feed.
--
Jeff Davis
Netsource Consulting
___
On 6/14/07, Nick Seraphin [EMAIL PROTECTED] wrote:
On Thu, 14 Jun 2007, C F wrote:
On 6/13/07, Erik Anderson [EMAIL PROTECTED] wrote:
On 6/13/07, C F [EMAIL PROTECTED] wrote:
This is just weird I wrote it in caps so you can read it but you still
didn't read it so here it is again:
What I am thinking is that a CSU could provide mutiple functions,
error handling, diagnostics and signal boosting, which is not built
into the Panasonic equipment, but the lower level signaling that a CSU
could provide is built into it, and that's why it works.
As far as I knew before I read it
On Thu, 14 Jun 2007, C F wrote:
but the lower level signaling that a CSU could provide is built into it
Possible. In any event, it is this function that describes the
essential aspects of a CSU. But I think the standard is very clear on
the requirements for OAMP stuff too.
--
Alex
On Thu, 14 Jun 2007, C F wrote:
Bottom line is, no matter what the FCC says... and if somehow you managed
to get it to work without a CSU... I believe the phone company would have
a fit if they knew you connected equipment to their network without a CSU
on it. They're very big on
I have two phones on a network behind NAT. Enabling canreinvite=yes on
the Asterisk server allows them to talk to each other very effectively
through the local network.
Unfortunately, calling any outside destinations yields one-way media
issues where the far end can hear me but I can't hear
can anybody help me to choose the most reliable fax solution for * .
after googling the net i found that there are at least two solutions
for this, app_rxfax+spandsp and iaxmodem+hylafax.
- what's the differences between these two?
- which one's better? why?
thanks
78 matches
Mail list logo
