Lee Archer wrote:
I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium TE110P
card in E1 mode. I've recently noticed in my logs the following
Jan 5 01:27:11 VERBOSE[22490] logger.c: [chan_zap.so]Jan 5 01:27:11
VERBOSE[22490] logger.c: [chan_zap.so] = (Zapata Telephony w/PRI)
Sorry I should have stated that I've tried +x, -x, x.y and x and I still
get the same.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: 05 January 2007 08:10
To: Asterisk Users Mailing List - Non-Commercial
I am running a HP DL360 G3 ans want to know the optimal g729 module for it.
There don't seem to be any optimised for Xeon's. I am currently using i686,
but is there a better one to match my Xeon CPU's?
[EMAIL PROTECTED] ~]# cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
On Fri, Jan 05, 2007 at 07:47:15AM -, Lee Archer wrote:
I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium TE110P
card in E1 mode. I've recently noticed in my logs the following
Jan 5 01:27:11 VERBOSE[22490] logger.c: [chan_zap.so]Jan 5 01:27:11
VERBOSE[22490]
Yes I get the same message after reload chan_zap.so
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 05 January 2007 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
On Fri, Jan 05, 2007 at 10:53:17AM +0200, Tzafrir Cohen wrote:
On Fri, Jan 05, 2007 at 07:47:15AM -, Lee Archer wrote:
I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium TE110P
card in E1 mode. I've recently noticed in my logs the following
Jan 5 01:27:11
On Wed, 3 Jan 2007 21:29:45 -
Chris Bagnall [EMAIL PROTECTED] wrote:
Does anyone know if there's a maintained 1.4 ebuild for Gentoo? Even
with the ~amd64 keyword, latest in the official Portage repository is
1.2.13.
hi!
I would not rely on portage for running asterisk on gentoo imho, it
Hi All,as good?
Steve Underwood will not work more with channel Unicall for the Asterisk?
It will be discontinued?
Best Regards
Josué
2007/1/4, Moises Silva [EMAIL PROTECTED]:
1.2, Zap and Unicall work fine
1.4 Only Zap working, Unicall is broken
On 1/4/07, Erick Perez [EMAIL PROTECTED]
HI, I use fc6 , the latest stable asterisk 1.4, zaptel 1.4 and libpri 1.4
after I installed zaptel and libpri. I can start zaptel. and my te410p card got
green lamp. but when I continue to compile and install asterisk, I can't find
chan_zap.so compiled.
and in my asterisk cli. I can't 'help
On 01/05/07 06:18 Zoa said the following:
It used to be a problem to have very big iax2 trunks (e.g. 100 channels).
anyone remember why this was so, and if a bug was opened on this for 1.2 ?
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED]
The absolute best results I have had were with m0n0wall (m0n0.ch) which
worked perfectly for me to bounce voip calls over vpns with other traffic
and no user any the wiser. Second after that but with lots of plus points
for value come the draytek routers. A couple of years ago, their firmware
used
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Allen Casteran
Sent: Friday, January 05, 2007 12:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] POE draw on Aastra 480i
Anyone know what the POE draw is for the Aastra 480i
On Thu, 4 Jan 2007, Noah Miller wrote:
Hi Damon -
Can anyone comment on the overhead added when a SIP call comes into one
asterisk box, is routed to another with IAX instead of SIP, and is then
sent
to the UA from the second box with SIP?
DTMF passthrough issues?
I've got a client with
Hello,
I am trying to build asterisk 1.4.0 (stable). The problem is make does
not build res_crypto.so
I have installed:
gcc
libc6
m4
openssl
zlibc
libkrb5-dev
libncurses5
libncurses5-dev
libssl-dev
zlib1g-dev
I know I will need res_crypto.so for iax, so...
Can anyone tell me how to do?
Anyone know what the POE draw is for the Aastra 480i phones?
We have switches that will do 15 watts on 12 ports but only do 7.7
watts on all 24 ports. A Cisco 3560 switch will do
15.6 watts on all 24 ports.
Just trying to find out if we need that much power.
Can't seem to find
Hi,
I receive data from our ISDN PTP Line (by T-Com, Germany) that seems to be
not processed correctly by Asterisk 1.2.13-BRIstuffed-0.3.0-PRE-1w
pri debug:
1 Length (57) of 0x38 component is too long
1 !! Invalid DivertingLegInformation2 component received 0x38
This only happens on calls that
I'm trying to use the txfax application based on
spandsp in Asterisk 1.2. It seems to be working but I
would need a way to reliably check whether the fax has
been completely transferred or not. I'm using a
mail2fax system (as with email2fax and .call files)
but I can't seem to get it working.
If I
Hi!
Exactly what I needed.
It was the 209 part that I did not figure put.
Thanks!
//Mattias
At 03:53 2007-01-05, you wrote:
exten = _9070X./209,1,NoOP,SORRY CHARLIE
exten = _9070X./209,2,Congestion
This would block any call from 209 to 070X as
long as 9 was your outside digit.
I use the
I'm using 1.2.9.1, with the metermaid patches to show parking spot
status on Snom BLF lights.
I see from http://www.asterisk.org/node/97 that the metermaid code has
changed substantially since 1.2.9.1.
Is anyone successfully using the new metermaid functionality in 1.4.x?
Did anyone get
Hello group
I have been asked to get IM via the X-Ten softphone to work with
Asterisk. Anyone have any ideas? I have looked on google and other
places with no luck.
Our system is as followed
Linux CentOS 4.4
Asterisk 1.4.0-beta3
X-Lite v3.0 for Windows
Thanks!
Eric Hall
Hi Users,
I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER,
After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk,
In openser.cfg ... is not hiiting the Asterisk server
. ... any one help me
Couldnt agree with you more Lee.
I think its very difficult for a software company to be able to stay focused
on developing the software while been profitable, thats why many companies
turn to consulting services (Sun), other develop hardware (IBM and OS/2 :))
Digium has been doing a great job
Hi Josue, as of today at least, Steve Underwood is focusing his efforts into
making unicall be the basis for openpbx so will not be devoting more time
into unicall and asterisk.
This could change maybe but thats what he told me a few days ago.
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Olivier wrote:
Hi,
For a 20 users prospective customer, I'm wondering if any GUI would
allow and end user to edit an Asterisk IVR tree ?
For instance, I'm looking for something allowing to edit interactions like :
wait up to 20 seconds and say this to reach sales department, type 1,
to
Please can you provided me by a radius module name for asterisk,or how to
authorize user and get cdr from radius server.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by
Allen Casteran wrote:
Anyone know what the POE draw is for the Aastra 480i phones?
We have switches that will do 15 watts on 12 ports but only do 7.7
watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all
24 ports.
Just trying to find out if we need that much power.
Eric Bishop wrote:
I am running a HP DL360 G3 ans want to know the optimal g729 module for
it. There don't seem to be any optimised for Xeon's. I am currently
using i686, but is there a better one to match my Xeon CPU's?
Xeon is not an architecture, it's a brand name. Various Xeons have been
I think we are going to do it if we get big problems with those many
queues. From what I'm seeing, the biggest problems seem to be related to
agents, so maybe we can have a try at using straight terminals instead of
agents.
l.
On Fri, 05 Jan 2007 01:14:08 +0100, Leo Ann Boon [EMAIL
On 1/5/07, Anton Krall [EMAIL PROTECTED] wrote:
when did asterisk turn from an open source project with very good developers
nto a business that only focuses in $$$?
They are not mutually exclusive.
That's why openpbx was born I guess
I dont think so. I think is more because of technical
I have 2 asterisk servers connected together on internet, when placing one
or two calls, things goes fine, but when placing more calls, i am getting
the below messages on the far end:
Jan 5 17:25:00 ERROR[2679] chan_sip.c: Call from peer 'switch' rejected due
to usage limit of 16
Jan 5 17:25:00
By Trixbox, do you mean FreePBX (formely AMP) ?
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On 1/5/07, Lenz [EMAIL PROTECTED] wrote:
I think we are going to do it if we get big problems with those many
queues. From what I'm seeing, the biggest problems seem to be related to
agents, so maybe we can have a try at using straight terminals instead of
agents.
l.
Being somewhat familiar
On Friday 05 January 2007 10:31, Benko wrote:
On Wed, 3 Jan 2007 21:29:45 -
Chris Bagnall [EMAIL PROTECTED] wrote:
Does anyone know if there's a maintained 1.4 ebuild for Gentoo? Even
with the ~amd64 keyword, latest in the official Portage repository is
1.2.13.
Short answer not at the
Hi,
I have installed asterisk on Ubuntu 6.06 server CD
All required packages has been installed and upgraded
When start sudo make menuselect from addons, I can't select all addons
that require mysqlclient (app_addon_sql_mysql, cdr_addon_mysql,
res_config_mysql).
If I run apt-cache search
On 1/4/07, Douglas Garstang [EMAIL PROTECTED] wrote:
Richard,
We have underscores all over the place in our config files, including others in
queues.conf. I don't think that's the murder weapon.
I think, in general, queues are one of Asterisks biggest features, and also one
of it's shakiest.
When you have a bunch of analog phones that you want to connect to
asterisk, but those analog phones have no transfer button, what are
the options to allow the phones to transfer a call?
--
Erick Perez
Panama Sistemas
Integradores de
So anyone else any ideas?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 05 January 2007 09:30
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] chan_zap.c: Failed to read gains:
Invalidargument
On Fri, Jan 05, 2007
Drew Gibson wrote:
Allen Casteran wrote:
Anyone know what the POE draw is for the Aastra 480i phones?
We have switches that will do 15 watts on 12 ports but only do 7.7
watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all
24 ports.
Just trying to find out if we need that
I think you are misunderstanding several points here Moises.
I do give Digium a break like you said, thats why you have options, you can
use digium cards or sangoma cards, it's up to you, I use digium cards from
time to time because I like to support digium in what they are doing.
But from the
Can you set verbose and debug levels to 10 and run these commands:
*CLI module unload app_cbmysql.so
*CLI module load app_cbmysql.so
*CLI core set verbose 10
Verbosity was 0 and is now 10
*CLI module unload app_cbmysql.so
Unable to unload resource app_cbmysql.so
Command 'module unload
On Jan 5, 2007, at 12:02 PM, Erick Perez wrote:
When you have a bunch of analog phones that you want to connect to
asterisk, but those analog phones have no transfer button, what are
the options to allow the phones to transfer a call?
You can setup a dial rule to do transfering based on
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.
thanks in advance
Biju
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Hello,
I have asterisk 1.4 with ooh323c addons installed. (As I am a newbie in
voip world...my question might be idiot...! ;) Please forgive me!)
I succeed to make H323 call when ooh323c is configured as gateway
(gatekeeper=DISABLE in ooh323.conf).
When I put gatekeeper= ip_address, and
T1s can use many different signalling types. You need to find out
which
one is running, what the line encoding is, etc. PRI vs T1 are not the
only
distinctions...
On 1/4/07, Mark Greene [EMAIL PROTECTED] wrote:
Alright guys here is my question. What is do I need to set
switchtype, and
*CLI core set verbose 10
Verbosity was 0 and is now 10
*CLI module unload app_cbmysql.so
Unable to unload resource app_cbmysql.so
Command 'module unload app_cbmysql.so' failed.
*CLI [Jan 5 11:09:04] WARNING[30610]: loader.c:465
ast_unload_resource:Firm unload failed for app_cbmysql.so
In the route[1] you do not show a t_relay. Is this just be a typo or is
the t_relay actually missing from this route block? If it is missing how
do you initiate the relay which you hope to end up in voicemail?
Daniel-Constantin Mierla wrote:
Hello,
watch the network traffic with ngrep -d
Anton Krall wrote:
after all, like you said, it is open
source..
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Moises Silva
|At the end, is open source/freesoftware, if you dont like it, nobody
|is stopping you from change it.
When you have a bunch of analog phones that you want to
connect to asterisk, but those analog phones have no
transfer button, what are the options to allow the phones
to transfer a call?
Check out features.conf
You can specify key presses for things such as transfer.
Don Pobanz
Hi Anton, thank's will be its reply.
It would be good for asking for to the Steve that did not abandon this
project (unicall and asterisk-1.4.x), in the versions 1.0.9 and 1.2.x when I
had problems and always I referred it, aiming at the improvement of channel.
It would be a great loss for all.
On Fri, Jan 05, 2007 at 05:44:28PM +0100, Luca Lafranchi Lists wrote:
Hi,
I have installed asterisk on Ubuntu 6.06 server CD
All required packages has been installed and upgraded
When start sudo make menuselect
As a rule, make as a user, make install as root. No need for sudo
for
Hi folks,
I'm using a fewestcall queue here, and I'm having the follow problem:
I have 3 static agents in my default queue:
2001
2002
2003
User 2001 and 2002 are logged in, but 2003 are logged out. When someone call
to my default queue, the queue try to ring 2003 (that isn't logged). There
is
Erick Perez wrote:
When you have a bunch of analog phones that you want to connect to
asterisk, but those analog phones have no transfer button, what are
the options to allow the phones to transfer a call?
Press the switch hook for 1 second. In some parts of the world FLASH
is called RECALL.
At 09:35 PM 1/4/2007, you wrote:
Just trying to find out if we need that much power.
I'd guess not, I can tell that my 480iCT is using power because I
think I can feel warmth, but it sure doesn't feel like there's a 7
watt light bulb inside. It claims to need a 48V .13 amp wall wart
if
I'm still learning some of the basics. Can someone explain in layman's
terms what's the difficulty for Asterisk to support SIP/TCP (and even
RTP/TCP)?
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To
Don, I suppose that in order for this to work i need canreinvite=no, right?
On 1/5/07, Don Pobanz [EMAIL PROTECTED] wrote:
When you have a bunch of analog phones that you want to
connect to asterisk, but those analog phones have no
transfer button, what are the options to allow the phones
Hello,
are there any (possibly experimental) asterisk debian packages (or at
least a debian/ directory to build our own)?
Previously I used to modify debian/ directory from earlier version,
but 1.4 changed build process, so this is not that easy.
Thank you,
Juraj.
I was thinking of an HP DL140 with two 250gig sata disks and one
3.8Xeon CPU with 2gig RAM.
Should be plenty if not an overkill. One of our setups: 20 phones, 8
outgoing/incoming SIP trunks, MeetMe conferencing with ztdummy and no
Zap hardware. IVR/voice mail/MOH/Recordings/etc. Runs on a
On Fri, Jan 05, 2007 at 09:16:07PM +0100, Juraj Bednar wrote:
Hello,
are there any (possibly experimental) asterisk debian packages (or at
least a debian/ directory to build our own)?
Previously I used to modify debian/ directory from earlier version,
but 1.4 changed build process, so
The problem is that you've setup 2003 as a static user. In FreePBX, static
users are ALWAYS in the queue, no matter what.
Take this guy out of the queue as a static agent, and have him login and
logout as he needs. (login via ##* and logout via ##**, where ## is the
number you've given your
Hi,
I wrote a few days ago about my problem with calls via voipstunt
stopping ringing after 5-6 rings, subj. SIP Dial out timeout. Though
if the remote station picks up before that, everything works flawlessly.
I am not entirely sure when this phenomenon popped up, but I used the
this
Hi all,
Quick question. Is there a way to have multiple people have an
extension, say 900, to their polycom 501 SIP phones on one of the blue
buttons to where when a call comes in, I can have it simul-ring and
folks can pick up the line on their phone? I'd like to set up a tech
support
On Friday 05 January 2007 21:06, Phil Finkler wrote:
Hi all,
Quick question. Is there a way to have multiple people have an
extension, say 900, to their polycom 501 SIP phones on one of the blue
buttons to where when a call comes in,
exten = 900,1,Set(CALLERID(name)=TechSupport)
exten =
Erick Perez
Don, I suppose that in order for this to work i need
canreinvite=no, right?
No! It doesn't matter what you have for 'canreinvite' since
'canreinvite' is a SIP attribute, not an analog phone attribute.
For analog phones, Asterisk will always be in the call path. :-)
--
Don
UDP is a stream of packets with no layer 3 receipt acknowledgments.
Great for games and media.
If a packet is dropped or damaged, the receiver just skips it and uses the next
packet.
TCP is a more controlled transmission of packets with receipt acknowledgements
sent back after a certain number
You're quite right, I typed before thinking. Upload is the problem anyways,
since it usually (in homes) uses much more limited bandwidth than
downloading does.
No answer to my question though: How do you people handle QoS without
relying on the phones to do that? I'd like a box that can be
Hi Mike,
The Linksys WRT54G can do QoS, and I've found it to be a great little
router... I install the DD-WRT open source firmware on mine for additional
features, but the stock firmware works well also.
Alex
On 1/5/07, Mike [EMAIL PROTECTED] wrote:
You're quite right, I typed before
TCP is a connection oriented protocol ..as others mentioned, it superiority
comes because it knows when packets are dropped to resend them. It also has
mechanisms for flow control etc.. SIP is a connection-less protocol. It uses
'best effort' transmissions..if u want its delivery guaranteed you
Hi all,
I am attempting to build a horizontally scalable Asterisk deployment and
am getting very close to achieving that goal. With Asterisk 1.4 I now
have an IMAP backend for Voicemail messages which is great as users can
check the same messages either through the voice portal or using
I seem to be having a problem that I have narrowed down to a
disagreement on codec negotiation or codec setup of some kind in an IAX
peering arrangement. Here's a non-ASCII art version of the setup:
DID origination provider
via SIP/gsm
to
Call routing asterisk server
via IAX/gsm
Anybody selling DID flat rate for less then $4 with sip or iax incoming? Let me
know
Thanks
CM
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
Well it would be interesting to know what FXS device you are using to
connect the analog phones. I use an SPA-3000 fxo/fxs and with it you could
bypass Asterisk and connect the FXO to FXS or dial directly if it were so
configured, so reinvite would work but wwould probably not be desired but
that
Mike wrote:
You're quite right, I typed before thinking. Upload is the problem
anyways, since it usually (in homes) uses much more limited bandwidth
than downloading does.
No answer to my question though: How do you people handle QoS without
relying on the phones to do that? I'd like a box
Does this model give you functioning mwi?
-Original Message-
From: Ray Jackson [mailto:[EMAIL PROTECTED]
Sent: Friday, January 05, 2007 3:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail personalised greetings using
Ray,
Have you considered using a VM architecture?
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Jan 5, 2007, at 5:17 PM, Ray Jackson wrote:
Hi all,
I
On Fri, Jan 05, 2007 at 04:11:15PM -0600, James R. Stevens wrote:
TCP is a connection oriented protocol ..as others mentioned, it
superiority comes because it knows when packets are dropped to resend
them. It also has mechanisms for flow control etc.. SIP is a
connection-less protocol. It
On Wed, 2006-12-13 at 21:51 -0300, Josué Conti wrote:
Kevin, contributes with the list, somebody can have this problem and
you it can help. The list is here for helping, but also we must
contribute with it. :)
Best Regards
I have the same problem.. any one know can I solve it?
Best
Ray Jackson wrote:
If there a way of storing these greetings in a database table or using
IMAP?
The current implementations of ODBC and IMAP for voicemail use it only
for voicemail, not for greetings. However, there is still work being
done by community members on both methods of storage, so
Hi everyone,
We are running Polycom 601's. I can't seem to find anything to say one
way or another on this issue, so I figured I would ask. I have call
waiting notification working on the phones when a user is on the phone.
However, is it possible to see the notification on the screen or hear
I've got a curious one: all of a sudden my .call files and my manager
API 'Originate' actions are no longer producing a CSV file. The call
still generates just fine, and Master.csv is updated. However, I don't
get the usual CSV file in the form of xx.csv where xx=account
number.
I
Vieri Di Paola wrote:
I'm trying to use the txfax application based on
spandsp in Asterisk 1.2. It seems to be working but I
would need a way to reliably check whether the fax has
been completely transferred or not. I'm using a
Then you'd want to use iaxmodem and HylaFAX+.
Doug
--
Ben
From: James R. Stevens [EMAIL PROTECTED]
TCP is a connection oriented protocol ..as others mentioned, it superiority
comes because it knows when packets are dropped to resend them. It also has
mechanisms for flow control etc.. SIP is a connection-less protocol. It
uses 'best effort'
Hi Douglas,
Yes, MWI works fine as each Asterisk server looks after it's own set of
registered users. I am simply created a shared backend for voicemail
storage (IMAP) and MySQL for voicemail configurations. The missing
piece is how to store personalised greetings in a shared backend.
Hi Bryan,
I was trying to avoid creating an architecture dedicated to VM, but have
Asterisk handle VM in a horizontally scalable way. I understand there
are some issues with MWI etc. if you separate out the VM from Asterisk?
Could you point me at any good examples of a VM architecture I
Kevin Smith wrote:
We are running Polycom 601's. I can't seem to find anything to say one
way or another on this issue, so I figured I would ask. I have call
waiting notification working on the phones when a user is on the phone.
However, is it possible to see the notification on the screen or
Hi Kevin,
Thanks for your response. That answers a few questions I had. I am
very happy to get involved in this area if I can help. Using IMAP and
REALTIME I have a really nice VM solution with MWI, Webmail access etc.
and it scales horizontally - I just add a new server into the mix when
Ray Jackson wrote:
necessary. Until we get a generlized storage subsystem in place, I may
look at a 'hack' to get the personalised greetings going... Do you think
a shared NFS mount is risky for this? Should I do an rsync periodically
perhaps to keep greetings on all servers up to date with
what if I go with full g711-no transcoding?
remember that I will have an E1 coming in, so my usage can be up to 30
channels at once.
if that is an overkill machine config, and for obvious reasons I cant
use old hardware, what are your suggestions?
thanks,
On 1/5/07, Luki [EMAIL PROTECTED]
On 1/5/07, Doug Crompton [EMAIL PROTECTED] wrote:
Well it would be interesting to know what FXS device you are using to
connect the analog phones. I use an SPA-3000 fxo/fxs and with it you could
bypass Asterisk and connect the FXO to FXS or dial directly if it were so
configured, so reinvite
I've got a curious one: all of a sudden my .call files and my manager
API 'Originate' actions are no longer producing a CSV file. The call
still generates just fine, and Master.csv is updated. However, I
don't
get the usual CSV file in the form of xx.csv where xx=account
number.
On 1/5/07, Ray Jackson [EMAIL PROTECTED] wrote:
Hi Kevin,
Thanks for your response. That answers a few questions I had. I am
very happy to get involved in this area if I can help. Using IMAP and
REALTIME I have a really nice VM solution with MWI, Webmail access etc.
and it scales
Erick Perez wrote:
what if I go with full g711-no transcoding?
remember that I will have an E1 coming in, so my usage can be up to 30
channels at once.
if that is an overkill machine config, and for obvious reasons I cant
use old hardware, what are your suggestions?
I would suggest you go for a
UDP is preferred for VoIP because by the time a dropped packet is
detected, the retransmission request is sent to the originator, and the
replacement packet arrives, it's too late (unless you are running a very
large jitter buffer which introduces problems of its own).
Conversely if your
David Thomas wrote:
In the DUNDi * cluster we're designing phones can register with any of
our asterisk boxes. Actually sometimes phones are registered to
multiple boxes. I'm wondering if the new IMAP/MWI would have any
problems with this type setup. Any experiences here?
I am using SRV
I have been asked to get IM via the X-Ten softphone to work with Asterisk.
Anyone have any ideas? I have looked on google and other places with no
luck.
Our system is as followed
Linux CentOS 4.4
Asterisk 1.4.0-beta3
X-Lite v3.0 for Windows
If by IM, you mean the built-in Jabber stuff in
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 phones always have higher priority traffic
than the PCs. (and not rely on the phones to do the QoS because some PCs
On 1/5/07, Dan Austin [EMAIL PROTECTED] wrote:
Trunk has already moved on and code compatible with 1.4, may have
problems on it. For a sanity check, I wiped out my test system
and rebuilt it with fresh components for 1.4 (libpri, zaptel, asterisk,
asterisk-addons), and I have no issues with
use a simple agi - php is easy to do.
--- O.Kamal [EMAIL PROTECTED] wrote:
I just need to retrieve a value from a field in a postgres database,
and
playback this value when someone dial a specific extension.
On 1/4/07, Thomas Kenyon [EMAIL PROTECTED] wrote:
O.Kamal wrote:
I need to
That's what i told you Mattias.
On 1/5/07, Mattias Andersson [EMAIL PROTECTED] wrote:
Hi!
Exactly what I needed.
It was the 209 part that I did not figure put.
Thanks!
//Mattias
At 03:53 2007-01-05, you wrote:
exten = _9070X./209,1,NoOP,SORRY CHARLIE
exten = _9070X./209,2,Congestion
This
Kenneth
Thanks for the reply. What I'm looking to do is listed here
http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging
However the patch does not work on the system listed below.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kenneth
Padgett
On 1/5/07, Anton Krall [EMAIL PROTECTED] wrote:
I think you are misunderstanding several points here Moises.
May be
I do give Digium a break like you said, that's why you have options
I dont understand this. How is related that you give Digium a break,
with the fact
that I have the option of
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