Cisco 7960's: (SIPified)
1. Cheap
2. 6 lines is plenty
3. simple to config
4. stable
On Jan 6, 2008, at 11:03 PM, William Herrera wrote:
> Alright, enough.
> At first I was to ignore to you all making statements like this one
> but I
> feel at this point that if I do not stop this it seems it
That's like asking 'what is the best car in the world and why?'
If you need to haul lots of people, you buy a minivan.
If you live in the US and need to haul lots of people, you buy an SUV.
If you need speed, to buy a motorcycle.
If you need..
You need to decide what features are required, desi
I thought this was interesting, if you are in China and need a job, you
might also...
http://www.cisco.apply2jobs.com/index.cfm?fuseaction=mExternal.showJob&RID=771671&CurrentPage=1
"* Working knowledge : Asterisk PBX; SIP Proxy Servers."
___
--
You can buy smartnet on a single phone for something like $8 a year. This will
get you in legally.
-Original Message-
From: [EMAIL PROTECTED] on behalf of David Parcerisa
Sent: Fri 2/23/2007 1:57 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] cisco sip firmware update f
I don't think you need the pipe in there. I've used this with the "w" option
before, which adds a wait. Then continues .5 seconds later.
RTM: http://www.voip-info.org/wiki-Asterisk+cmd+dial
Try these:
exten => s,2,Dial(SIP/TelaSip-gw4/5198881212D12345678)
or
exten => s,2,Dial(SIP/TelaSip-gw4/519
How does one answer more than 100% of the calls in less than 60 seconds?
techsupp has 0 calls (max 20) in 'leastrecent' strategy (104s
holdtime),
W:0, C:3, A:2, SL:166.7% within 60s
___
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aste
So, after reading this, I wonder if anyone has 1.4 and MySQL working...
Is there a non-standard version I can download?
more /usr/src/asterisk-1.4.0/doc/mysql.txt
MYSQL LICENSING UPDATE
==
We were recently contacted by MySQL and informed that the MySQL client
libraries are no
ers] Guide to better performance using * ?
Tim Connolly wrote:
> Can someone point me in the right direction to find documentation
> on best practices when setting up a new Asterisk server? I'm using
> RHES4 and Dell 1750 with TE412P. My current problems are frequent
> cr
nt: Thursday, 15 February 2007 5:54 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Guide to better performance using * ?
>
> Tim Connolly wrote:
> > Can someone point me in the right direction to find
> documentation on
>
Can someone point me in the right direction to find documentation on
best practices when setting up a new Asterisk server? I'm using RHES4
and Dell 1750 with TE412P. My current problems are frequent crashes and
choppy audio so I think I can easily tweak these out of the picture.
___
rcial Discussion
Subject: Re: [asterisk-users] Rt db lookup
On 1/15/07, Tim Connolly <[EMAIL PROTECTED]> wrote:
>Which command effects whether or not the * server will lookup a
> peer from the db even though the phone isn't registered locally?
>
>I have several * serv
Which command effects whether or not the * server will lookup a
peer from the db even though the phone isn't registered locally?
I have several * servers but I want any server to be able to
lookup and send a call to phones registered on another server (SIP
cluster?).
Thanks
Tim
__
Its not quad band and in my opinion doesn't perform well enough to be
used for anything but basic email and phone calls. This phone, even on
the newest version of firmware (Sprint) hangs when syncing with exchange
to the point where you miss calls even though you tried to answer them.
If you turn
When I send a call from my TE410P using E&M, the legacy PBX answers
the call but doesn't route it. Any idea what this could be? I assume the
digits aren't being delivered properly to the legacy pbx. Any
suggestions on what config settings to muck with?
Asterisk SVN-branch-1.2-r40901 built b
My garage door is...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Crompton
Sent: Monday, December 04, 2006 11:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Is there any Asterisk controllable
thermos
Anyone know the command that tells * to load a sipfriend
from the realtime db rather than saying no such host? I've tried various
combinations of the rt commands:
rtcachefriends=yes;
;rtcache=yes
;rtAutoClear=yes
;rtautoreg=yes
;rtIgnoreRegExpire=yes
;rtupdate=yes
rtfromcontact=
Fyi... My apologies if this is a dupe.
-Original Message-
From: Cisco Technical Support
[mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 13, 2006 8:52 AM
To: Tim Connolly
Subject: New Software available on Cisco.com
New software images are available on Cisco.com for the product
This is more of a MySQL question.. But its going to look something like:
ALTER TABLE `extensions_table` ADD `variable_name` DEFAULT '0'
NOT NULL ;
>From the specs page:
http://dev.mysql.com/doc/refman/5.0/en/alter-table.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL P
A little more RTFM'ing and voila!
Using MeetMeCount I should be able to record only the first user.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMeCount
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Connolly
Sent: Tuesday, Decemb
Has anyone found a way to monitor a meetme conference for only
the first user? I find have one recording per user is pretty hard on the
server performance wise... Suggestions?
___
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asteri
Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686
running Linux on 2006-06-17
When I used monitor, I seem to get most calls cut off if they run
very long. Sometimes two minutes, sometimes 5 or 15.. Seems random. Any
ideas what might kill the recording process? I'm begi
Does anyone know what triggers the 7970 to update its config? I
was able to get it to update to SIP, but the config I used initially
won't go away. I am making small changes to the SEPxxx.cnf.xml file and
rebooting the phone, the phone is downloading the (TFTP) new config
file, but I do
I'm seeing the same issue, options tTH doesn't seem to help
either...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
HishamSent: Monday, July 17, 2006 3:25 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[asterisk-users] DTMF in QUEUES dont wor
Looks like the MWI broke on 8-3 also...
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After upgrading my phones I now see routine error messages:
-- Got SIP response 400 "Bad Request" back from 10.5.1.94
Asterisk SVN-trunk-r7230
Cisco 7960 SIP version 8-3-0.
Sip show peer:
* Name : 14012
Secret :
MD5Secret:
Context : labcm33
Subscr.Cont.
Can someone provide an example of how to use HINT priority with
Cisco 7960/SIP phones? I don't fully understand what exactly the hint does,
but I believe it mimics a legacy PBX's bridge-appearance function. Is this
correct?
___
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Justin asked me to post a note about a new version of
NVBackgroundDetect coming out very shortly. Be patient!
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ht
Google is your friend, or you enemy, either way they usually have an answer:
"/Dial a single destination, ringing for a maximum of 20 seconds. Limit
the call length to 60 seconds, warning the caller when only 20 seconds
remain:/
exten => 200,1,Dial(SIP/1234,20,L(6:2))
" -- http://www.
Did you upgrade all the mysql packages, or just the server? I would bet
you missed the -dev or -lib package.
kritikus Araklidas wrote:
Hi everyone:
I installed the lates version of Asterisk with Asterisk Add-Ons. A
month ago i upgraded my database form mysql 4.1 to mysql 5.0. So after
to st
I was offered an upgrade path for my two Dell 1750's (2.8 Dual Xeon)
to get into a pair of new Dual Core Dual Opteron servers. Assuming I can
get the IRQ BS worked out so my TE411XP doesn't flip out, this should be
a pretty significant upgrade. Has anyone been able to quantify any
benefits t
I've had a few, even on 7.4+, that were impossible to recover the
password. I usually end up looking at the current network settings and
putting an IP alias on my tftp server so it will answer the tftp get
requests coming from the phone. It gets tricky when the original config
has the TFTP s
I've been seeing this for a while. No clue how to fix. The source I have
from my last update says extra_log=0, so it "shouldn't" be showing this
message at all...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JR
Richardson
Sent: Tuesday, March 28, 2006
Figured this was worth passing on...
This was reported due to the proxy IP address showing up in CallerID on
the phone.
-Original Message-
Sent: Thursday, March 23, 2006 12:01 PM
Tim,
I have tracked down the source of the change in the SIP firmware. The
behavior was changed as a fix t
Non-Commercial Discussion"
>
> Sent: Monday, March 13, 2006 8:44 PM
> Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?
>
>
>> We rolled back to 7.4 cause of that too. 7.5 has a strange bug where
>> if the server loses connection, the phone'
Sure, just make your voicemail wait 5 seconds before
answering the call.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Navneet
ShahSent: Thursday, March 16, 2006 10:45 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Dialplan :
Forwarding call to voicemail a
olled back to 7.4 cause of that too. 7.5 has a strange bug where
>> if the server loses connection, the phone's just don't try
>> re-registering.
>>
>> Aaron
>>
>> Tim Connolly wrote:
>>> Just curious, why not 7.5 ? -Original Message-
&g
I'm using P0S3-08-2-00.. I noticed the callerID started showing
up with the number, then @... So the callerID on the phone
looks like: [EMAIL PROTECTED] which of course is logged in the
missed calls exactly like that, and completely foobars the dialing
string if you try to dial a missed cal
Just curious, why not 7.5 ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Monday, March 13, 2006 2:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists pro
Has anyone tried the Polycom 4000 on SIP/* ?
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/etc/asterisk/musiconhold.conf:
[default]
mode=files
directory=/var/lib/asterisk/mohmp3
application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing MusicOnHold("Zap/1-1", "") in new stack
-- Started music on hold,
I'm replying to this mainly to add my comments to the archive and then all
the webcrawlers...
I found a deprecated command "curl" which I though had simply been converted
from an app to a function, was actually completely non-working. Anytime my
call hit a "exten => s,1,set(CURL=curl()), the chan
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Monday, February 06, 2006 4:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TE405p -- loopback for the phone company?
Tim Connolly wrote:
> I wonder if Digium has any intenti
I wonder if Digium has any intentions of fixing this. I brought this to
their attention shortly after purchasing a pair of TE411's. You can issue a
loopup on span 2 only to get a message saying "looping span1" which is to
say, a bit scary when you only have two active PRI and one is already down
fo
Were you able to acomplish this?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan
Sent: Thursday, October 27, 2005 5:31 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Zapbarge feature available?
We would like to beable to lis
Wouldn't it be easier to keep the agents in the table all the time, and
simply update the logged_in status column for that row?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Tuesday, August 30, 2005 12:23 PM
To: Asterisk Users Mailin
I light to come on. It's probably
considered half-assed, but it works like a charm for us.
Aaron
Olle E Johansson wrote:
> Tim Connolly wrote:
>> I have a pair of servers tied to PRI's which only do SIP/ZAP/IAX
>> terminations and routing. The IVR/Voicemail is on
I have a pair of servers tied to PRI's which only do SIP/ZAP/IAX
terminations and routing. The IVR/Voicemail is on a separate server which
accepts calls from the front line servers via Iax. I am trying to use the
MWI on our Cisco 7960 phones, which isn't working, but I think its because
the voi
I've got realtime sipfriends running
pretty well. One this I noticed is that if I make a change to the DB, the
server's 'sip show peer 1234' never shows the update until after I do a 'sip
reload'. My info, cvs-head from 12/17/05 on a Dell 1750. the mysql db is
on a seperate server, as
I'm looking for ideas on how to implement voicemail notification on
Cisco 7960 SIP phones. Something like a light on the legacy pbx-phone
would be perfect. Even maybe go so far as a quick ring to the extension
every 15 minutes or so, but then that would increment the on-screen
missed call co
I've got a cvs-head box running RHES4. I have it spread across two
vlans as I am using 802.1q trunking to my cisco switch. On the "front" side,
everything works great. If I move a phone from the front to the rear, change
its IP address and its config to reflect that move, and update the SIP
Asterisk cvs-head (up to date) keeps core dumping on me. I finally
tracked it down to my register command for Vonage in the sip.conf file. If I
remove the username and password from the register command, it won't core
dump, but of course won't register either... This is odd. Any suggestions
SJPhone and a few others. Seems to work well. A little small for my head
though. Not bad for $50. The Logitech is right up there
with it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Tuesday, August 16, 2005 12:20 AM
To: 'Asterisk Us
I’ve got an application where I need to
simply dial the console (local sound card using OSS driver) and pass any DTMF tones to the
console. No matter whether I come in on a zap/sip/iax channel, the DTMF is
always being muted. Is there anyway to disable this? I’m not specifying
a
Almost.. A call on hold doesn't represent the true bandwidth and CPU that a
*real* call utilitizes. Short of producing an echo or feedback on each call
to make it look like a real call, I'm not sure how you could create a real
call test scenario.
-Original Message-
From: [EMAIL PROTECTED]
You might try to su - apache and make sure apache can read the file.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy Vega
Sent: Saturday, August 13, 2005 5:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] vmail.cgi
I'm trying to ge
On that note... IPSec tunnels seem to reek havoc on the echo
canceling/training process. Anytime our Cisco PIX loads up, the echo
complaints start coming in. Stay away from the IPSec tunnels.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Travers
S
If you need a FXS, Vonage starts at $15. If you want to simply go soft-only,
Broadvoice would be a better choice. After the marketing and all the
features that nobody uses are thrown out, it comes down to consistency.
Broadvoice has had some problems in the past 6 months, Vonage hasn't (that I
know
Goodyear
Sent: Saturday, August 13, 2005
12:59 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Suggestions for mainstream hardware compatiblewith TE411P.
On Aug 12, 2005, at 7:06 PM, Tim Connolly wrote:
I checked the list of
I checked the list of what not to use, but am
still having no fun trying to find a working box. Can someone suggest a Compaq
or Dell or MPC or … any other brand and model that is known to work well
with the TE411P ? Will an old Proliant do?
___
You might start by running /usr/src/zaptel/zttest. See if you stay at 100%.
That's going to be the first thing digium checks. You might also run the
autosupport script and take a look at it for anything obvious.
I'm having lots of stability problems with my 411's. I'm not blaming the 411
yet, just
I could probably shoot about 115 calls towards you, would that do ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Friday, August 12, 2005 8:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Lo
Wow! Not sure what else to say. This ranks right up there with my ability to
open my garage door from asterisk...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, August 07, 2005 1:46 PM
To: asterisk-users@lists.digium.com
Su
I'm seeing this same issue. The following message will popup on the console:
-- Attempting native bridge of Zap/1-1 and Zap/74-1
At the same time my call is briefly muted, I hear a quick DTMF tone, then it
unmutes. The whole process takes about 1.5 seconds. Is there any way to stop
these attemp
I'm trying to pass a 65K DATA call in one channel on my Digium
TE411P to another channel on a different span. Any idea what could keep this
call from going through?
-- Accepting call from '' to '5444' on channel 0/1, span 1
-- Executing Goto("Zap/1-1", "sendto-definity|5444|1") in
I use the TE110P to connect my Avaya Definity to my * via a TIE/PRI. I just
received my two TE411P's. w00T!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Maroney
Sent: Wednesday, July 20, 2005 5:46 PM
To: asterisk-users@lists.digium.com
Subject: [
I'm installing my new TE411P's but found in my new system,
/dev/zap/channel is missing. Anyone know off hand what actually creates that
file? Modprobe didn't complain when I loaded zaptel and wct4xxp modules.
Ztcfg complains too:
> ztcfg
Notice: Configuration file is /etc/zaptel.conf
line 0
I just discovered an 18 hour call to Brazil that was 60 seconds of
an employee calling a customer, then 18 hours and 47 minutes of background
noise in their office. The Cisco 7960's have an issue where you sometime
don't realize the phone is still off hook as was the case for this call.
You might look at the "r" options in the Dial command. Seems like one of
these should fit:
r: Generate a ringing tone for the calling party, passing no audio from the
called channel(s) until one answers. Use with care and don't insert this by
default into all your dial statements as you are killin
Has anyone figured out how to mimick a traditional bridged-appearance? My
guys like the ability to put a call on hold on line "3" and it's the same
call on line "3" on everyone else's phone.
___
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Asterisk-Users@lists.digium.com
I'm trying to setup
remote CDR logging, as directed by:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20odbc
Anyone have example
of what I need to change to make an asterisk server log on a remote mysql
server?
___
Asterisk-Use
I recently began using the curl cmd to do an external callerid
lookup on my own customer database. I've noticed certain lookups will cause
a crash and not show anything in the messages file or the console. The curl
command is connecting to an external webserver which has a oracle db
connect
If everyone running windoze had a xserver running, it would be easy... Just
have the * display a window on the users windoze box.
The most useful command I've found so far, it the curl(URL) command. I use
this to do a lookup on inbound callers ${CALLERIDNUM} and see if matches an
existing customer
I’m looking for a zaptel type device with
one (or more) FXO and one (or more) FXS port. Basically this guy would sit
in-line of your phone line (PCI card). Any suggestions? TDM400 would be
overkill.
___
Asterisk-Users mailing lis
Or have a small solar panel on the back of the phone. Stick it on the dash
of your car, assuming it doesn't burst into flames from heat; it should be
fully charged in an hour or two.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Sent: Sunday, May 15
Maybe try a version of redhat that was
released in the past 5 years? Seriously, why do you require RH7.3 over Fedora
or even RH 9?
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Thierry Wehr
Sent: Sunday, May 15, 2005 5:58 PM
To:
asterisk-users@lists.digium.com
S
oint -- you shouldn't have to tweak your dialplan because a
service only works "sometimes." That's just isn't good enough.
--Luki
On 5/14/05, Tim Connolly <[EMAIL PROTECTED]> wrote:
>
> Has anyone been watching and logging when broadvoice bec
Has anyone been watching and logging when
broadvoice becomes unstable? Is it only peak hours, or is it random? If its
somewhat consistant, I’d like to enforce some time of day routing in my
dialplan. Otherwise I may just close the account altogether…
__
Does this path already exist???
"/var/spool/asterisk/fax/2201001/”
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ma Zhiyong
Sent: Sunday, May 08, 2005 11:27
PM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] spandsp
Hi,
I
installed
Yeah, Broadvoice sucks, everybody cancel your service so I can use it! I
have yet to find another provider with as many free calls from a basic rate
with no strings attached (Learn from this Vonage!).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Disgrun
Can your receptionist handle 6 active
conversations? Once she transfers the call, it would disappear from those 6
lines.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Manjit Riat
Sent: Sunday, May 08, 2005 5:09 PM
To: 'Asterisk Users Mailing List -
Non-Commercia
This could be expanded to about any hardware. I can envision using this
instead of a callmanager to provide on the fly Cisco 7960 configs.
Good work wiki-ing this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham
Sent: Sunday, May 08, 2005 1:
I fought with my ata186 until I decided to start dorking with the
settings. I found no outbound faxes could be sent (fax handshake never could
complete) until I set the "AudioMode 0x00050005".
Basically this sets the ATA for fax mode which is documented on:
http://www.cisco.com/en/U
I've got three dual Xeon's running Redhat Enterprise 4 with 2.6.9
and CVS-HEAD from about a month ago. I didn't have any problems whatsoever,
other than the problems I blame on being reluctant to RTFM. No problems with
the SMP side whatsoever.
-Original Message-
From: [EMAIL PROTEC
I see the same error on codec 100 when I
try to rxfax. The faxes fail btw…
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomas Sia
Sent: Friday, May 06, 2005 12:27
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] unknown
RTP codec 72
can
Pass through has the same functionality as a modem with a "line" and a
"phone" connection. Line is where you plug in the dialtone, the dial passes
through the "phone" connection unless the card picks up (like a modem does).
I have a X100P clone that is setup as a passthrough. I've never seen a pas
Can someone send me an example of a CVS-head extension.conf
excerpt that utilizes the faxdetect and “fax” extension feature. I’m
tired of seeing these:
Apr 29 17:33:15 NOTICE[3541] chan_zap.c: Fax detected, but
no fax extension
___
Asterisk
Me thinks broadvoice needs to add a few more proxies to the US and
other hot spots...
I'd like to be able to accept calls from any of their proxies. I can
see us registering with all 3 and choosing an outbound using lowest latency.
okay, I'm out..
-Original Message--
Somebody correct me if I'm wrong here, but without reinvite being disabled,
I don't think the * can inject audio on the middle of the call.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Wednesday, May 04, 2005 11:41 PM
To: 'Asteris
Let's see your sip.conf and a sip show users.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Callum
McGillivray
Sent: Wednesday, May 04, 2005 11:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Problem with real
How about:
exten => 9111222,1,answer
exten => 9111222,2,wait(10)
exten => 9111222,3,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tulika Pradhan
Sent: Wednesday, May 04, 2005 11:29 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Do you have dial command in there with
option t or T? What’s the log say right before a call is “dropped”
?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Barton Fisher
Sent: Wednesday, May 04, 2005
11:13 PM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-
First off, yes, canreinvite=no would be a good choice.
Secondly, did you "make mpg123" from the asterisk source directory?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Wednesday, May 04, 2005 10:49 PM
To: [EMAIL PROTECTED]; 'Aste
I know somebody out there has a little perl
script that parses the cdr file into calls per hour and calls per month. Anyone
want to save me an hour? Please? My wife will thank you!
Thanks in advance,
Tim
<>___
I use MRTG to graph Active/Configured SIP channels and Active/Total
PRI/ZAP channels, but I don't monitor the up/down status. You could probably
write a little perl script to tail the logfile and watch for certain events,
then forward them by mail. Actually, I think I might do that too sinc
No.. but...
In their defense though, Cisco sold us a million dollars in routers
taunting they could handle the load. 6 months later we were trading them in
for Junipers because they were only able to handle the load as long as it
was low in packet per second count. Sometime they just don't
Or at least a username and password. How would * be able to differentiate
between the two clients?
Try ading:
Username=
Secrect=
For both...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Primoz Kragelj
Sent: Monday, May 02, 2005 2:39 PM
To: 'Asterisk U
Discussion
Subject: Re: [Asterisk-Users] voicemail volume with sipura 3000
On 5/2/05, Tim Connolly <[EMAIL PROTECTED]> wrote:
> Use wav, not gsm or wav49.
> /etc/asterisk/voicemail.conf
> ;
> ; Voicemail Configuration
> ;
> [general]
> format=wav
For future ref
http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
This will help.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Huddleston,
Robert
Sent: Monday, May 02, 2005 1:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Su
Use wav, not gsm or wav49.
/etc/asterisk/voicemail.conf
;
; Voicemail Configuration
;
[general]
format=wav
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of snacktime
Sent: Monday, May 02, 2005 1:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discu
I'm about to add a new wiki page, but wanted some input. This is a
list of locations where asterisk specific files are located. In my case,
this is RHES4 specific, I'm telling my backup software to backup:
/etc
/usr/src (yes I know, but there is a lot of custom crap in there)
/usr/lib/aste
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