Re: [asterisk-users] PoE budget
They're not silent, but they're not deafeningly loud. I doubt you'll ever find a silent PoE switch, since they have to supply far more power than your average switch. I wouldn't install one of these switches outside of a comms room if I could avoid it - but then again, that holds true for /any/ switched network. Jerry Jones wrote: On Jun 7, 2008, at 9:51 AM, Rob Hillis wrote: On the Linksys side, we have a load of SRW-224P switches out in the wild powering 24 Snom 370s (around 7W each) off each switch. Likewise, we sell these things by the bucket load and have no problems powering phones from all 24 ports. Just curious - have these ever gotten quieter? We installed one when they first came out and it was WAY to loud for an office environment, data center would be OK. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:484b1e9967791919312391! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad ringback tone on zap channel
The ringback is coming from the Zap channel, since that's the destination of the call. Therefore, the bad ring is more likely to be coming from the remote end. What type of line are you making the call to? Analogue? E1/T1? If it's analogue, I'd be guessing you have a faulty PSTN line. James Lamanna wrote: Hmm ok. This was a call from a SIP phone registered with Asterisk outbound on a Zap trunk. So would Asterisk or the phone be generating the ringback tone in that case? It also happens very intermittently (maybe 1 in 10 calls at most...) -- James Rob Hillis wrote: In my experience, the ringback you get over a zap channel (be it analogue or digital) is generated by the remote end, /not/ Zaptel. The ringback you get over a SIP or IAX2 channel is often generated by either Asterisk or the SIP/IAX2 device you're calling from. James Lamanna wrote: Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:484b09be67791587961402! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up, the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference with about 12 other people from around the USA. Bandwidth issues aside, will this work or will all the different latencies cause issues? Yea I know, I could just try it and find out but it is going to take alot of time to get everyones schedule to line up, I don't want to go through the trouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up, the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference with about 12 other people from around the USA. Bandwidth issues aside, will this work or will all the different latencies cause issues? Yea I know, I could just try it and find out but it is going to take alot of time to get everyones schedule to line up, I don't want to go through the trouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to set name of call wav recording file in outgoing/call file?
When I mv a file to /var/spool/asterisk/outgoing in order to place a call from a user extension that will always be recorded, what parameter do I set in the call file in order to specify an exact name for the wav file? This is on Trixbox and at the moment I'm considering setting an extra variable and calling through a new calling context that reads this variable and sets the recording name. -HJC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow-up Question Was: Question on DeadAGI
Hi,
i noticed a alot of mistake on what i did.
i have this macro
[macro-dialout-trunk]
exten =gt; s,1,Wait(1)
exten =gt; s,n,SetMusicOnHold(${ARG3})
exten =gt; s,Set(TIMEOUT(absolute)=${ARG4})
exten =gt; s,n,Dial(SIP/[EMAIL PROTECTED],30,t)
exten =gt; s,n,Hangup()
exten =gt; h,1,ResetCDR(w)
exten =gt; h,n,NoCDR()
exten =gt; h,n,DEADAGI(get-total.php)
[outbound-trunk-100]
exten =gt; _00.,1,AGI(call-compute.php)
exten =gt; _00.,n,GotoIf($[${CALLSTATUS} = 1]?80)
exten =gt; _00.,n,Hangup
exten =gt; _00.,80,Macro(dialout-trunk|${EXTEN}|intl-trunk|moh-100|${OUTTIME})
exten =gt; _00.,n,Hangup
I tried calling my mobile , call-compute.php was executed,i'm able to see
details i need for start accounting.
When i answer my phone and hangup, get-total is executed also.
My prob is ifnbsp; i cancel my the call on my mobile, ip phone keeps on
dialing it. How can i detect that the other end canceled the call?
Another is if i dial any number, even invalid ones, my script get-total.php
still thinks it is an answered call, so it still does deducting on the balance.
will really appreciate any help.nbsp; TIA.
--- On Sat, 6/7/08, Nhadie Ramos lt;[EMAIL PROTECTED]gt; wrote:
From: Nhadie Ramos lt;[EMAIL PROTECTED]gt;
Subject: Re: [asterisk-users] Question on DeadAGI
To: asterisk-users@lists.digium.com
Date: Saturday, June 7, 2008, 10:52 PM
Thanks to all the help. I think i have it now. I reset the CDR on the hangup
channel.
[macro-dialout-trunk]
exten =gt; s,1,Wait(1)
exten =gt; s,n,Dial(SIP/[EMAIL PROTECTED],30,t)
exten =gt; s,n,Hangup()
exten =gt; h,1,ResetCDR(w)
exten =gt; h,n,NoCDR()
exten =gt; h,n,DEADAGI(get-total.php)
AGI Rx lt;lt; EXEC Noop ROWCOUNT=1
nbsp;nbsp;nbsp; -- AGI Script Executing Application: (Noop) Options:
(ROWCOUNT=1)
AGI Tx gt;gt; 200 result=0
AGI Rx lt;lt; EXEC Noop BILLSEC=21
nbsp;nbsp;nbsp; -- AGI Script Executing Application: (Noop) Options:
(BILLSEC=21)
now i can see my billsec. thanks again for all the help.
regards,
nhadie
--- On Sat, 6/7/08, Nhadie Ramos lt;[EMAIL PROTECTED]gt; wrote:
From: Nhadie Ramos lt;[EMAIL PROTECTED]gt;
Subject: Re: [asterisk-users] Question on DeadAGI
To: Asterisk Users Mailing List - Non-Commercial Discussion
lt;asterisk-users@lists.digium.comgt;
Date: Saturday, June 7, 2008, 10:39 PM
Hi Sir,
I tried it this way, and now i can see my DEADGI being called next prob is
onnbsp; that script i query the cdr table with the uniqueid. tried counting
the row result first , and result was 0.
how can i make sure that it was already at the CDR table before i call my agi?
i tried to use ResetCDR() and also without ResetCDR() but still 0 result on the
row.
but when i query manully on the mysql console, i can see the cll was logged.
Thank You
[macro-dialout-trunk]
exten =gt; s,1,Wait(1)
exten =gt; s,n,Dial(SIP/[EMAIL PROTECTED],30,t)
exten =gt; s,n.ResetCDR()
exten =gt; s,n,Hangup
exten =gt; h,1,DEADAGI(get-total.php)
---
On Sat, 6/7/08, Lenz
lt;[EMAIL PROTECTED]gt; wrote:
From: Lenz lt;[EMAIL PROTECTED]gt;
Subject: Re: [asterisk-users] Question on DeadAGI
To: Asterisk Users Mailing List - Non-Commercial Discussion
lt;asterisk-users@lists.digium.comgt;
Date: Saturday, June 7, 2008, 12:50 PM
You should use it on the hang-up extension and only after the channel is
technically dead.
It works fine for that.
l.
On Sat, 07 Jun 2008 01:25:37 +0200, Nhadie Ramos lt;[EMAIL PROTECTED]gt;
wrote:
gt; Hi,
gt;
gt; How can i get the deadAGI to work at this scenario
gt;
gt; Basically when someonc calls international,amp;nbsp; i will get the
gt; remaining balance using AGI get-available.php.
gt;
gt; but after the call i would like to get the usage by calling
gt; get-usage.php so
i
can update users balance, but looking at the debug it
gt; seems the AGI was not called. is there som
gt;
gt; exten =amp;gt; _00.,1,AGI(get-available.php)
gt; exten =amp;gt; _00.,n,GotoIf($[${CALLSTATUS} =
1]?70)
gt; exten =amp;gt; _00.,n,GotoIf($[${CALLSTATUS} =
2]?80)
gt; exten =amp;gt; _00.,70,Dial(SIP/[EMAIL PROTECTED])
gt; exten =amp;gt; _00.,n,Hangup
gt; exten =amp;gt; _00.,n,DEADAGI(get-usage.php)
gt; exten =amp;gt; _00.,80,Busy
gt; exten =amp;gt; _00.,n,Hangup
gt;
gt;
gt; Regards,
gt; Nhadie
gt;
gt;
gt;
--
Loway Research - Home of QueueMetrics
http://queuemetrics.com
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Re: [asterisk-users] MiixMonitor filename for queue calls.
I am using the following entry to define my filename
exten =
8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH},G
MT+8,%C%y%m%d%H%M)})
This will display QUEUE-NOC (Caller ID number) (and time stamp)
I would also like to add the answering Agent ID to the file name. Any idea
what this variable name is?
Thank you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Smith
Sent: Saturday, June 07, 2008 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MiixMonitor filename for queue calls.
Hi Ed,
Glad to see you figured out your problem. I'm not sure what the
differences are between your config and mine, but maybe this will help
others too.
I add and remove my agents from the queue. So my agents.conf file is
just the presistentagens=yes. Also I just run the command in the dial
plan like below which saved mine items just fine. No configurations in
the queue.conf file for the monitor type.
exten = 852,n,MixMonitor(/mercury/recordings/holding/${UNIQUEID}.gsm|b|)
From there, in the hangup extension, I run a php script to take the CDR
record and the file (rename it of course to
queue-extension-callerid-callid-timestamp.gsm), and place it into the
agents folder and the database for our agents/supervisors to review or
download them.
Kevin
Ed Nunez wrote:
Can anyone give me input on the following issue?
I have a queue with MixMonitor enabled.
This is also enabled in agents.conf.
On my extensions.conf, I am setting the monitor filename as fillows,
although I see the filename as desired in the console as I make my
test call, the system is only using the default file name to save the
mixmonitor file (agented + uniqueID)
Agents.conf
[general]
persistentagents=yes
[agents]
maxlogintries=3
musiconhold = default
updatecdr=yes
recordagentcalls=yes
recordformat=wav49
urlprefix=http://pbx.netoneint.com/calls/
savecallsin=/var/calls
agent = 1000,1000,Ed Test1
agent = 1001,1001,Ed Test2
queues.conf
[noi-noc]
monitor-format = wav49
monitor-type = MixMonitor
member = Agent/1001
member = Agent/1000
extensions.conf
exten =
8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH)
exten = 8484,1,answer
exten = 8484,2,Queue(noi-noc)
Console output
-- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1,
MONITOR_FILENAME=QUEUE-NOC-4073844200-Fri Jun 6 15:06:38 2008) in
new stack
-- Executing [EMAIL PROTECTED]:2] Queue(Zap/1-1, noi-noc) in
new stack
-- Started music on hold, class 'default', on Zap/1-1
-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing [EMAIL PROTECTED]:1]
Dial(Local/[EMAIL PROTECTED],2, SIP/1658) in new stack
-- Called 1658
-- SIP/1658-087e7610 is ringing
-- Agent/1001 is ringing
-- SIP/1658-087e7610 answered Local/[EMAIL PROTECTED],2
-- Agent/1001 answered Zap/1-1
-- Stopped music on hold on Zap/1-1
[Jun 6 15:06:40] WARNING[3976]: app_queue.c:3014 try_calling: The
device state of this queue member, Agent/1001, is still 'Not in Use'
when it probably should not be! Please check UPGRADE.txt for correct
configuration settings.
== Begin MixMonitor Recording Zap/1-1
== Spawn extension (numberplan-custom-3, 1658, 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'
== Spawn extension (incoming-att, 8484, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
== End MixMonitor Recording Zap/1-1
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--
Kevin Smith
---
Mercury Network
Technical Support
Phone: 989.837.3790
Toll Free: 888.866.4638
www.mercury.net
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Re: [asterisk-users] MeetMe Limits
12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up, the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference with about 12 other people from around the USA. Bandwidth issues aside, will this work or will all the different latencies cause issues? Yea I know, I could just try it and find out but it is going to take alot of time to get everyones schedule to line up, I don't want to go through the trouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 via MAX TNT ??
Alex.. would you point us in the right direction, or perhaps consider sending a sample max tnt config reflecting how this is done? Thank you.. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, June 06, 2008 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? I believe the ISDN call plan can be configured as part of the trunk group / route. Joe Carroll wrote: We talked with the LEC and discovered that 911 has to be sent as Unknown instead of National... Anyone know how we might tell the TNT to do this? Apparently, according to the carrier, all Special Access Numbers, 411, 611, 911, etc require this special code ??? PRI DEBUG FOLLOWS: --nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08 Bearer_Cap 80 90 A2 (Speech,Rate=64K) Channel_Id A1 83 83 (Pref,Intf=0,Chan=3) Calling_Num (National,Restricted,Failed) 229317 Called_Num (National) 911 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Thursday, June 05, 2008 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
Matthew, Nothing as serious as a broken email system. I forgot to attach the attachment. Sorry about that. Here it is. Thanks, John On Sat, Jun 7, 2008 at 8:12 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: John Morey wrote: I switch the wires in lines 5-8 (i.e. reversed tip and ring) and reran fxotune to tune the lines. fxotune.conf ended up looking exactly the same as before the change. Since I was expecting/hopping to see a change but did not I switched everything back to the way it was. Is there a way to test the lines, using a multi-meter maybe, to tell if the tip and ring are correct or reversed? After putting things back I reran fxotune to get the verbose output. It, foxtune.out.gz, is attached. fxotune seems to have had a better time with It seems that one way or another the attachment didn't go through. Can you email the tarball to me directly or post it to a website? Thanks, Matthew Fredrickson line 7 during this run. fxotune.conf now contains: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=4,0,0,0,0,0,0,0,0 8=7,255,251,251,2,255,255,1,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 I tried calling directly into the lines above and it seems lines 5,6,8 have much more echo than lines 7,9,10. So just for fun I edited fxotune.conf to the following and reloaded (fxotune -s) it: 5=5,0,0,0,0,0,0,0,0 6=5,0,0,0,0,0,0,0,0 7=4,0,0,0,0,0,0,0,0 8=5,0,0,0,0,0,0,0,0 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 Unless I am just spacing out the echo on 5,6,8 seems less now. I really have no idea what is going on. John On Fri, Jun 6, 2008 at 1:31 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: John Morey wrote: Tilghman, Thanks for the pointer. I'll check this tomorrow and let you know. Also, I would like to see the output without the -d flag and with the -v flag. This will output a lot of data (the echo ratio for every possible coefficient setting it has tried per port). Matthew Fredrickson John On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users foxtune_v.out.gz Description:
Re: [asterisk-users] 911 via MAX TNT ??
Joe,
I am not sure if your 911 call is incoming or outgoing on PRIs.
#assume you have a T1 in {1 1 1}
Read t1 { 1 1 1}
Set line send-dnis-type-of-number ?
You will see options. Some 911 providers support media-before-connect. Plz
make sure your all of TNT support 183.
Hope it can help you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Sunday, June 08, 2008 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Alex.. would you point us in the right direction, or perhaps consider
sending a sample max tnt config reflecting how this is done? Thank you..
-Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Friday, June 06, 2008 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
I believe the ISDN call plan can be configured as part of the trunk
group / route.
Joe Carroll wrote:
We talked with the LEC and discovered that 911 has to be sent as Unknown
instead of National... Anyone know how we might tell the TNT to do this?
Apparently, according to the carrier, all Special Access Numbers, 411, 611,
911, etc require this special code ???
PRI DEBUG FOLLOWS:
--nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08
Bearer_Cap 80 90 A2 (Speech,Rate=64K)
Channel_Id A1 83 83 (Pref,Intf=0,Chan=3)
Calling_Num (National,Restricted,Failed) 229317
Called_Num (National) 911
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Thursday, June 05, 2008 6:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Yes, we are using the max tnt to aggregate several PRIs both inbound and
outbound from multiple carriers. This PRI is a normal two way circuit that
a carrier would deliver to an end user...
From: [EMAIL PROTECTED]
[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
[EMAIL PROTECTED]
Sent: Thursday, June 05, 2008 9:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 911 via MAX TNT ??
On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote:
On June 4, 2008 06:20:57 pm Joe Carroll wrote:
Interestingly enough, on the syslog messages from the TNT we are seeing
Called = 911, Q850 Cause = 28, SIP Response = 484
That really looks like the switch that the TNT is talking to is rejecting
the
number, not the TNT...
Remember: 9-1-1 is a *dialling pattern*, not a *directory number*;
it's entirely possible that trunks wouldn't accept it directly.
This *is* a *LEC* trunk, right?
Cheers,
-- jra
--
Jay R. Ashworth Baylink
[EMAIL PROTECTED]
Designer The Things I Think RFC
2100
Ashworth Associates http://baylink.pitas.com '87
e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647
1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Joseph Stalin)
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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Re: [asterisk-users] MeetMe Limits
I've got to agree.. I've never given it much thought either... All of my calls are SIP/IAX based, coming in from the PSTN from a peer like that too.. I've never tracked the total number of conference users... But I'll bet we've hit at least 10.. And I've never seen the CPU go above 10%.. And that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it will be setup-specific.. So I would look at your CPU and memory stats, and run some tests and monitor that.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 08 June 2008 16:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Limits 12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up,the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference withabout 12 other people from around the USA. Bandwidth issues aside, willthis work or will all the different latencies cause issues? Yea I know,I could just try it and find out but it is going to take alot of timeto get everyones schedule to line up, I don't want to go through thetrouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 via MAX TNT ??
We are providing voip services, these 911 calls are going out from our
subscribers to the lec to be delivered to the 911 PSAP.. Would this apply in
that scenario ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leon Sun
Sent: Sunday, June 08, 2008 3:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Joe,
I am not sure if your 911 call is incoming or outgoing on PRIs.
#assume you have a T1 in {1 1 1}
Read t1 { 1 1 1}
Set line send-dnis-type-of-number ?
You will see options. Some 911 providers support media-before-connect. Plz
make sure your all of TNT support 183.
Hope it can help you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Sunday, June 08, 2008 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Alex.. would you point us in the right direction, or perhaps consider
sending a sample max tnt config reflecting how this is done? Thank you..
-Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Friday, June 06, 2008 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
I believe the ISDN call plan can be configured as part of the trunk
group / route.
Joe Carroll wrote:
We talked with the LEC and discovered that 911 has to be sent as Unknown
instead of National... Anyone know how we might tell the TNT to do this?
Apparently, according to the carrier, all Special Access Numbers, 411, 611,
911, etc require this special code ???
PRI DEBUG FOLLOWS:
--nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08
Bearer_Cap 80 90 A2 (Speech,Rate=64K)
Channel_Id A1 83 83 (Pref,Intf=0,Chan=3)
Calling_Num (National,Restricted,Failed) 229317
Called_Num (National) 911
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Thursday, June 05, 2008 6:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Yes, we are using the max tnt to aggregate several PRIs both inbound and
outbound from multiple carriers. This PRI is a normal two way circuit that
a carrier would deliver to an end user...
From: [EMAIL PROTECTED]
[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
[EMAIL PROTECTED]
Sent: Thursday, June 05, 2008 9:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 911 via MAX TNT ??
On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote:
On June 4, 2008 06:20:57 pm Joe Carroll wrote:
Interestingly enough, on the syslog messages from the TNT we are seeing
Called = 911, Q850 Cause = 28, SIP Response = 484
That really looks like the switch that the TNT is talking to is rejecting
the
number, not the TNT...
Remember: 9-1-1 is a *dialling pattern*, not a *directory number*;
it's entirely possible that trunks wouldn't accept it directly.
This *is* a *LEC* trunk, right?
Cheers,
-- jra
--
Jay R. Ashworth Baylink
[EMAIL PROTECTED]
Designer The Things I Think RFC
2100
Ashworth Associates http://baylink.pitas.com '87
e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647
1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Joseph Stalin)
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Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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[asterisk-users] Asterisk can handle only 200 to 300 SIP device registrations
Hi All, Is this still the cause in 1.4 and 1.6 as per: http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration. Do people recommend OpenSER in front for deployments bigger than 300 end points? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Diverted Call Information on PRI
Hi Everyone, I have an asterisk installation nicely working, I'm running 1.4; I'm hooked up to a BT Pri and an old Panasonic PBX using a RedFone foneBridge, everything is sweet. I have voicemail sorted for all internal users; the call gets routed to a SipX box and then into a Microsoft UM Server which gives us fully unified voicemail, fax and e-mail... Now onto the fun part. I currently have my mobile phones call diversion set up to route calls through to one of the spare DDI's we have in our block instead of my service providers voicemail; so when a call rings out on my mobile, my DDI or my internal number; they hit the same mailbox... Now I'd obviously like to offer this functionality to some of my users, however I dont want to have to dedicate a DDI to each mailbox. Is there any way I can tell if a call is a diversion from an external phone if it comes in on our PRI? If so, is there also any way I can find out what number the call was diverted from? I've done some logging with PRI intense debug; and I cant seem to see anything about it being a divert; but I could be missing something. If only the first part can be achieved then I'll just divert anyones mobiles calls to their own DDI and then in the dialplan for their DDI's I can check if it's a diversion and direct the call to the appropriate mailbox. Ideally though I'd like to have a dedicated mailbox number that does this checking, but that would obviously require data regarding which mobile phone the call was originally intended for... Is this a pure pipe dream? does PRI carry call diversion information? Thanks Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diverted Call Information on PRI
Mike Hardman wrote:
Is there any way
I can tell if a call is a diversion from an external phone if it comes
in on our PRI? If so, is there also any way I can find out what number
the call was diverted from? I've done some logging with PRI intense
debug; and I cant seem to see anything about it being a divert; but I
could be missing something.
I seem to remember ${RDNIS} back in the 1.2 days had this information.
I think it's called something like ${CALLERID(RDNIS)} now, although
you'll want to check to be sure.
It'll give you the number the caller actually dialed before the
diversion redirected them to your PRI.
Trevor
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Re: [asterisk-users] MeetMe Limits
Hello, We routinely run meetme with over 140 ULAW channels connected to 70 meetme rooms with no issues on an Intel Core 2 Quad core CPU. The major factor in capacity would be your CPU and RAM capacity. If you have at least a base-level P4 you don't need to worry about 12 participants. MATT--- On 6/8/08, Adrian Marsh [EMAIL PROTECTED] wrote: I've got to agree.. I've never given it much thought either... All of my calls are SIP/IAX based, coming in from the PSTN from a peer like that too.. I've never tracked the total number of conference users... But I'll bet we've hit at least 10.. And I've never seen the CPU go above 10%.. And that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it will be setup-specific.. So I would look at your CPU and memory stats, and run some tests and monitor that.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 08 June 2008 16:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Limits 12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up,the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference withabout 12 other people from around the USA. Bandwidth issues aside, willthis work or will all the different latencies cause issues? Yea I know,I could just try it and find out but it is going to take alot of timeto get everyones schedule to line up, I don't want to go through thetrouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
Matt, Could you share the CPU usage, memory, and load average in the scenario you describe? What type of ULAW channels (SIP,DAHDI,IAX), or does it not matter? Thanks, Steve Totaro On Sun, Jun 8, 2008 at 5:29 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, We routinely run meetme with over 140 ULAW channels connected to 70 meetme rooms with no issues on an Intel Core 2 Quad core CPU. The major factor in capacity would be your CPU and RAM capacity. If you have at least a base-level P4 you don't need to worry about 12 participants. MATT--- On 6/8/08, Adrian Marsh [EMAIL PROTECTED] wrote: I've got to agree.. I've never given it much thought either... All of my calls are SIP/IAX based, coming in from the PSTN from a peer like that too.. I've never tracked the total number of conference users... But I'll bet we've hit at least 10.. And I've never seen the CPU go above 10%.. And that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it will be setup-specific.. So I would look at your CPU and memory stats, and run some tests and monitor that.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 08 June 2008 16:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Limits 12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up,the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference withabout 12 other people from around the USA. Bandwidth issues aside, willthis work or will all the different latencies cause issues? Yea I know,I could just try it and find out but it is going to take alot of timeto get everyones schedule to line up, I don't want to go through thetrouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diverted Call Information on PRI
I'll get right on hunting about RDNIS, thank you VERY much! :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk can handle only 200 to 300 SIP device registrations
On Sun, Jun 8, 2008 at 4:42 PM, Gavin Henry [EMAIL PROTECTED] wrote: Hi All, Is this still the cause in 1.4 and 1.6 as per: http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration. Do people recommend OpenSER in front for deployments bigger than 300 end points? I have heard that FreeSwitch is better at handling higher volumes of SIP. Haven't tried it myself. OpenSER is what I have used, but I will be evaluating FreeSwitch very carefully (OpenSER is a little difficult to get configured), even if it just a man in the middle to Asterisk. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
Hello, The load is usually quite high because this is VICIDIAL inbound call center traffic with full Asterisk-based recording. On a system with 70-80 Meetme rooms running with 2 participants each doing full Asterisk-based recording in each Meetme room the loadavg stays between 2.00-4.00 on a Quad-core Intel core 2 Quad processor with 4GB RAM. I have three systems like this in place at different call centers and the load is consistent for all three of them. Usually we put less load on a single server, but these were inbound-only scenarios which is less load than outbound. MATT--- On 6/8/08, Steve Totaro [EMAIL PROTECTED] wrote: Matt, Could you share the CPU usage, memory, and load average in the scenario you describe? What type of ULAW channels (SIP,DAHDI,IAX), or does it not matter? Thanks, Steve Totaro On Sun, Jun 8, 2008 at 5:29 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, We routinely run meetme with over 140 ULAW channels connected to 70 meetme rooms with no issues on an Intel Core 2 Quad core CPU. The major factor in capacity would be your CPU and RAM capacity. If you have at least a base-level P4 you don't need to worry about 12 participants. MATT--- On 6/8/08, Adrian Marsh [EMAIL PROTECTED] wrote: I've got to agree.. I've never given it much thought either... All of my calls are SIP/IAX based, coming in from the PSTN from a peer like that too.. I've never tracked the total number of conference users... But I'll bet we've hit at least 10.. And I've never seen the CPU go above 10%.. And that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it will be setup-specific.. So I would look at your CPU and memory stats, and run some tests and monitor that.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 08 June 2008 16:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Limits 12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up,the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference withabout 12 other people from around the USA. Bandwidth issues aside, willthis work or will all the different latencies cause issues? Yea I know,I could just try it and find out but it is going to take alot of timeto get everyones schedule to line up, I don't want to go through thetrouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
Re: [asterisk-users] MeetMe Limits
Forgot to address your second question. DAHDI, that's a good one :) The channel type doesn't seem to matter. One has all agents on Zap channels through channelbanks with all calls coming in over IAX and monitoring done through SIP. One has all SIP agents with all calls coming in over SIP trunks, and another has SIP agents with calls coming in over Zap T1 channels. MATT--- On 6/8/08, Matt Florell [EMAIL PROTECTED] wrote: Hello, The load is usually quite high because this is VICIDIAL inbound call center traffic with full Asterisk-based recording. On a system with 70-80 Meetme rooms running with 2 participants each doing full Asterisk-based recording in each Meetme room the loadavg stays between 2.00-4.00 on a Quad-core Intel core 2 Quad processor with 4GB RAM. I have three systems like this in place at different call centers and the load is consistent for all three of them. Usually we put less load on a single server, but these were inbound-only scenarios which is less load than outbound. MATT--- On 6/8/08, Steve Totaro [EMAIL PROTECTED] wrote: Matt, Could you share the CPU usage, memory, and load average in the scenario you describe? What type of ULAW channels (SIP,DAHDI,IAX), or does it not matter? Thanks, Steve Totaro On Sun, Jun 8, 2008 at 5:29 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, We routinely run meetme with over 140 ULAW channels connected to 70 meetme rooms with no issues on an Intel Core 2 Quad core CPU. The major factor in capacity would be your CPU and RAM capacity. If you have at least a base-level P4 you don't need to worry about 12 participants. MATT--- On 6/8/08, Adrian Marsh [EMAIL PROTECTED] wrote: I've got to agree.. I've never given it much thought either... All of my calls are SIP/IAX based, coming in from the PSTN from a peer like that too.. I've never tracked the total number of conference users... But I'll bet we've hit at least 10.. And I've never seen the CPU go above 10%.. And that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it will be setup-specific.. So I would look at your CPU and memory stats, and run some tests and monitor that.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 08 June 2008 16:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Limits 12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up,the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference withabout 12 other people from around the USA. Bandwidth issues aside, willthis work or will all the different latencies cause issues? Yea I know,I could just try it and find out but it is going to take alot of timeto get everyones schedule to line up, I don't want to go through thetrouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL
Re: [asterisk-users] Logitech DiNovo Mini keyboard with myth
On Fri, Jun 6, 2008 at 9:03 PM, OCG Technical Support [EMAIL PROTECTED] wrote: Has anyone create the necessary config/kbd file to allow the DiNovo mini to work well with myth? (Mapped all of the multimedia buttons etc) Is that in extensions.conf or chan_dinovo.conf? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk On Public IP
I have installed Asterisk. I want friends to connect to my asterisk server from their SIP Phones are talk to me. I have tried two ways 1.) Have the Asterisk server run within the firewall, opened all the ports for that server in firewall port forwarding, does not work (One way audio issue). I have heard many thing about NAT issue etc. I have taken care of all the issues as suggested, never worked for me. Then I connected the linux server (asterisk server) directly to the internet (no firewall in between). The SIP phones would not connect to the server. It give 408 error. Any idea how to fix the issue? Thanks, SR. _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk video alternatives
Hello, I am also planning to implement Video Conf. There is AppConference you could evaluate. Curious, Did you get any response from anyone on your question? Thanks, SR. To: asterisk-users@lists.digium.com From: [EMAIL PROTECTED] Date: Fri, 6 Jun 2008 00:24:24 +0200 Subject: [asterisk-users] Asterisk video alternatives Hi. At the company I work for, we use Asterisk to communicate with our offices all around the world. Recently, I've been asked to implement a video conference system, asterisk compatible/interoperable as possible. It's preferred but not required to be an open source solution. What options do I have? wich would you suggest me to try? Any good experience with any of these systems? Thanks a lot. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Find hidden words, unscramble celebrity names, or try the ultimate crossword puzzle with Live Search Games. Play now! http://g.msn.ca/ca55/212___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.28 + Realtime Queues - Thinks Queue is empty
Gentlemen, I have a particularly strange problem, just started happening. One of my clients is running Asterisk 1.2.28 and has mysql realtime queues. We log in a member, and then place a test call to the 0 queue but since joinempty is set to no, and Asterisk thinks the queue has no members, they're kicked out. WHY would Asterisk think the queue has no members? here's some relevant output: carp*CLI show queue 0 0has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: 9001 (Invalid) has taken no calls yet No Callers -- Executing Answer(SIP/pri-006ba540, ) in new stack -- Executing Wait(SIP/pri-006ba540, 1) in new stack -- Executing Dial(SIP/pri-006ba540, IAX2/127.0.0.1/[EMAIL PROTECTED]) in new stack -- Called 127.0.0.1/[EMAIL PROTECTED] -- Accepting AUTHENTICATED call from 127.0.0.1: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (ulaw|gsm), priority = mine -- Call accepted by 127.0.0.1 (format gsm) -- Format for call is gsm -- Executing Goto(IAX2/carp-12321, nwi|1) -- Goto (inbound,nwi,1) -- Executing AGI(IAX2/carp-12321, recfile) -- Launched AGI Script /var/lib/asterisk/agi-bin/recfile -- AGI Script recfile completed, returning 0 -- Executing MixMonitor(IAX2/carp-12321, call_2008060834.wav) == Begin MixMonitor Recording IAX2/carp-12321 -- Executing Set(IAX2/carp-12321, CDR(userfield)=call_2008060834.wav) -- Executing Goto(IAX2/carp-12321, carp-nwi|nwi-main|1) -- Goto (carp-nwi,nwi-main,1) -- Executing Answer(IAX2/carp-12321, ) -- IAX2/carp-1957 answered SIP/pri-006ba540 -- Executing Background(IAX2/carp-12321, silence/1) -- Playing 'silence/1' (language 'en') -- Executing Set(IAX2/carp-12321, __ACCOUNTCODE=12129889551212988954.2) -- Executing Read(IAX2/carp-12321, act0|ivr/new/nwi-welcome|1) -- Accepting a maximum of 1 digits. -- Playing 'ivr/new/nwi-welcome' (language 'en') -- User entered '0' -- Executing gotoif(IAX2/carp-12321, 0?carp-nwi|nwi-trans|1) -- Executing gotoif(IAX2/carp-12321, 1?carp-nwi|nwi-oper|1) -- Goto (carp-nwi,nwi-oper,1) -- Executing playback(IAX2/carp-12321, ivr/nwi-operator) -- Playing 'ivr/nwi-operator' (language 'en') -- Executing Queue(IAX2/carp-12321, 0|t) -- Executing Goto(IAX2/carp-12321, carp-nwi|nwi-main|3) -- Goto (carp-nwi,nwi-main,3) -- Executing Set(IAX2/carp-12321, __ACCOUNTCODE=12129889651212988954.2) -- Executing Read(IAX2/carp-12321, act0|ivr/new/nwi-welcome|1) -- Accepting a maximum of 1 digits. -- Playing 'ivr/new/nwi-welcome' (language 'en') -- Hungup 'IAX2/carp-1957' == Spawn extension (pri-inbound, s, 3) exited non-zero on 'SIP/pri-006ba540' -- User disconnected -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
