Re: [asterisk-users] Question about Do Not Disturb
On Thu, 26 Feb 2009, Haim Dimer wrote:
Hello,
Some of my users have phones lacking a DND button. I need to provide
an extension they can dial that will put them in DND, i.e. tell the
server not to send them any calls until they get off the DND.
I've researched it for almost 3 days now and tried a range of
configurations. I'm hoping somebody here has an answer. Currently, I
have this in extensions.conf
[app-dnd-on]
exten = *78,1,Answer
exten = *78,n,NoOp(${CALLERID(num)} channel ${CHANNEL} is going on
DND ACTIVE)
exten = *78,n,Set(DB(DND/${CALLERID(num)})=On)
exten = *78,n,UserEvent(ASTDB|Family: DND^Channel: ${CHANNEL} ^Value:
On)
exten = *78,n,Playback(do-not-disturbactivated)
exten = *78,n,Hangup
[app-dnd-off]
exten = *79,1,Answer
exten = *79,n,NoOp(${CALLERID(num)} is going OFF DND)
exten = *79,n,DBdel(DND/${CALLERID(num)})
exten = *79,n,UserEvent(ASTDB|Family: DND^Channel: ${CHANNEL} ^Value:
^)
exten = *79,n,Playback(do-not-disturbde-activated)
exten = *79,n,Hangup
Using the above config, if I dial *78 I hear Allison's voice telling
me that do not disturb is activated but I can still be called (either
directly or as part of a queue). BTW, there are many people on the
wiki stuck with the same problem :
http://www.voip-info.org/wiki/index.php?page_id=787tk=c7f21c26a40ee72393d7comments_page=1
I've no idea what your UserEvent is doing other than adding more into the
astdb, but you've not done anything to actually act on the astDB values.
All you're doing here is changing an astDB value on or off. Asterisk
doesn't look at the astDB when it places a call - thats for you to do.
In your call-handling code, you need to read the astDB value then decided
whether to dial the extension or not - and you're not doing this, or if
you are you haven't put the code here.
So I have a rather large macro that handles internal calls and it has a
section for DND:
; Check for Do Not Disturb
exten = s,n,Set(DND=${DB(${MACRO_EXTEN}/doNotDisturb)})
exten = s,n,GotoIf(${DND}?:doneDoNotDisturb)
exten = s,n,Wait(90)
exten = s,n,Hangup()
exten = s,n(doneDoNotDisturb),Noop(Carrying on after DO NOT DISTURB Check)
So in my case, if an extension has the doNotDisturb flag set
(manipulated by code similar to your above), then it basically keeps the
caller ringing for 90 seconds then hands up...
Gordon
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Re: [asterisk-users] Problems with Outbound Calls
Hey, thanks for the help David, Tzafrir. Lots of config tips there :-) We managed to find a fix through the following (For anyone who's interested): Running /sbin/ztcfg -vv to configure Zaptel initially resulted in an error of: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' We then Chmodded everything under /dev/zap/ , rebooted and almost fell off our chairs when it worked! We were initially on the impression that Zaptel is only used with Analogue - can anyone verify this? YK ___ -- 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] call file concurrency
2009/2/27 Bill Michaelson b...@cosi.com Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? Afaik only by limiting the number of call files in the directory. ___ -- 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] Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available
Hi, I am trying to log queue_log to odbc (MS SQL) I have res_odbc.conf configured and modules.conf have preload = res_odbc.so preload = res_config_odbc.so extconfig.conf has queue_log = odbc,asterisk. When I start asterisk I get the following messages. The important one being: Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available I can see that res_odbc is loading and registering Config Engine odbc, but that's after logger has started. Any clue what I am doing wrong? with regards, raj Asterisk 1.6.0.5, Copyright (C) 1999 - 2008 Digium, Inc. and others. snip = == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found == Binding queue_log to odbc/asterisk/queue_log Started Asterisk Event Logger Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available Asterisk Event Logger Started /var/log/asterisk/event_log 3 modules will be loaded. Connecting sqlserver Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': == Found == Parsing '/etc/asterisk/res_odbc.conf': == Found res_odbc: Connected to sqlserver [DSN_NAME] Registered ODBC class 'sqlserver' dsn-[DSN_NAME] res_odbc loaded. res_odbc.so = (ODBC resource) Registered Config Engine odbc ___ -- 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] incoming call problem
paste your sip.conf. David 2009/2/26 michel freiha mich...@gmail.com Dear All, I have created an inbound context in SIP .conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with OpenSIPS and cal success..Any suggestion here? Thanks ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] incoming call problem
Dear David, Please find on http://pastebin.com/m69b8559d my sip.conf file Thanks a lot On Fri, Feb 27, 2009 at 1:05 PM, David fire ddf...@gmail.com wrote: paste your sip.conf. David 2009/2/26 michel freiha mich...@gmail.com Dear All, I have created an inbound context in SIP .conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with OpenSIPS and cal success..Any suggestion here? Thanks ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Problems with Outbound Calls
Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' We then Chmodded everything under /dev/zap/ , rebooted and almost fell off our chairs when it worked! By right, if the problem is due to this error, you should see a permission error message in /var/log/asterisk/messages. What it means is the directory permissions might be wrong somewhere in the beginning. This may not be related to your original warning. Warning [2630]: config.c:768 process_text_line: Unknown Directive at line 231 of /etc/asterisk/../zaptel.conf We were initially on the impression that Zaptel is only used with Analogue – can anyone verify this? No, it is responsible for PRI channels as well. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wye-khe Kwok Sent: Friday, 27 February 2009 9:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problems with Outbound Calls Hey, thanks for the help David, Tzafrir. Lots of config tips there ☺ We managed to find a fix through the following (For anyone who’s interested): Running /sbin/ztcfg –vv to configure Zaptel initially resulted in an error of: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' We then Chmodded everything under /dev/zap/ , rebooted and almost fell off our chairs when it worked! We were initially on the impression that Zaptel is only used with Analogue – can anyone verify this? YK ___ -- 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] Odd Read App Issues - RESOLVED
FYI to everyone... It was an issue on Vitelity's end on the gateway I was assigned to. They switched me, and it's working fine now. -- Regards, Robert Broyles Brent Davidson wrote: Robert Broyles wrote: I turned on DTMF debugging. It looks like the extra digits coming in are less than the minimum duration of 100ms Anyone know how to force that minimum duration? [Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '1' received on SIP/carrier-c4022740 [Feb 26 12:15:07] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '1' on SIP/carrier-c4022740 [Feb 26 12:15:07] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '1' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:07] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '1' on SIP/carrier-c4022740 [Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '1' received on SIP/carrier-c4022740 [Feb 26 12:15:07] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '1' on SIP/carrier-c4022740 [Feb 26 12:15:07] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '1' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:07] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '1' on SIP/carrier-c4022740 [Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '1' received on SIP/carrier-c4022740 [Feb 26 12:15:07] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '1' on SIP/carrier-c4022740 [Feb 26 12:15:07] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '1' received on SIP/carrier-c4022740, duration 20 ms [Feb 26 12:15:07] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '1' on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '2' received on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '2' on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '2' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '2' on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '2' received on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '2' on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '2' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '2' on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '2' received on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '2' on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '2' received on SIP/carrier-c4022740, duration 20 ms [Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '2' on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '3' received on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '3' on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '3' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '3' on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '3' received on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '3' on SIP/carrier-c4022740 [Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '3' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '3' on SIP/carrier-c4022740 [Feb 26 12:15:09] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '4' received on SIP/carrier-c4022740 [Feb 26 12:15:09] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '4' on SIP/carrier-c4022740 [Feb 26 12:15:09] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '4' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:09] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '4' on SIP/carrier-c4022740 [Feb 26 12:15:09] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '4' received on SIP/carrier-c4022740 [Feb 26 12:15:09] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '4' on SIP/carrier-c4022740 [Feb 26 12:15:09] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '4' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:09] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '4' on SIP/carrier-c4022740 [Feb 26 12:15:10] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '5' received on SIP/carrier-c4022740 [Feb 26 12:15:10] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '5' on SIP/carrier-c4022740 [Feb 26 12:15:10]
Re: [asterisk-users] Problems with Outbound Calls
On Fri, Feb 27, 2009 at 07:12:41PM +0900, Wye-khe Kwok wrote: We managed to find a fix through the following (For anyone who's interested): Running /sbin/ztcfg -vv to configure Zaptel initially resulted in an error of: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' We then Chmodded everything under /dev/zap/ , rebooted and almost fell off our chairs when it worked! That's odd. The files under /dev/ are actually on a ramdisk. That is: they are wiped on reboot. I can't see how your chmod had any effect. What's the output of: df /dev/zap/ctl We were initially on the impression that Zaptel is only used with Analogue - can anyone verify this? Also with E1, T1, (J1?), BRI and TDM over Ethernet. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] call-limit on a per destination basis
Hello
OK I have tried this in my dialplan:
exten = _0262XX,1,Set(GROUP()=Reunion)
exten = _0262XX,2,GotoIf(${GROUP_COUNT(Reunion)} 24 ? 500)
exten = _0262XX,n,NoOp(This channel is member of group: ${GROUP()})
exten = _0262XX,n,NoOp(Number of channels is ${GROUP_COUNT(Reunion)})
exten = _0262XX,n,Set(SPYGROUP=1003)
exten = _0262XX,n,Dial(IAX2/dedibox-etang-sale/${EXTEN})
exten = _0262XX,n,Congestion()
exten = _0262XX,500,NoOp(Total channels congested, retuning NOCAV)
exten = _0262XX,501,Congestion()
However here's what i see on the CLI:
-- IAX2/dedibox-etang-sale-34 is making progress passing it to
SIP/5060-006edf50
-- IAX2/dedibox-etang-sale-6 is making progress passing it to
SIP/5060-007654f0
-- Executing [0262211...@route:1] Set(SIP/5060-0070b9d0,
GROUP()=Reunion) in new stack
-- Executing [0262211...@route:2] GotoIf(SIP/5060-0070b9d0, 21 24 ?
500) in new stack
-- Goto (route,0262211459,500)
-- Executing [0262211...@route:500] NoOp(SIP/5060-0070b9d0, Total
channels congested| retuning NOCAV) in new stack
-- Executing [0262211...@route:501] Congestion(SIP/5060-0070b9d0, )
in new stack
I am *totally puzzled* with this:
GotoIf(SIP/5060-0070b9d0, 21 24 ? 500) in new stack
-- Goto (route,0262211459,500)
What GotoIf 21 24 returns true
Any ideas?
Cheers
Jean-Michel.
2009/2/26, Klaus Darilion klaus.mailingli...@pernau.at:
I have no clue about IAX, but if IAX does not support it you can program
it yourself using the GROUP and GROUPCOUNT functions.
regards
klaus
Jean-Michel Hiver wrote:
Hello,
I use asterisk to to IAX2 trunking between London POP Reunion Island
pop. I would like to know if it's possible to do a kind of call-limit
(i.e. restrict to XX) channels but on a per dialcode and / or
destination basis.
For example:
[trunk]
; reunion proper, i want to send no more than 24 channels
exten = _0262XX,1,Dial(IAX2/mytrunk/${EXTEN})
; reunion mobile, i want to send no more than 12 channels
exten = _0692XX,1,Dial(IAX2/mytrunk/${EXTEN})
exten = _0693XX,1,Dial(IAX2/mytrunk/${EXTEN})
How would you go about it? Currently my IAX2 peer definition looks like
this:
# machine in london
[mytrunk]
type=friend
host=$reunion_ip
trunk=yes
qualify=yes
context=route
# machine in reunion island
[mytrunk]
type=friend
host=$london_ip
trunk=yes
qualify=yes
context=route
I use version Asterisk 1.4.11, production environment currently doing
25,000 minutes / day (that means if i want to upgrade i need to do it on
separate servers just in case something goes wrong).
Cheers,
Jean-Michel.
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--
Jean-Michel Hiver - Synapse co-founder CTO
GSM +262 692 828 070
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[asterisk-users] change language and playback issue
Hi, I have problem with Asterisk 1.6.0.1. I need to change language for playing prompts in Lithuanian. But in Asterisk 1.6.0.1 version everytime plays in English, but in Asterisk 1.4.x I haven't any problem. Maybe it is a bug ...? So I paste my test dialpan and prompt's locations. I hope this helps you. Files are: [r...@voip asterisk]# find /var/lib/asterisk/sounds/test -name '*.wav' /var/lib/asterisk/sounds/test/lt/enter-conf-pin-number_8.wav /var/lib/asterisk/sounds/test/enter-conf-pin-number_8.wav Dialplan: [test-prompt] exten = s,1,Answer exten = s,n,Wait(1) exten = s,n(begin),Set(TIMEOUT(digit)=3) exten = s,n,Noop(New Call) exten = s,n,Set(TIMEOUT(response)=60) exten = s,n,Background(choose_language) exten = s,n,WaitExten(8) exten = 1,1,Set(CHANNEL(language)=lt) exten = 1,n,Goto(123,1) exten = 2,1,Set(CHANNEL(language)=en) exten = 2,n,Goto(123,1) exten = 123,1,Playback(test/enter-conf-pin-number_8) The output: -- Executing [37052058...@from-trunk:1] Goto(SIP/111-b4091d40, test-prompt,s,1) in new stack -- Goto (test-prompt,s,1) -- Executing [...@test-prompt:1] Answer(SIP/111-b4091d40, ) in new stack -- Executing [...@test-prompt:2] Wait(SIP/111-b4091d40, 1) in new stack -- Executing [...@test-prompt:3] Set(SIP/111-b4091d40, TIMEOUT(digit)=3) in new stack -- Digit timeout set to 3 -- Executing [...@test-prompt:4] NoOp(SIP/111-b4091d40, New Call) in new stack -- Executing [...@test-prompt:5] Set(SIP/111-b4091d40, TIMEOUT(response)=60) in new stack -- Response timeout set to 60 -- Executing [...@test-prompt:6] BackGround(SIP/111-b4091d40, choose_language) in new stack -- SIP/111-b4091d40 Playing 'choose_language.slin' (language 'en') == CDR updated on SIP/111-b4091d40 -- Executing [...@test-prompt:1] Set(SIP/111-b4091d40, CHANNEL(language)=lt) in new stack -- Executing [...@test-prompt:2] Goto(SIP/111-b4091d40, 123,1) in new stack -- Goto (test-prompt,123,1) -- Executing [...@test-prompt:1] Playback(SIP/111-b4091d40, test/enter-conf-pin-number_8) in new stack -- SIP/111-b4091d40 Playing 'test/enter-conf-pin-number_8.slin' (language 'lt') -- Auto fallthrough, channel 'SIP/111-b4091d40' status is 'UNKNOWN' Thanks -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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] [FIXED] Re: call-limit on a per destination basis
The correct syntax for GotoIf is:
exten = _0262XX,n,GotoIf($[${GROUP_COUNT(Reunion)}24]?500)
Otherwise it seems to evaluate the string number 24 which is always
true.
Duh...
Thx
JM
--
Jean-Michel Hiver - Synapse co-founder CTO
GSM +262 692 828 070
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[asterisk-users] [HOWTO] Priorize one destination over another on a link
Hello List,
The list sorted my problem thus I shall contribute back ;-)
PROBLEM:
I am posting this example, where I have a Reunion link of 30 channels. If
i send all the traffic (proper + mobile) on the link, the less profitable
proper traffic fills the link and leaves no channel for more profitable
mobile traffic. Some kind of priority is needed to always leave space for
mobile trafic: you don't want to be terminating traffic that yields 0.001 /
min of profit when you could be terminating traffic yielding 10x as much
instead.
SOLUTION
Use asterisk grouping and conditional fonctions to dynamically limit the
proper traffic in order to always keep a few channels free.
For example, imagine you have 0 channels of mobile : allow proper to use up
to 26 channels
For example, imagine you have 5 channels of mobile : allow proper to use up
to 21 channels
For example, imagine you have 10 channels of mobile : allow proper to use up
to 16 channels
For example, imagine you have 28 channels of mobile : allow proper to use up
to 0 channels
In order to do this, you set the SAME group for both mobile and proper
channels and then you apply a conditional only on the traffic which you want
to limit.
You could also be using the same kind of technique if you had two different
classes of customers: retail and wholesale. You want wholesalers to fill
your pipes of course (with best effort), but you do not want this traffic to
affect your retail service.
IMPLEMENTATION
==
This is the implementation on my production server, seems to work well, feel
free to modify to suit up your needs.
; Reunion Proper : use a conditional statement to dynamically limit number
of channels
exten = _0262XX,1,Set(GROUP()=Reunion)
exten = _0262XX,n,NoOp(This channel is member of group: ${GROUP()})
exten = _0262XX,n,NoOp(Number of channels is ${GROUP_COUNT(Reunion)})
exten = _0262XX,n,GotoIf($[${GROUP_COUNT(Reunion)}26]?500)
exten = _0262XX,n,Dial(IAX2/dedibox-etang-sale/${EXTEN})
exten = _0262XX,500,NoOp(Total channels congested, retuning NOCAV)
exten = _0262XX,501,Congestion()
; Reunion Mobile : always gets through which increments the channel count...
and thus reduces proper capacity
exten = _0692XX,1,Set(GROUP()=Reunion)
exten = _0692XX,n,NoOp(This channel is member of group: ${GROUP()})
exten = _0692XX,n,NoOp(Number of channels is ${GROUP_COUNT(Reunion)})
exten = _0692XX,n,Dial(IAX2/dedibox-etang-sale/${EXTEN})
exten = _0693XX,1,Set(GROUP()=Reunion)
exten = _0693XX,n,NoOp(This channel is member of group: ${GROUP()})
exten = _0693XX,n,NoOp(Number of channels is ${GROUP_COUNT(Reunion)})
exten = _0693XX,n,Dial(IAX2/dedibox-etang-sale/${EXTEN})
I hope this piece of information is of use to somebody, some day!
Cheers
Jean-Michel
http://ykoz.net/
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Re: [asterisk-users] changing /etc/dahdi/system.conf
On Feb 26, 2009, at 3:05 PM, Tzafrir Cohen wrote: On Thu, Feb 26, 2009 at 01:44:19PM -0700, Brandon B. wrote: At the top of my /etc/dahdi/system.conf file is this line: # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009 -- do not hand edit OK, so how do I adjust the timing source and LBO numbers, and echo cancellers if I'm not supposed to edit this file? Hmm I guess you should n't take that too seriously :-( Maybe this text should be changed. Go, go, gadget bugtracker! http://bugs.digium.com/view.php?id=14569 JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- 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] building a phone
Hi folks A common wisdom here is that one should use a proper hardware phone rather that an extra software on the user's PC. Why is that such a big issue? One thing that bothers me with the current crop of hardware SIP phones is that they are hopelessly properitary. So what would it take to build a fully-adaptable phone? Here are some of my thoughts. This is not anything I plan to do soon (if at all), but I really find it strange that there aren't such phones already. == Small Quantities: When you look at such systems it becomes aparant that you can get much nicer prices if you buy large quanities. But this is something that will be a problem. Not only for prototying. The fact that you're limited to a strict hardware setting is very limiting. No mixing and matching like in a standard PC. I'm not exactly sure how to overcome that. == Platforms: There are many embedded platforms nowadays. I assume that the relevant application requires some non-trivial CPU power. I would exclude e.g. a 486-based systems. My target phone should be able to handle at least two concurrent Speex calls. Preferrebly 6 speex calls and above. OTOH, I can't afford a monster CoreDuo. I need a quiet system with no fan. Thus the target CPU may be higher end VIA or Atom. Not sure about Geode. There are also some interesting ARM-based boards around. I'm completely unfamiliar with them but I suspect that they may prove to be cheaper. == SIP Software: Not really sure here. There must be something close to usable already, I guess. == Micro Browser: Hell no! The device should have an LCD display, and the content of that display should be programmable. Programming it using a HTML renderred is a bad design decision. The device should be a good phone. It should not attempt to be a web browser, as it will be a lousy one. == Handset: I suppose that an obvious starting point for a handset is skype phones such as USB handsets from yealink. Far from an optimal design, but a driver already exists. == Ease of Use: A phone must be usable. The target device must be something my mom can use. However that does not mean it must be easy to program. It must be programmable and hackable. But I can live with a complicated user interface for that. If such phones become successful and useful, better interfaces will eventually be written. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Current state of Asterisk and Virtualization?
On Feb 26, 2009, at 6:01 PM, Senad Jordanovic wrote: Gavin Henry wrote: Hi all, In a pure VoIP env, what is the current state of do's and don't s of virtualizing * in order to provide multiple separate instances, say for hosting lots of Asterisk-gui/FreePBX/a-n-other gui? I've read lots of threads going back to 2007 and I'm in the general option that kvm is the way to go now, if at all. If dadhi_dummy/zt_dummy is still an issue for conferencing etc. a conference box could be put along side the vm hardware and have a card in it. Thoughts, experiences and being told to shut up are all very much appreciated. Thanks. http://www.bicomsystems.com/products/C/P/797/411/ ...and also: http://voxilla.com/2009/02/12/amazon-ec2-voip-1096 http://voxilla.com/2009/02/13/asterisk-amazon-ec2-1178 http://voxilla.com/2009/02/18/asterisk-on-the-cloud-with-a-click-1405 and: http://www.simionovich.com/?p=180 http://www.simionovich.com/?p=243 ...and lots of others on Nir's blog. Even MORE resources/questions/answers: http://www.google.com/search?hl=enq=ztxenbtnG=Google+Searchaq=foq= http://www.google.com/search?num=30hl=ensafe=offq=xen+and+ztdummybtnG=Search JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- 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] CDR - Asterisk-Stat and PHP5
Hi Paulo, It's right! I've changed the zend.zel_compatibility_mode to Off, following your suggestion, and asterisk-stat is still working on PHP5. Thank you! Just for clarity: the default values for the two keys on OpenSuse 10.2 (updated to latest revision), and following, is Off. Best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com Tiago Durante wrote: On Thu, Feb 26, 2009 at 12:52 PM, Paulo Santos paulo.r.san...@sapo.pt wrote: Marco Signorini wrote: Hi Tiago. I've it working on PHP 5.2.6 but only after having modified the php.ini default configuration keys: zend.ze1_compatibility_mode = Off short_open_tag = Off Though my zend.ze1_compatibility_mode is set to Off, short_open_tag is set to On and it is working. Those are my defaults, at least I never changed them. Installed with apt-get on Debian 4.0, PHP version 5.2.0-8+etch13. Cool, I'm gonna test it and I let you guys know if worked or not. 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
Re: [asterisk-users] building a phone
Tzafrir Cohen wrote: Hi folks A common wisdom here is that one should use a proper hardware phone rather that an extra software on the user's PC. Why is that such a big issue? One thing that bothers me with the current crop of hardware SIP phones is that they are hopelessly properitary. no more so than a vcr or other consumer device. So what would it take to build a fully-adaptable phone? why bother when there are lots of $100 perfectly fine phones already ? Here are some of my thoughts. This is not anything I plan to do soon (if at all), but I really find it strange that there aren't such phones already. why not just take an existing device such as the n770 or nintendo ds. load a sip client of your choice on, and you are done - wireless phone, touchscreen, runs any other software you want, already low cost, no manufacturing. (both devices already have wifi, speaker and microphone as well as a colour touchscreen, so what else is missing ?) there are also iphone clones coming down into the same price range that are fully programmable. == Small Quantities: When you look at such systems it becomes aparant that you can get much nicer prices if you buy large quanities. But this is something that will be a problem. Not only for prototying. The fact that you're limited to a strict hardware setting is very limiting. No mixing and matching like in a standard PC. I'm not exactly sure how to overcome that. == Platforms: There are many embedded platforms nowadays. I assume that the relevant application requires some non-trivial CPU power. I would exclude e.g. a 486-based systems. My target phone should be able to handle at least two concurrent Speex calls. Preferrebly 6 speex calls and above. OTOH, I can't afford a monster CoreDuo. I need a quiet system with no fan. Thus the target CPU may be higher end VIA or Atom. Not sure about Geode. There are also some interesting ARM-based boards around. I'm completely unfamiliar with them but I suspect that they may prove to be cheaper. == SIP Software: Not really sure here. There must be something close to usable already, I guess. == Micro Browser: Hell no! The device should have an LCD display, and the content of that display should be programmable. Programming it using a HTML renderred is a bad design decision. The device should be a good phone. It should not attempt to be a web browser, as it will be a lousy one. == Handset: I suppose that an obvious starting point for a handset is skype phones such as USB handsets from yealink. Far from an optimal design, but a driver already exists. == Ease of Use: A phone must be usable. The target device must be something my mom can use. However that does not mean it must be easy to program. It must be programmable and hackable. But I can live with a complicated user interface for that. If such phones become successful and useful, better interfaces will eventually be written. ___ -- 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] building a phone
Tzafrir Cohen wrote: Hi folks A common wisdom here is that one should use a proper hardware phone rather that an extra software on the user's PC. Why is that such a big issue? Marketability for one. People worldwide understand the telephone paradigm. You have a handset and a box with numbers. You pick it up and dial, talk through the handset, and listen in the other end. It's simple. It's an elegant design. And everyone from 1 year olds to my 97 year old grandfather can use it. Software phones? Not so much. In fact, not even close. The additional complexity of running software on a machine ALONE would keep my grandfather and that 1 year old from using it. Headsets? Seriously? Since when have those been user-friendly OR comfortably. In essence, adherence to a software phone paradigm breaks a century of design advancement in telephone ergonomics, psychology, and reliance, and replaces it with something that's clearly just a kludgy add-on to a product which was never originally designed for the task. One thing that bothers me with the current crop of hardware SIP phones is that they are hopelessly properitary. So what would it take to build a fully-adaptable phone? Here are some of my thoughts. This is not anything I plan to do soon (if at all), but I really find it strange that there aren't such phones already. == Small Quantities: When you look at such systems it becomes aparant that you can get much nicer prices if you buy large quanities. But this is something that will be a problem. Not only for prototying. The fact that you're limited to a strict hardware setting is very limiting. No mixing and matching like in a standard PC. I'm not exactly sure how to overcome that. This is one of the biggest reasons all the hardware phones are proprietary -- they're each written for different basic hardware. == Platforms: There are many embedded platforms nowadays. I assume that the relevant application requires some non-trivial CPU power. I would exclude e.g. a 486-based systems. My target phone should be able to handle at least two concurrent Speex calls. Preferrebly 6 speex calls and above. OTOH, I can't afford a monster CoreDuo. I need a quiet system with no fan. Thus the target CPU may be higher end VIA or Atom. Not sure about Geode. There are also some interesting ARM-based boards around. I'm completely unfamiliar with them but I suspect that they may prove to be cheaper. == SIP Software: Not really sure here. There must be something close to usable already, I guess. == Micro Browser: Hell no! The device should have an LCD display, and the content of that display should be programmable. Programming it using a HTML renderred is a bad design decision. The device should be a good phone. It should not attempt to be a web browser, as it will be a lousy one. == Handset: I suppose that an obvious starting point for a handset is skype phones such as USB handsets from yealink. Far from an optimal design, but a driver already exists. == Ease of Use: A phone must be usable. The target device must be something my mom can use. However that does not mean it must be easy to program. It must be programmable and hackable. But I can live with a complicated user interface for that. If such phones become successful and useful, better interfaces will eventually be written. Just a note here -- a complicated user interface, though you personally may be able to live with it, will pretty much ensure that the phones never become successful enough for a better one to be written. UI design is about 10% code and 90% psychology (and so FEW people who call themselves UI 'programmers' understand that). Just having a UI that can get you from point A to point B without typing in commands is NOT a UI worth making, as it will never be a UI worth using. ___ -- 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] codec_dahdi and Asterisk 1.6.0.6
1) How can I use codec_dahdi? Would it be useful when passing a call from one dahdi channel to another dahdi channel? It is used whenever you need G729 or G723 transcoding (or any other format supported by the Digium transcoding board). If you don't have a Digium transcoding board then you don't need that module and you can disable it in modules.conf ERROR[18854]: codec_dahdi.c:399 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory That simply means you don't have loaded the dahdi_transcode module. If you wanted to enable it you can with modprobe dahdi_transcode, that will create /dev/dahdi/transcode device, however if you don't have a Digium transcoding board the device is useless. Is this because I do not have a hardware trancoding device? Can I safely ignore this error or is it a bug? Yes, you can ignore it, or better yet, disable dahdi_codec module in modules.conf Moisés Silva -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] [FIXED] Re: call-limit on a per destination basis
Just a tip: throw extensions.conf away and use extensions.ael - much
more easy:
_0262XX = {
Set(GROUP()=Reunion);
if( ${GROUP_COUNT(Reunion)} 24) {
NoOp(Total channels congested, retuning NOCAV);
Congestion();
} else {
NoOp(This channel is member of group: ${GROUP()});
NoOp(Number of channels is ${GROUP_COUNT(Reunion)});
Set(SPYGROUP=1003);
Dial(IAX2/dedibox-etang-sale/${EXTEN});
Congestion();
}
Further, I would use a macro:
macro checkMaxCallsMakro(groupid,limit) {
if ( ${GROUP_COUNT(${groupid})} = ${limit} ) {
NoOp(ERROR: Limit ${hardlimit} reached for ${groupid}!);
Hangup(34); //Cause No. 34: no circuit/channel av. (SIP 503)
}
Set(GROUP()=${groupid});
}
context foobar {
_0262XX = {
checkMaxCallsMakro(Reunion,24)
Set(SPYGROUP=1003);
Dial(IAX2/dedibox-etang-sale/${EXTEN});
Congestion();
}
}
regards
klaus
Jean-Michel Hiver schrieb:
The correct syntax for GotoIf is:
exten = _0262XX,n,GotoIf($[${GROUP_COUNT(Reunion)}24]?500)
Otherwise it seems to evaluate the string number 24 which is always
true.
Duh...
Thx
JM
--
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GSM +262 692 828 070
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Re: [asterisk-users] building a phone
It's a dream! It's since years that I'm thinking to have an open hardware project targeted to a SIP application. I'm thinking, for example, to have a modular system that can be targeted to different custom appliances like, for example, (video) door bell opener/intercom, or building/desktop music streamer, or SIP compliant actuators. I have a (very) little experience on electronic projects. Is there something I can do to help starting a similar project? Thank you and best regards. Marco Signorini Tzafrir Cohen wrote: Hi folks A common wisdom here is that one should use a proper hardware phone rather that an extra software on the user's PC. Why is that such a big issue? One thing that bothers me with the current crop of hardware SIP phones is that they are hopelessly properitary. So what would it take to build a fully-adaptable phone? Here are some of my thoughts. This is not anything I plan to do soon (if at all), but I really find it strange that there aren't such phones already. == Small Quantities: When you look at such systems it becomes aparant that you can get much nicer prices if you buy large quanities. But this is something that will be a problem. Not only for prototying. The fact that you're limited to a strict hardware setting is very limiting. No mixing and matching like in a standard PC. I'm not exactly sure how to overcome that. == Platforms: There are many embedded platforms nowadays. I assume that the relevant application requires some non-trivial CPU power. I would exclude e.g. a 486-based systems. My target phone should be able to handle at least two concurrent Speex calls. Preferrebly 6 speex calls and above. OTOH, I can't afford a monster CoreDuo. I need a quiet system with no fan. Thus the target CPU may be higher end VIA or Atom. Not sure about Geode. There are also some interesting ARM-based boards around. I'm completely unfamiliar with them but I suspect that they may prove to be cheaper. == SIP Software: Not really sure here. There must be something close to usable already, I guess. == Micro Browser: Hell no! The device should have an LCD display, and the content of that display should be programmable. Programming it using a HTML renderred is a bad design decision. The device should be a good phone. It should not attempt to be a web browser, as it will be a lousy one. == Handset: I suppose that an obvious starting point for a handset is skype phones such as USB handsets from yealink. Far from an optimal design, but a driver already exists. == Ease of Use: A phone must be usable. The target device must be something my mom can use. However that does not mean it must be easy to program. It must be programmable and hackable. But I can live with a complicated user interface for that. If such phones become successful and useful, better interfaces will eventually be written. ___ -- 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] building a phone
Marco Signorini wrote: It's a dream! It's since years that I'm thinking to have an open hardware project targeted to a SIP application. there is already a project called openmoko - join it and buy some hardware. The phone is large and clunky - the idea is good, but not something you're ever going to carry in your pocket, and somewhat silly when there is already smaller hardware out there that runs linux at less cost than their device. I'm thinking, for example, to have a modular system that can be targeted to different custom appliances like, for example, (video) door bell opener/intercom, or building/desktop music streamer, or SIP compliant actuators. I have a (very) little experience on electronic projects. Is there something I can do to help starting a similar project? Thank you and best regards. Marco Signorini Tzafrir Cohen wrote: Hi folks A common wisdom here is that one should use a proper hardware phone rather that an extra software on the user's PC. Why is that such a big issue? One thing that bothers me with the current crop of hardware SIP phones is that they are hopelessly properitary. So what would it take to build a fully-adaptable phone? Here are some of my thoughts. This is not anything I plan to do soon (if at all), but I really find it strange that there aren't such phones already. == Small Quantities: When you look at such systems it becomes aparant that you can get much nicer prices if you buy large quanities. But this is something that will be a problem. Not only for prototying. The fact that you're limited to a strict hardware setting is very limiting. No mixing and matching like in a standard PC. I'm not exactly sure how to overcome that. == Platforms: There are many embedded platforms nowadays. I assume that the relevant application requires some non-trivial CPU power. I would exclude e.g. a 486-based systems. My target phone should be able to handle at least two concurrent Speex calls. Preferrebly 6 speex calls and above. OTOH, I can't afford a monster CoreDuo. I need a quiet system with no fan. Thus the target CPU may be higher end VIA or Atom. Not sure about Geode. There are also some interesting ARM-based boards around. I'm completely unfamiliar with them but I suspect that they may prove to be cheaper. == SIP Software: Not really sure here. There must be something close to usable already, I guess. == Micro Browser: Hell no! The device should have an LCD display, and the content of that display should be programmable. Programming it using a HTML renderred is a bad design decision. The device should be a good phone. It should not attempt to be a web browser, as it will be a lousy one. == Handset: I suppose that an obvious starting point for a handset is skype phones such as USB handsets from yealink. Far from an optimal design, but a driver already exists. == Ease of Use: A phone must be usable. The target device must be something my mom can use. However that does not mean it must be easy to program. It must be programmable and hackable. But I can live with a complicated user interface for that. If such phones become successful and useful, better interfaces will eventually be written. ___ -- 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] building a phone
Jon Pounder wrote: Marco Signorini wrote: It's a dream! It's since years that I'm thinking to have an open hardware project targeted to a SIP application. there is already a project called openmoko - join it and buy some hardware. The phone is large and clunky - the idea is good, but not something you're ever going to carry in your pocket, and somewhat silly when there is already smaller hardware out there that runs linux at less cost than their device. Thank you Jon, Really interesting project! I'll follow it. Best regards, Marco Signorini ___ -- 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] ATA recommendation (wih FTP provisioning)
I don't think they are locking the same device that you buy when you buy the -NA version. I believe that Linksys is making pre-configured devices for these large buyers and selling them much cheaper to them in bulk than they sell the -NA version to the community at large. I'm sure you are correct. The basic firmware is the same, but they come pre-loaded with configuration that automatically goes to the provider's web site. Again, that's not bad as long as there is a factory reset that can restore it, which there is, but it is password protected and the provider won't give out the password. If the device were owned by the provider, that would be acceptable, but the provider sells them to the customer so the customer deserves to be able to use them for other purposes if they need to. Wilton ___ -- 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] building a phone
Thanks for your reply, On Fri, Feb 27, 2009 at 10:53:21AM -0500, SIP wrote: Tzafrir Cohen wrote: Hi folks A common wisdom here is that one should use a proper hardware phone rather that an extra software on the user's PC. Why is that such a big issue? Marketability for one. People worldwide understand the telephone paradigm. You have a handset and a box with numbers. You pick it up and dial, talk through the handset, and listen in the other end. It's simple. It's an elegant design. And everyone from 1 year olds to my 97 year old grandfather can use it. Software phones? Not so much. In fact, not even close. The additional complexity of running software on a machine ALONE would keep my grandfather and that 1 year old from using it. Headsets? Seriously? Since when have those been user-friendly OR comfortably. In essence, adherence to a software phone paradigm breaks a century of design advancement in telephone ergonomics, psychology, and reliance, and replaces it with something that's clearly just a kludgy add-on to a product which was never originally designed for the task. But imposes many stupid design limitations as well. A limitation of CPU power. A limitation of screen space. A limitation of a pointing device. A user has a keyboard to enter URLs. You can do that with a dialpad. Sort of. And you curse whoever invented that. How do you dial to an address pointed from the page you were browsing? == Small Quantities: When you look at such systems it becomes aparant that you can get much nicer prices if you buy large quanities. But this is something that will be a problem. Not only for prototying. The fact that you're limited to a strict hardware setting is very limiting. No mixing and matching like in a standard PC. I'm not exactly sure how to overcome that. This is one of the biggest reasons all the hardware phones are proprietary -- they're each written for different basic hardware. That's no inherent reason for being proprietary. It's proprietary because that's how they can make money of it. I think that from selling a hardware for which there's a good programmable phone you can eventually make more money. But then that's pure speculation. == Ease of Use: A phone must be usable. The target device must be something my mom can use. However that does not mean it must be easy to program. It must be programmable and hackable. But I can live with a complicated user interface for that. If such phones become successful and useful, better interfaces will eventually be written. Just a note here -- a complicated user interface, though you personally may be able to live with it, will pretty much ensure that the phones never become successful enough for a better one to be written. UI design is about 10% code and 90% psychology (and so FEW people who call themselves UI 'programmers' understand that). Just having a UI that can get you from point A to point B without typing in commands is NOT a UI worth making, as it will never be a UI worth using. It's a UI worth making because I don't spend a year over it. And because I'm not a UI designer and hte phone is first and foremost for me :-) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] building a phone
I assume that the relevant application requires some non-trivial CPU power. I would exclude e.g. a 486-based systems. I'm not sure that's the case. The industry has gone in the direction of throwing lots of silicon at a problem, often as an excuse for poorly written code, sometimes in an interpreted language. There are a number of high integration CPUs out there that I suspect could do this sort of thing. I develop device controllers for a variety of industry needs. They tend to have Ethernet, RS-232, sometimes 1 Mb/s synchronous communication. G711, quarter VGA color LCD with touchscreen and control loops running at about a 1 ms rate. The entire code takes less than 256K in C. My choice of processor is the DStni Ex (made by Lantronix and sold by Grid Connect) which is a high integration, high speed 186 core with two 10/100 Ethernet Ports and 256K of RAM on it in addition to the usual assortment of other stuff. The above required platform adds three support chips (one being the LCD controller). The CPU can run over 100 MHz. Memory accesses take one clock and typical instructions take two or three. Cost is in the $10 to $20 range for the chip and power consumption is around 1 W (the LCD backlight takes more than that!) I'm sure there are several other comparable platforms out there, such as by Digi International. The Geode is a good candidate as are some VIA chips, if one wants to use protected mode x86. The biggest thing for this is don't even consider Intel. For most of their life they have not provided cutting edge solutions for embedded use. Most of their stuff consumes too much power. And most importantly, they are targeting the very volatile, short lived PC market. By the time you get an embedded design up and running and reach market penetration, you won't be able to buy the chip any more. Wilton ___ -- 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] building a phone
Wilton Helm wrote: I assume that the relevant application requires some non-trivial CPU power. I would exclude e.g. a 486-based systems. I'm not sure that's the case. The industry has gone in the direction of throwing lots of silicon at a problem, often as an excuse for poorly written code, sometimes in an interpreted language. just look at windows for a good/bad example of that There are a number of high integration CPUs out there that I suspect could do this sort of thing. I develop device controllers for a variety of industry needs. They tend to have Ethernet, RS-232, sometimes 1 Mb/s synchronous communication. G711, quarter VGA color LCD with touchscreen and control loops running at about a 1 ms rate. The entire code takes less than 256K in C. My choice of processor is the DStni Ex (made by Lantronix and sold by Grid Connect) which is a high integration, high speed 186 core with two 10/100 Ethernet Ports and 256K of RAM on it in addition to the usual assortment of other stuff. The above required platform adds three support chips (one being the LCD controller). The CPU can run over 100 MHz. Memory accesses take one clock and typical instructions take two or three. Cost is in the $10 to $20 range for the chip and power consumption is around 1 W (the LCD backlight takes more than that!) These days if you are writing stuff in C, the hardware platform is really not of ultimate importance since you can cross compile with gcc from just about anything to anything. What matters more is there are drivers available for all the other bits besides the cpu itself. I was just saying the other day, how I used to write pretty involved programs in 256 bytes in the original basic stamps. Most people today could not even conceive how a program could do anything useful in 256bytes let alone 256k or 256mb. I'm sure there are several other comparable platforms out there, such as by Digi International. The Geode is a good candidate as are some VIA chips, if one wants to use protected mode x86. The biggest thing for this is don't even consider Intel. For most of their life they have not provided cutting edge solutions for embedded use. Most of their stuff consumes too much power. And most importantly, they are targeting the very volatile, short lived PC market. By the time you get an embedded design up and running and reach market penetration, you won't be able to buy the chip any more. Wilton ___ -- 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] building a phone
On Fri, 27 Feb 2009, Tzafrir Cohen wrote: Hi folks A common wisdom here is that one should use a proper hardware phone rather that an extra software on the user's PC. Why is that such a big issue? Both my laptop and desktop PC have poor quality microphone inputs. I use a USB phone on my laptop when out and about. One thing that bothers me with the current crop of hardware SIP phones is that they are hopelessly properitary. SIP is an open standard... So what would it take to build a fully-adaptable phone? Persuade an existing manufacturer to provide an SDK for their phone... And thinking about it, don't Snoms run Linux? anyone asked if an SDK is avalable? Here are some of my thoughts. This is not anything I plan to do soon (if at all), but I really find it strange that there aren't such phones already. == Small Quantities: When you look at such systems it becomes aparant that you can get much nicer prices if you buy large quanities. But this is something that will be a problem. Not only for prototying. The fact that you're limited to a strict hardware setting is very limiting. No mixing and matching like in a standard PC. I'm not exactly sure how to overcome that. == Platforms: There are many embedded platforms nowadays. I assume that the relevant application requires some non-trivial CPU power. I would exclude e.g. a 486-based systems. My target phone should be able to handle at least two concurrent Speex calls. Preferrebly 6 speex calls and above. OTOH, I can't afford a monster CoreDuo. I need a quiet system with no fan. Thus the target CPU may be higher end VIA or Atom. Not sure about Geode. There are also some interesting ARM-based boards around. I'm completely unfamiliar with them but I suspect that they may prove to be cheaper. Custom DSP or ARM. Don't forget power requirements too. PoE may well be able to supply 15W, but imagine a building with 100 x 15W phones... Everything I've plugged into my meter idles at about 2W (Grandstream, Snom, Siemens, ATL) == SIP Software: Not really sure here. There must be something close to usable already, I guess. == Micro Browser: Hell no! Executives want it... The device should have an LCD display, and the content of that display should be programmable. Programming it using a HTML renderred is a bad design decision. The device should be a good phone. It should not attempt to be a web browser, as it will be a lousy one. I've actually used the RSS reader in my Grandstream Video phones.. It's usable, but I take your point about the full web thing! == Handset: I suppose that an obvious starting point for a handset is skype phones such as USB handsets from yealink. Far from an optimal design, but a driver already exists. Seen these: http://www.fit-pc.co.uk/meet-fit-pc.html#tiny And of-course the power line thing mentioned here a few days earlier - eg. http://www.theinquirer.net/inquirer/news/143/1051143/pc-plug-coming-soon-wall-near if you're going to use a USB phone, these platforms are ideal, if a little pricey.. I did play with a yealink phone and Linux some time back - get the display and keyboard working and it'll be very functional... (I got it to almost work with Zoiper, so a custom app. would be easy) Alas, I gave it to a friend, then go anothe from the same source (tesco), thinking it would be identical - and it was - in packaging, but was a totally different phone )-: == Ease of Use: A phone must be usable. The target device must be something my mom can use. However that does not mean it must be easy to program. It must be programmable and hackable. But I can live with a complicated user interface for that. If such phones become successful and useful, better interfaces will eventually be written. Dial the number, push the green button, off you go... Gordon ___ -- 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] building a phone
2009/2/27 Wilton Helm wh...@compuserve.com I assume that the relevant application requires some non-trivial CPU power. I would exclude e.g. a 486-based systems. I'm not sure that's the case. The industry has gone in the direction of throwing lots of silicon at a problem, often as an excuse for poorly written code, sometimes in an interpreted language. There are a number of high integration CPUs out there that I suspect could do this sort of thing. I develop device controllers for a variety of industry needs. They tend to have Ethernet, RS-232, sometimes 1 Mb/s synchronous communication. G711, quarter VGA color LCD with touchscreen and control loops running at about a 1 ms rate. The entire code takes less than 256K in C. My choice of processor is the DStni Ex (made by Lantronix and sold by Grid Connect) which is a high integration, high speed 186 core with two 10/100 Ethernet Ports and 256K of RAM on it in addition to the usual assortment of other stuff. The above required platform adds three support chips (one being the LCD controller). The CPU can run over 100 MHz. Memory accesses take one clock and typical instructions take two or three. Cost is in the $10 to $20 range for the chip and power consumption is around 1 W (the LCD backlight takes more than that!) I'm sure there are several other comparable platforms out there, such as by Digi International. The Geode is a good candidate as are some VIA chips, if one wants to use protected mode x86. The biggest thing for this is don't even consider Intel. For most of their life they have not provided cutting edge solutions for embedded use. Most of their stuff consumes too much power. And most importantly, they are targeting the very volatile, short lived PC market. By the time you get an embedded design up and running and reach market penetration, you won't be able to buy the chip any more. Wilton ___ -- 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 I wonder what kind of hardware snom use, they got linux, they got openvpn. I would be nice to have that, and yes i want a gui, maybe not embedded to reduce load, but something like an external config generator software would be nice. ___ -- 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] building a phone
On Fri, Feb 27, 2009 at 09:40:09AM -0700, Wilton Helm wrote: I assume that the relevant application requires some non-trivial CPU power. I would exclude e.g. a 486-based systems. I'm not sure that's the case. The industry has gone in the direction of throwing lots of silicon at a problem, often as an excuse for poorly written code, sometimes in an interpreted language. Actually I have stated a bit above that that I would like the system to support 2 SIP lines (and preferably 6 and more) and allow using Speex in them. Supposrt of 2 lines is a must for features such as forwarding. There are a number of high integration CPUs out there that I suspect could do this sort of thing. I develop device controllers for a variety of industry needs. They tend to have Ethernet, RS-232, sometimes 1 Mb/s synchronous communication. G711, G.711 indeed requires much less CPU power. It is good enough for a LAN-only setup. Though with a bit of forward-thinking, G.722 support would also be nice. quarter VGA color LCD with touchscreen and control loops running at about a 1 ms rate. The entire code takes less than 256K in C. My choice of processor is the DStni Ex (made by Lantronix and sold by Grid Connect) which is a high integration, high speed 186 core with two 10/100 Ethernet Ports and 256K of RAM on it in addition to the usual assortment of other stuff. The above required platform adds three support chips (one being the LCD controller). The CPU can run over 100 MHz. Memory accesses take one clock and typical instructions take two or three. Cost is in the $10 to $20 range for the chip and power consumption is around 1 W (the LCD backlight takes more than that!) Again, the main reason for me to require a higher end CPU is audio compression. But I also want the system to be run by a standard OS. It needs to be easy to add your own application there. Here's a plug that would cost you 99$: http://www.globalscaletechnologies.com/t-sheevaplugdetails.aspx#extern It has a USB and Ethernet output. The core of the system is: http://www.marvell.com/products/embedded_processors/kirkwood/index.jsp (yes, also two TDM ports with slics. No idea how to use them) That plug supports running quite a number of Linux distributions (Debian, Fedora, Gentoo, and some others) What would it take me to port Asterisk or Yate to it? An existing web interface? Hackability and ease of development is a must. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Problems with Outbound Calls
Two things The dev/zap problem was probably fixed by a modprobe that occurred on the reload and therefore had no relevance to the chmod. Ztdummy is created by zaptel and used in some non-analog functions AFAIK -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, February 27, 2009 6:06 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problems with Outbound Calls On Fri, Feb 27, 2009 at 07:12:41PM +0900, Wye-khe Kwok wrote: We managed to find a fix through the following (For anyone who's interested): Running /sbin/ztcfg -vv to configure Zaptel initially resulted in an error of: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' We then Chmodded everything under /dev/zap/ , rebooted and almost fell off our chairs when it worked! That's odd. The files under /dev/ are actually on a ramdisk. That is: they are wiped on reboot. I can't see how your chmod had any effect. What's the output of: df /dev/zap/ctl We were initially on the impression that Zaptel is only used with Analogue - can anyone verify this? Also with E1, T1, (J1?), BRI and TDM over Ethernet. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] call file concurrency
Some variant of the ulimit command would accomplish this but YMMV and Caveat Emptor. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill Michaelson Sent: Thursday, February 26, 2009 7:59 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] call file concurrency Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? ___ -- 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] call file concurrency
Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? On Fri, 27 Feb 2009, Danny Nicholas top posted: Some variant of the ulimit command would accomplish this but YMMV and Caveat Emptor. Which one? -fs::sedwards:~$ ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited file size (blocks, -f) unlimited pending signals (-i) 1024 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 1024 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 16114 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited Limiting the number of open files or file locks would not have the intended effect. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] API command monitor to record only the input channel
I am doing: Action: Monitor File: /tmp/testing Format: gsm then Mix:0 only records the output Mix: 1 combines in the input and output I wish (in this case) to only record the input - how do I do this? Jerry ___ -- 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] building a phone
On Fri, 27 Feb 2009, Tzafrir Cohen wrote: The core of the system is: http://www.marvell.com/products/embedded_processors/kirkwood/index.jsp (yes, also two TDM ports with slics. No idea how to use them) That plug supports running quite a number of Linux distributions (Debian, Fedora, Gentoo, and some others) What would it take me to port Asterisk or Yate to it? An existing web interface? You don't want a PBX in it - you want a command-line VoIP client. I was asking (here and elsewhere) if such a beast existed some time back. All I got back was put asterisk on it and use the OSS/ALSA stuff. Talk about using a sledge hammer to crack a nut.. I had a quick look at libiax and ran out of time... Hackability and ease of development is a must. Quite. Now, way back I have (still have) a Nokia 770 tablet - that runs Gizmo - but, again it's a GUI application. Get a basic OS on the box, get a command-line phone that uses USB audio, stick a mini web server for configuration and off you go... Gordon ___ -- 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] API command monitor to record only the input channel
You specify a channel. Just specify the channel on the other leg of the call. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ From: Jerry Geis ge...@pagestation.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 27 Feb 2009 12:40:37 -0500 To: asterisk-users@lists.digium.com Subject: [asterisk-users] API command monitor to record only the input channel I am doing: Action: Monitor File: /tmp/testing Format: gsm then Mix:0 only records the output Mix: 1 combines in the input and output I wish (in this case) to only record the input - how do I do this? Jerry ___ -- 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] building a phone
On Fri, Feb 27, 2009 at 05:48:28PM +, Gordon Henderson wrote: On Fri, 27 Feb 2009, Tzafrir Cohen wrote: The core of the system is: http://www.marvell.com/products/embedded_processors/kirkwood/index.jsp (yes, also two TDM ports with slics. No idea how to use them) That plug supports running quite a number of Linux distributions (Debian, Fedora, Gentoo, and some others) What would it take me to port Asterisk or Yate to it? An existing web interface? You don't want a PBX in it - you want a command-line VoIP client. I was asking (here and elsewhere) if such a beast existed some time back. All I got back was put asterisk on it and use the OSS/ALSA stuff. Talk about using a sledge hammer to crack a nut.. I had a quick look at libiax and ran out of time... Yate is the only h32, IAX and SIP free software soft phone I know. I figure adding one extra interface wouldn't be that tough :-) Linphone has a command-line interface. Not sure about others. But then again, maybe the system will run X11. That can simplify some hings. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Switch Options for a service provider
Hi, I have a growing voip business Im i looking a solution that can handle at least 3000-4000 concurrent calls with great performance. Also with a billing platform, reports, reseller platform, LCR, call routing,real time reports, SQL dababase access real time Load Reports. Any recommendation? Thanks Ignacio ___ -- 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] building a phone
Gordon Henderson wrote: On Fri, 27 Feb 2009, Tzafrir Cohen wrote: The core of the system is: http://www.marvell.com/products/embedded_processors/kirkwood/index.jsp (yes, also two TDM ports with slics. No idea how to use them) That plug supports running quite a number of Linux distributions (Debian, Fedora, Gentoo, and some others) What would it take me to port Asterisk or Yate to it? An existing web interface? You don't want a PBX in it - you want a command-line VoIP client. I was asking (here and elsewhere) if such a beast existed some time back. All I got back was put asterisk on it and use the OSS/ALSA stuff. Talk about using a sledge hammer to crack a nut.. I had a quick look at libiax and ran out of time... this mentality mystifies me too - why would you want a pbx running on a handset ? ok so you can, but once that novelty wears off in 5secs what is the practical use ? The only reason would be if you don't have any other phones, and if you don't why do you need a pbx to start with ? Hackability and ease of development is a must. Quite. Now, way back I have (still have) a Nokia 770 tablet - that runs Gizmo - but, again it's a GUI application. Get a basic OS on the box, get a command-line phone that uses USB audio, stick a mini web server for configuration and off you go... on this subject - any tips on getting gizmo to actually work ? I could get rings through but no audio or even connecting the call after ringing, and its on a network where ekiga and grandstream phones work just fine with no stun or anything like that (network is private ips but directly routed to the server) Gordon ___ -- 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] building a phone
Again, the main reason for me to require a higher end CPU is audio compression. But I also want the system to be run by a standard OS. It needs to be easy to add your own application there. Mutually exclusive. I don't know any standard OS that doesn't waste about 10 x as many CPU cycles as a SIP phone should ever need. The problem is generalization. A standard OS is designed to support a wide variety of devices, including a wide range of screen sizes. The abstraction layers that make this possible often consume more CPU resources than the application they are supporting. Most of that isn't needed for this application. Compatibility with WXVGA isn't required. Even a full blown file system is a luxury. Linux is about the closest thing because it can be pared down. But it takes someone with considerable experience to know how and what to trim. I supervised a system that used busy box to create a compact system that lived on a small flash card an some RAM. As an example, I have been a Palm owner for a number of years. I laughed when the Win CE stuff came out to compete. The Palm OS was written for the task at hand. I could go a week or more on a charge. The Win CE devices had to be recharged after 8 hours! Why? The OS required too much which required far more compute power, which ate batteries. The SIP phone you propose could be done with about 1 W of power plus a couple more for backlighting. An OS based version would start at 5 W + backlight and could easily go to 15 W or higher. Not the end of the world, I suppose on a desk (if there aren't a hundred of them and I'm not paying the electric bill) but a huge difference if it has to run on batteries. Wilton ___ -- 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] building a phone
Tzafrir Cohen wrote: On Fri, Feb 27, 2009 at 05:48:28PM +, Gordon Henderson wrote: On Fri, 27 Feb 2009, Tzafrir Cohen wrote: The core of the system is: http://www.marvell.com/products/embedded_processors/kirkwood/index.jsp (yes, also two TDM ports with slics. No idea how to use them) That plug supports running quite a number of Linux distributions (Debian, Fedora, Gentoo, and some others) What would it take me to port Asterisk or Yate to it? An existing web interface? You don't want a PBX in it - you want a command-line VoIP client. I was asking (here and elsewhere) if such a beast existed some time back. All I got back was put asterisk on it and use the OSS/ALSA stuff. Talk about using a sledge hammer to crack a nut.. I had a quick look at libiax and ran out of time... Yate is the only h32, IAX and SIP free software soft phone I know. I figure adding one extra interface wouldn't be that tough :-) Linphone has a command-line interface. Not sure about others. But then again, maybe the system will run X11. That can simplify some hings. ekiga (formerly gnome-meeting) seems to be the most included in distros etc., but it has a lot of shortcomings and is buggy IMO - continuously reruns wizard for some reason on a mint distro, wants to bind to vpn ip when its up instead of sticking to the previous setting, needs -c in the command line for calls so can't easily handle a sip url from a browser with a standard setting for helpers, barfs when you run from command line like that if there is already an instance running (should - just tell the original instance to handle the call via some form of ipc and shut down the new instance) they all seem like pretty easy problems to fix but how did something get to this state of maturity with some major shortcomings ? anyone know anything better ? ___ -- 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] building a phone
Wilton Helm wrote: Again, the main reason for me to require a higher end CPU is audio compression. But I also want the system to be run by a standard OS. It needs to be easy to add your own application there. Mutually exclusive. I don't know any standard OS that doesn't waste about 10 x as many CPU cycles as a SIP phone should ever need. The problem is generalization. A standard OS is designed to support a wide variety of devices, including a wide range of screen sizes. The abstraction layers that make this possible often consume more CPU resources than the application they are supporting. Most of that isn't needed for this application. Compatibility with WXVGA isn't required. Even a full blown file system is a luxury. This is not entirely true - many of the nokia phones use a java OS as a core, and you can load pretty much any java software you want on them, but all the points about power and battery use are still valid. (and whether you really consider that truly an OS is questionable, but its out there) Linux is about the closest thing because it can be pared down. But it takes someone with considerable experience to know how and what to trim. I supervised a system that used busy box to create a compact system that lived on a small flash card an some RAM. As an example, I have been a Palm owner for a number of years. I laughed when the Win CE stuff came out to compete. The Palm OS was written for the task at hand. I could go a week or more on a charge. The Win CE devices had to be recharged after 8 hours! Why? The OS required too much which required far more compute power, which ate batteries. The SIP phone you propose could be done with about 1 W of power plus a couple more for backlighting. An OS based version would start at 5 W + backlight and could easily go to 15 W or higher. Not the end of the world, I suppose on a desk (if there aren't a hundred of them and I'm not paying the electric bill) but a huge difference if it has to run on batteries. Wilton ___ -- 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] building a phone
On Fri, Feb 27, 2009 at 01:07:57PM -0500, Jon Pounder wrote: this mentality mystifies me too - why would you want a pbx running on a handset ? ok so you can, but once that novelty wears off in 5secs what is the practical use ? The only reason would be if you don't have any other phones, and if you don't why do you need a pbx to start with ? Because almost all the free software phones I know are X11 :-) I would prefer a decent phone. But then again, if a certain software is also a PBX in addition to being a phone, I don't really care. I would care if being a PBX makes it worse as a phone than some alternatives. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] building a phone
On Fri, Feb 27, 2009 at 11:11:35AM -0700, Wilton Helm wrote: Again, the main reason for me to require a higher end CPU is audio compression. But I also want the system to be run by a standard OS. It needs to be easy to add your own application there. Mutually exclusive. I don't know any standard OS that doesn't waste about 10 x as many CPU cycles as a SIP phone should ever need. It's not about what is wasted. It's about what you're left with. Anyway, my new home PBX is an Alix (alix6b2) unit: http://www.pcengines.ch/alix6b2.htm It's running Debian Lenny. chao:~# cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 5 model : 10 model name : Geode(TM) Integrated Processor by AMD PCS stepping: 2 cpu MHz : 498.056 cache size : 128 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de pse tsc msr cx8 sep pge cmov clflush mmx mmxext 3dnowext 3dnow bogomips: 998.10 clflush size: 32 power management: chao:~# free total used free sharedbuffers cached Mem:256640 251900 4740 0 78488 145972 -/+ buffers/cache: 27440 229200 Swap: 224868 52 224816 It also has an openvpn link, an Astribank device and the network of my very small LAN. Its standard load avarage is, well, 0. I never even bothered optimizing it. The problem is generalization. A standard OS is designed to support a wide variety of devices, including a wide range of screen sizes. The abstraction layers that make this possible often consume more CPU resources than the application they are supporting. Most of that isn't needed for this application. Compatibility with WXVGA isn't required. Even a full blown file system is a luxury. Ask yourself how all those small mp4 players even work. I suppose most of them run Linux. See also http://www.rockbox.org/ Linux is about the closest thing because it can be pared down. But it takes someone with considerable experience to know how and what to trim. I supervised a system that used busy box to create a compact system that lived on a small flash card an some RAM. This trimming can also be automated, if I see that a standard distribution will not do. See e.g. http://astlinux.org/ . As an example, I have been a Palm owner for a number of years. I laughed when the Win CE stuff came out to compete. The Palm OS was written for the task at hand. I could go a week or more on a charge. The Win CE devices had to be recharged after 8 hours! Why? The OS required too much which required far more compute power, which ate batteries. The device does not run on batteries. The plug I mentioned above is 5W. Here's what happens when someone starts optimizing the power consumption: http://www.rowetel.com/blog/?p=61 Anyway, a general-purpose OS consumes extra memory, but relatively little power. I expect that a SIP phone (Requiring transport of many small packet = many interrupts) is a large power consumer. The SIP phone you propose could be done with about 1 W of power plus a couple more for backlighting. An OS based version would start at 5 W + backlight And this is based on? and could easily go to 15 W or higher. Not the end of the world, I suppose on a desk (if there aren't a hundred of them and I'm not paying the electric bill) but a huge difference if it has to run on batteries. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] call file concurrency
Here is a link to a better, but possibly dangerous answer. http://www.netadmintools.com/art295.html Since a typical linux box probably allows about 250K files to be simultaneously open, and you need about 2K for system and * overhead, by cutting the max number of files down to about 3K, you would limit the number of calls to about 1K, assuming that each open call is one file handle. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, February 27, 2009 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call file concurrency Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? On Fri, 27 Feb 2009, Danny Nicholas top posted: Some variant of the ulimit command would accomplish this but YMMV and Caveat Emptor. Which one? -fs::sedwards:~$ ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited file size (blocks, -f) unlimited pending signals (-i) 1024 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 1024 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 16114 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited Limiting the number of open files or file locks would not have the intended effect. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] building a phone
This is not entirely true - many of the nokia phones use a java OS as a core, and you can load pretty much any java software you want on them, but all the points about power and battery use are still valid. (and whether you really consider that truly an OS is questionable, but its out there) Java is the worst offender. Its resource requirements often exceed those of the application it is running. Java is useful for things like displaying web pages that are not time critical and where its write once, run everywhere philosophy is valuable. But anyone trying to actually do things like I/O control, call setup, transcoding, etc. in Java are asking for every issue I raised. If WCE can get 8 hours of battery life, Java would be about 3. Wilton ___ -- 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] call file concurrency
Set the ctime of the spool file in the future and Asterisk will not process the file until that time. Danny Nicholas wrote: Here is a link to a better, but possibly dangerous answer. http://www.netadmintools.com/art295.html Since a typical linux box probably allows about 250K files to be simultaneously open, and you need about 2K for system and * overhead, by cutting the max number of files down to about 3K, you would limit the number of calls to about 1K, assuming that each open call is one file handle. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, February 27, 2009 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call file concurrency Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? On Fri, 27 Feb 2009, Danny Nicholas top posted: -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- 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] Continue in dialplan on hangup
Is there a way to force a channel to continue in the dialplan after the remote end hangs up? Specifically, I am trying to play around with setting up a fax server. I can receive the fax, but sometimes the sending fax hangs up before my System command for printing can run and the fax never prints. I know I can work around by setting up a custom context and use the 'h' extension, but I am hoping for a more simple method. Daniel ___ -- 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] Continue in dialplan on hangup
Daniel Hazelbaker wrote: Specifically, I am trying to play around with setting up a fax server. I can receive the fax, but sometimes the sending fax hangs up If your looking into setting up a reliable fax server and your not doing it over IP, then your best results will be using HylaFAX+ and iaxmodem with Asterisk. HylaFAX+ handles the printing/re-faxing/fax2email of all inbound/outbound faxes via it's FaxDispatch script. It's a 'Set and forget (tm)' package. I absolutely love it. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Continue in dialplan on hangup
Daniel Hazelbaker wrote: Um.. Your=You're! -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Current state of Asterisk and Virtualization?
2009/2/27 John Todd jt...@digium.com: On Feb 26, 2009, at 6:01 PM, Senad Jordanovic wrote: Gavin Henry wrote: Hi all, In a pure VoIP env, what is the current state of do's and don't s of virtualizing * in order to provide multiple separate instances, say for hosting lots of Asterisk-gui/FreePBX/a-n-other gui? I've read lots of threads going back to 2007 and I'm in the general option that kvm is the way to go now, if at all. If dadhi_dummy/zt_dummy is still an issue for conferencing etc. a conference box could be put along side the vm hardware and have a card in it. Thoughts, experiences and being told to shut up are all very much appreciated. Thanks. http://www.bicomsystems.com/products/C/P/797/411/ ...and also: http://voxilla.com/2009/02/12/amazon-ec2-voip-1096 http://voxilla.com/2009/02/13/asterisk-amazon-ec2-1178 http://voxilla.com/2009/02/18/asterisk-on-the-cloud-with-a-click-1405 and: http://www.simionovich.com/?p=180 http://www.simionovich.com/?p=243 ...and lots of others on Nir's blog. Even MORE resources/questions/answers: http://www.google.com/search?hl=enq=ztxenbtnG=Google+Searchaq=foq= http://www.google.com/search?num=30hl=ensafe=offq=xen+and+ztdummybtnG=Search JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ Great, links. Will be back with comments/questions later. Thanks. ___ -- 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] call file concurrency
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, February 27, 2009 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call file concurrency Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? On Fri, 27 Feb 2009, Danny Nicholas top posted: Some variant of the ulimit command would accomplish this but YMMV and Caveat Emptor. On Fri, 27 Feb 2009, Steve Edwards wrote: Which one? -fs::sedwards:~$ ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited file size (blocks, -f) unlimited pending signals (-i) 1024 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 1024 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 16114 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited Limiting the number of open files or file locks would not have the intended effect. On Fri, 27 Feb 2009, Danny Nicholas top posted: Here is a link to a better, but possibly dangerous answer. http://www.netadmintools.com/art295.html Since a typical linux box probably allows about 250K files to be simultaneously open, and you need about 2K for system and * overhead, by cutting the max number of files down to about 3K, you would limit the number of calls to about 1K, assuming that each open call is one file handle. I think proposing to control the number of concurrently processed call files by inducing file descriptor exhaustion is about 32 days premature. Calls would fail at random and you may or may not be able to log in or even execute a command line depending on if you were currently exhausted at any particular instant. I think the OP is looking for some knob to turn in pbx_spool.c Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] call file concurrency
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, February 27, 2009 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call file concurrency Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? On Fri, 27 Feb 2009, Eric Wieling, Asteria Solutions Group top posted: Set the ctime of the spool file in the future and Asterisk will not process the file until that time. This only controls when, not how many. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] call file concurrency
Agreed, but the OP seemed to be looking for a command-line solution, not a C one. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, February 27, 2009 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call file concurrency -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, February 27, 2009 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call file concurrency Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? On Fri, 27 Feb 2009, Danny Nicholas top posted: Some variant of the ulimit command would accomplish this but YMMV and Caveat Emptor. On Fri, 27 Feb 2009, Steve Edwards wrote: Which one? -fs::sedwards:~$ ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited file size (blocks, -f) unlimited pending signals (-i) 1024 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 1024 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 16114 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited Limiting the number of open files or file locks would not have the intended effect. On Fri, 27 Feb 2009, Danny Nicholas top posted: Here is a link to a better, but possibly dangerous answer. http://www.netadmintools.com/art295.html Since a typical linux box probably allows about 250K files to be simultaneously open, and you need about 2K for system and * overhead, by cutting the max number of files down to about 3K, you would limit the number of calls to about 1K, assuming that each open call is one file handle. I think proposing to control the number of concurrently processed call files by inducing file descriptor exhaustion is about 32 days premature. Calls would fail at random and you may or may not be able to log in or even execute a command line depending on if you were currently exhausted at any particular instant. I think the OP is looking for some knob to turn in pbx_spool.c Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Echo on SIP to SIP calls?
I know the subject of echo has been discussed ad nauseum, but I think I have a somewhat unusual problem. I am suddenly experiencing occasional echo on SIP to SIP calls. This is a new development and has never happened in all the years we've been running *. The phones involved are not junk phones (Cisco 7960's and Linksys 942's). I don't recall seeing any settings anywhere than have anything to do with echo cancellation on non-ZAP devices. Anyone have a clue where I should start looking? TIA Bruce Komito WPTI Telecom (775) 236-5815 ___ -- 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] Echo on SIP to SIP calls?
On Fri, 27 Feb 2009, Bruce Komito wrote: I know the subject of echo has been discussed ad nauseum, but I think I have a somewhat unusual problem. I am suddenly experiencing occasional echo on SIP to SIP calls. This is a new development and has never happened in all the years we've been running *. The phones involved are not junk phones (Cisco 7960's and Linksys 942's). I don't recall seeing any settings anywhere than have anything to do with echo cancellation on non-ZAP devices. Anyone have a clue where I should start looking? TIA Bruce Komito WPTI Telecom (775) 236-5815 Try adjusting the volume on the mics on either end. You could be getting feedback. Happens when I make SIP to SIP calls and I keep my phones on speakerphones. And before it's asked, has happened to Snom's, Polycoms, 7960's a 7970, Aastra. =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP Enough research will tend to support your conclusions. - Arthur Bloch A conclusion is the place where you got tired of thinking - Arthur Bloch 227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x5CCD6B5E ___ -- 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] Echo on SIP to SIP calls?
It can only be acoustic echo. Asterisk doesn't cancel that - it's the phone's job. Maybe it will fix it to reduce volume of the phones. Steve On 2/27/09, Bruce Komito bru...@bagel.com wrote: I know the subject of echo has been discussed ad nauseum, but I think I have a somewhat unusual problem. I am suddenly experiencing occasional echo on SIP to SIP calls. This is a new development and has never happened in all the years we've been running *. The phones involved are not junk phones (Cisco 7960's and Linksys 942's). I don't recall seeing any settings anywhere than have anything to do with echo cancellation on non-ZAP devices. Anyone have a clue where I should start looking? TIA Bruce Komito WPTI Telecom (775) 236-5815 ___ -- 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 -- Sent from my mobile device ___ -- 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] call file concurrency
Sent: Friday, February 27, 2009 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call file concurrency Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? On Fri, 27 Feb 2009, Danny Nicholas top posted: Some variant of the ulimit command would accomplish this but YMMV and Caveat Emptor. On Fri, 27 Feb 2009, Steve Edwards wrote: I think proposing to control the number of concurrently processed call files by inducing file descriptor exhaustion is about 32 days premature. Calls would fail at random and you may or may not be able to log in or even execute a command line depending on if you were currently exhausted at any particular instant. I think the OP is looking for some knob to turn in pbx_spool.c On Fri, 27 Feb 2009, Danny Nicholas top posted: Agreed, but the OP seemed to be looking for a command-line solution, not a C one. The OP didn't specify what kind of solution they were looking for. I wouldn't have considered introducing instability as a solution. It seems a reasonable request. Maybe the OP would like to request a feature or offer a bounty? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] TE121B server recommendation
Hello, If anyone is using a TE121B card and it works reliably (i.e. no HDLC Bad FCS or similar errors), could you pass along the make, model, and basic configuration of your Asterisk server? We tried upgrading our old Dell PowerEdge server to a SuperMicro system, but that didn't help. I would like a solid recommendation before I suggest another purchase. Thanks. -- Kevin DeGraaf ___ -- 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] changing /etc/dahdi/system.conf
Great -- thanks for that. Brandon. On Fri, Feb 27, 2009 at 8:34 AM, John Todd jt...@digium.com wrote: On Feb 26, 2009, at 3:05 PM, Tzafrir Cohen wrote: On Thu, Feb 26, 2009 at 01:44:19PM -0700, Brandon B. wrote: At the top of my /etc/dahdi/system.conf file is this line: # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009 -- do not hand edit OK, so how do I adjust the timing source and LBO numbers, and echo cancellers if I'm not supposed to edit this file? Hmm I guess you should n't take that too seriously :-( Maybe this text should be changed. Go, go, gadget bugtracker! http://bugs.digium.com/view.php?id=14569 JT --- John Todd email:jt...@digium.comemail%3ajt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- 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] TE121B server recommendation
Any Supermicro server configuration running a mobo with the PDSMA+ chipset has been bulletproof for us. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin DeGraaf Sent: Friday, February 27, 2009 3:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] TE121B server recommendation Hello, If anyone is using a TE121B card and it works reliably (i.e. no HDLC Bad FCS or similar errors), could you pass along the make, model, and basic configuration of your Asterisk server? We tried upgrading our old Dell PowerEdge server to a SuperMicro system, but that didn't help. I would like a solid recommendation before I suggest another purchase. Thanks. -- Kevin DeGraaf ___ -- 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] what is the effect of high LBO settings?
I'm working on an Asterisk system with all of it's PRI ports configured with the LBO setting at 5, like this: span=3,0,5,esf,b8zs As of yet, I am unwilling to change the LBO to 0 to where it probably should be because the system is working and I'm not sure exactly what the LBO does. I'm aware some changes were made to deal with low audio levels. Does anybody know what audio effects could this possibly have with short cabling? Also, I am trying to cross connect with another Asterisk system with the normal LBO setting (i.e. span=1,1,0,esf,b8zs) but as of yet the systems aren't seeing each other at all. Could the side with the high LBO be confusing the other side somehow? Brandon. ___ -- 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] dialing timing problem?
Preparing to use * for a 'real' installation shortly.
Meanwhile, I've got a single port clone thing, 00:06.0 Communication
controller: Motorola Wildcard X100P working to answer my landline and send
calls to my laptop or voicemail. Sweet!
Trying to call out from linphone, I set up this:
exten = _X.,1,Dial(DAHDI/1,${EXTEN})
Both SIP client and this extension are in default context.
Now, I get a dial tone, but the dialing is just not happening. Some tones are
generated, but not in time. Seems random... Clock issues?
OTOH, I can originate from the CLI to my SIP extension. What does this mean?
Any help with troubleshooting this appreciated. Some info:
/etc/asterisk/sip.conf
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
[authentication]
[basic-options](!)
dtmfmode=rfc2833
context=from-office
type=friend
[natted-phone](!,basic-options)
nat=yes
canreinvite=no
host=dynamic
[public-phone](!,basic-options)
nat=no
canreinvite=yes
[my-codecs](!)
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
[ulaw-phone](!)
disallow=all
allow=ulaw
[linphone]
type=friend
context=default
secret=secret
host=dynamic
dtmfmode=rfc2833
defaultuser=mykhyggz
defaultip=192.168.0.100
/etc/asterisk/chan_dahdi.conf
[trunkgroups]
[channels]
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
signalling=fxs_ks
group=1
channel = 1
#include /etc/asterisk/dahdi-channels.conf
; Autogenerated by /usr/sbin/dahdi_genconf on Thu Feb 26 16:17:56 2009 -- do
not hand edit
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/asterisk/chan_dahdi.conf that will include the
global settings
;
; Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER)
;;; line=1 WCFXO/0/0 FXSKS (In use)
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel = 1
callerid=
group=
context=default
I don't really know if these configs are all that matters to this problem.
Kernel clock/timer stuff:
zgrep RTC /proc/config.gz |grep -vP '^#'
CONFIG_RTC_LIB=m
CONFIG_RTC_CLASS=m
CONFIG_RTC_INTF_SYSFS=y
CONFIG_RTC_INTF_PROC=y
CONFIG_RTC_INTF_DEV=y
zgrep TIMER /proc/config.gz |grep -vP '^#'
CONFIG_TIMERFD=y
CONFIG_HPET_TIMER=y
CONFIG_X86_PM_TIMER=y
CONFIG_HANGCHECK_TIMER=y
CONFIG_SND_TIMER=m
# lsmod |grep rtc
genrtc 6092 0
# lsmod |grep timer
snd_timer 15428 2 snd_seq,snd_pcm
snd37348 11
snd_seq,snd_pcm_oss,snd_mixer_oss,snd_ens1371,snd_rawmidi,snd_seq_device,snd_ac97_codec,snd_pcm,snd_timer
Cheers,
--
|\ /|| | ~ ~
| \/ ||---| `|` ?
||ichael | |iggins\^ /
michael.higgins[at]evolone[dot]org
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Re: [asterisk-users] Continue in dialplan on hangup
On Friday 27 February 2009 14:03:19 Doug Lytle wrote: Daniel Hazelbaker wrote: Specifically, I am trying to play around with setting up a fax server. I can receive the fax, but sometimes the sending fax hangs up If your looking into setting up a reliable fax server and your not doing it over IP, then your best results will be using HylaFAX+ and iaxmodem with Asterisk. HylaFAX+ handles the printing/re-faxing/fax2email of all inbound/outbound faxes via it's FaxDispatch script. It's a 'Set and forget (tm)' package. I absolutely love it. Doug Or, if you're using Asterisk 1.6 and looking to try something new, take a look at http://messinet.com/AsteriskFAXGateway -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] dialing timing problem?
Michael Higgins wrote:
exten = _X.,1,Dial(DAHDI/1,${EXTEN})
This should be _X.,1,Dial(DAHDI/g1/${EXTEN})
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] [asterisk-biz] Switch Options for a service provider
Ignacio, Our Enswitch product matches all these requirements; indeed it goes well beyond them: - We scale far beyond 3000-4000 concurrent calls. We'd consider such a system medium sized. At this size the system is fully failover/redundant, and we have solved the telephony problems of queues, conferences, transfers, etc, with calls on multiple machines. - Full integrated prepaid/postpaid billing and invoicing. - Full real-time reports via web, SOAP API, and direct MySQL. - Full reseller platform with resellers able to set their own pricing, and able to rebrand the product as their own. - Full LCR and carrier failover. - You have root access to the machines and MySQL database. - Full hosted PBX and ITSP features for your customers which they can administer themselves using the web and/or SOAP API. It's in production today from a few hundred users on a single machine to over 150,000 users on large clusters. For more details, please see the links at the bottom of: http://integrics.com/products/enswitch/ In particular, I suggest reading the feature list link. Please do contact me off list for pricing and to arrange a demo installation. Alistair Cunningham +1 888 468 3111 +44 20 799 39 799 http://integrics.com/ Ignacio Ortega A. wrote: Hi, I have a growing voip business Im i looking a solution that can handle at least 3000-4000 concurrent calls with great performance. Also with a billing platform, reports, reseller platform, LCR, call routing,real time reports, SQL dababase access real time Load Reports. Any recommendation? Thanks Ignacio ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ -- 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] dahdi wcb4xxp and fax
I have a couple of suggestions: Make sure that your timing configuration is correct in /etc/dahdi/system.conf (that it has a valid timing source). Also, you probably will probably want to use the half_full buffer policy, and set the number of buffers used to something reasonable, like 8, to ensure you don't have any transmit buffer underruns on the B410P. You shouldn't need more than that, since you're not trying to deal with clock slips or timing drift in this configuration. You may also try explicitly disabling the echo canceller. It seems that sometimes the CED tone detection (which disables the EC) takes a really long period of time to happen, and if it does, disabling the EC in the middle of the fax will usually cause a fax failure. Matthew Fredrickson Digium, Inc. Olivier wrote: 2009/2/25 stoffell stoff...@gmail.com mailto:stoff...@gmail.com Hi all, I wanted to switch from my current setup (mISDN) to the native dahdi with b410p support (wcb4xp). All works fine for normal phone calls but not for faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual bad stuff (HDLC frame not byte-oriented.) What about outgoing faxes ? Our setup uses a digium b410p card with asterisk 1.6, latest libpri and dahdi, hylafax with iaxmodem, and all this on 1 machine. chan_dahdi.conf contains: faxdetect=both When receiving a fax call, hylafax (iaxmodem) answers the call after the obligatory wait of 3 seconds (fax detection) but to me it seems that echo cancellation is still being done. Theory is that any echo canceller hearing a 2100Hz fax signal would halt itself, so I wouldn't search in that direction first. Have you tried native 1.6 sendFax, receiveFax ? Maybe it would improve fax performance. Any pointers on this or workarounds? We're back to our old misdn setup for now ;) Here's some output from dahdi show channel 1 (the one that had the fax connection going), i cut out some non-related stuff : *CLI dahdi show channel 4 Signalling Type: ISDN BRI Point to Point Owner: DAHDI/4-1 Real: DAHDI/4-1 Callwait: None Threeway: None Confno: -1 DSP: yes Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: yes Pulse phone: no DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently ON PRI Flags: Call PRI Logical Span: Implicit Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Regards, stoffell ___ -- 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] call file concurrency
On Fri, Feb 27, 2009 at 4:14 AM, Christian Victor christ...@victormedia.de wrote: 2009/2/27 Bill Michaelson b...@cosi.com Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? Afaik only by limiting the number of call files in the directory. If you can get the outgoing directory (or a reaonable parent) on its own mountable partition or volume, you could accomplish this with disk quotas. It won't control how many Asterisk processes at once (does it even handle them in parallel?), but it will control how many can possibly be queued up waiting to be processed. -James ___ -- 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] Continue in dialplan on hangup
On Feb 27, 2009, at 1:35 PM, Anthony Messina wrote: On Friday 27 February 2009 14:03:19 Doug Lytle wrote: Daniel Hazelbaker wrote: Specifically, I am trying to play around with setting up a fax server. I can receive the fax, but sometimes the sending fax hangs up If your looking into setting up a reliable fax server and your not doing it over IP, then your best results will be using HylaFAX+ and iaxmodem with Asterisk. HylaFAX+ handles the printing/re-faxing/fax2email of all inbound/outbound faxes via it's FaxDispatch script. It's a 'Set and forget (tm)' package. I absolutely love it. Doug Or, if you're using Asterisk 1.6 and looking to try something new, take a look at http://messinet.com/AsteriskFAXGateway I'll take a look at both packages. I hadn't given HylaFAX(+) any thought as when I searched initially I found just the old version of HylaFAX that last had a release in 2007, which makes me a bit nervous. :) Daniel Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ -- 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] Continue in dialplan on hangup
On Friday 27 February 2009 17:02:16 Daniel Hazelbaker wrote: Or, if you're using Asterisk 1.6 and looking to try something new, take a look at http://messinet.com/AsteriskFAXGateway I'll take a look at both packages. I hadn't given HylaFAX(+) any thought as when I searched initially I found just the old version of HylaFAX that last had a release in 2007, which makes me a bit nervous. :) hylafax is excellent. i used it myself. i was just looking to try and start something simpler. also, hylafax uses the concept of modems, so even if you're doing an all-software solution, you'll still need iaxmodem or t38modem. in asterisk 1.6, the SendFAX and ReceiveFAX applications do all of that work for you and all you need is a way to streamline getting faxes into and out of asterisk. in my case, with the AsteriskFAXGateway, it's e-mail. but if you look at the script, it's just a bash script and rather than having incoming faxes be e-mailed, just have them go to a printer. here is the current script: http://messinet.com/viewvc/asterisk-fax-gw/trunk/fax-gw?view=markup on a side note, i'dlove some folks who are willing to test and help me work out scenarios that i've not thought of. there has been interest on digium's side of getting this to be part of the default tarball, but i'd like to get some testing and feedback (and devel help) before i do that. -a -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] call file concurrency
On Fri, 27 Feb 2009, James Sneeringer wrote: If you can get the outgoing directory (or a reaonable parent) on its own mountable partition or volume, you could accomplish this with disk quotas. It won't control how many Asterisk processes at once (does it even handle them in parallel?), but it will control how many can possibly be queued up waiting to be processed. I'm thinking bad things could happen if a call fails (causing the call file to be appended) when you are at the limit. Also, this implies that the process creating the call files can handle the quota error. This also creates a bit of a land mine for the next admin when he replaces the failed disk with one without the quota. I think it should be handled by munging the code in pbx_spool.c. I took a casual peek at the (1.2) code this morning, so don't hold me to my opinions :) The call file directory is scanned every once in a while and for each eligible call file, a detached thread is kicked off to handle it. Limiting the number of concurrent threads (call files) would mean incrementing a [locked] counter as each thread is created and decrementing the [locked, non-zero] counter as each thread finishes. Then, in the loop that scans the directory, if the counter is greater than the desired limit, exit the loop. I'm sure there are more details to be worked out, but I think this could be done easily. At the same time, it might be nice to add a feature to throttle the thread creation so that if a bunch of call files are dumped into the directory Asterisk doesn't spike trying to create concurrent-limit threads at once. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Question about Do Not Disturb
Thank you Gordon and Alexander. With your help, I got it working like
so:
[app-dnd-on]
exten = *78,1,Answer
exten = *78,n,NoOp(${CALLERID(num)} is going on DND ACTIVE)
exten = *78,n,Set(DB(DND/${CALLERID(num)})=On)
exten = *78,n,Playback(do-not-disturbactivated)
exten = *78,n,Hangup
[app-dnd-off]
exten = *79,1,Answer
exten = *79,n,NoOp(${CALLERID(num)} is going OFF DND)
exten = *79,n,DBdel(DND/${CALLERID(num)})
exten = *79,n,Playback(do-not-disturbde-activated)
exten = *79,n,Hangup
[...]
exten = _,1,Set(DND=${DB(DND/${EXTEN})})
exten = _,n,NoOp(For ${EXTEN}, DND is ${DND})
exten = _,n,GotoIf(${DND}?unavailable)
[... some normal dialing stuff ...]
exten = _,n(unavailable),Wait(2)
exten = _,n,Congestion
Works like a charm. Thank you again,
Haim.
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Re: [asterisk-users] call file concurrency
Steve Edwards wrote: On Fri, 27 Feb 2009, James Sneeringer wrote: If you can get the outgoing directory (or a reaonable parent) on its own mountable partition or volume, you could accomplish this with disk quotas. It won't control how many Asterisk processes at once (does it even handle them in parallel?), but it will control how many can possibly be queued up waiting to be processed. I'm thinking bad things could happen if a call fails (causing the call file to be appended) when you are at the limit. Also, this implies that the process creating the call files can handle the quota error. This also creates a bit of a land mine for the next admin when he replaces the failed disk with one without the quota. I think it should be handled by munging the code in pbx_spool.c. I took a casual peek at the (1.2) code this morning, so don't hold me to my opinions :) The call file directory is scanned every once in a while and for each eligible call file, a detached thread is kicked off to handle it. Limiting the number of concurrent threads (call files) would mean incrementing a [locked] counter as each thread is created and decrementing the [locked, non-zero] counter as each thread finishes. Then, in the loop that scans the directory, if the counter is greater than the desired limit, exit the loop. I'm sure there are more details to be worked out, but I think this could be done easily. At the same time, it might be nice to add a feature to throttle the thread creation so that if a bunch of call files are dumped into the directory Asterisk doesn't spike trying to create concurrent-limit threads at once. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 Just my 2c, but what I've done in the past is modify the sleep function in asterisk from one based on seconds to one based on either milliseconds or nanoseconds (don't remember which). Then I have a background daemon which looks to see how many files are in the directory, and if it's under threshold it pushes a new file from a queue into the directory. Then, as they say above, you set the ulimit to something like 'ulimit -n 10' or whatever it is you want. Of course, the purpose for sending out a bazillion calls is another question . . . play nice. ___ -- 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] rfc2833 vs. sipinfo and network weirdness
Further to a recent post about a problem whereby the server continues to spew packets to the phone after hangup (sometimes, not every time), I have found that this problem appears to be alleviated by using RFC2833 instead of SIP INFO, however in switching to RFC2833 I introduce another problem - that DTMF tones for navigating menus become unreliable. RFC2833 and SIP INFO are the only 2 options supported by the phone. Inbetween the phone and the server is the internet and 1 NAT traversal. Any ideas? Michael ___ -- 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] what is the effect of high LBO settings?
On Fri, 2009-02-27 at 14:07 -0700, Brandon B. wrote: As of yet, I am unwilling to change the LBO to 0 to where it probably should be because the system is working and I'm not sure exactly what the LBO does. I'm aware some changes were made to deal with low audio levels. LBO stands for Line Built Out... it's essentially a measurement of the distance between your demarcation point (d-marc/smart jack/NIU) and your Asterisk box. As you can see from a sample system.conf (from DAHDI) or zaptel.conf (from Zaptel), it's an integer value from the following table: 0: 0 db (CSU) / 0-133 feet (DSX-1) 1: 133-266 feet (DSX-1) 2: 266-399 feet (DSX-1) 3: 399-533 feet (DSX-1) 4: 533-655 feet (DSX-1) 5: -7.5db (CSU) 6: -15db (CSU) 7: -22.5db (CSU) As I understand it, the LBO is effectively an attenuation value, with a higher number meaning less attenuation. This way, you don't get too hot of a signal with a short cable, or two low of a signal on long cable. Just how far is your Asterisk box from the demarcation point? Also, I am trying to cross connect with another Asterisk system with the normal LBO setting (i.e. span=1,1,0,esf,b8zs) but as of yet the systems aren't seeing each other at all. Could the side with the high LBO be confusing the other side somehow? Shouldn't be... you did use a T1-crossover cable to cross-connect the two Asterisk boxes, right? I've got a little T1 cross-connect diagram at http://www.asteriskdocs.org/cables/ if you need a reference. -- Jared Smith Digium, Inc. | Training Manager ___ -- 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] building a phone
Witness the fact that the old Pingtel phones ran Java, and they were incredibly lame. I think part of what this thread misses is that DSP is a god chunk of what SIP phones need. A general purpose CPU is not the right tool for the task. A cheap DSP is better suited to compression, transcoding, etc. OTOH, presuming that the snom phones are Linux on a suitable platform soomeone could develop a custom software load for them and OEM the hardware. Michael --Original Message Text--- From: Wilton Helm Date: Fri, 27 Feb 2009 12:28:40 -0700 This is not entirely true - many of the nokia phones use a java OS as a core, and you can load pretty much any java software you want on them, but all the points about power and battery use are still valid. (and whether you really consider that truly an OS is questionable, but its out there) Java is the worst offender. Its resource requirements often exceed those of the application it is running. Java is useful for things like displaying web pages that are not time critical and where its write once, run everywhere philosophy is valuable. But anyone trying to actually do things like I/O control, call setup, transcoding, etc. in Java are asking for every issue I raised. If WCE can get 8 hours of battery life, Java would be about 3. Wilton -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- 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] connecting 66 analog phones to asterisk - hardware suggestions
Hi all, thanks for the excellent information about the banks and usb banks. some tech details will prevent us from using usb units. The trunks will be 500 feet away from the new location of the ip-pbx so we have decided to go with channel banks for the trunks and sending the E1 signal over cat 5 (E1 signal can travel un-repeated over 5000 feet) So far we are reading/evaluating about rhino channel banks and a quad E1/T1 (pci-e) on the asterisk box. thanks again -- Erick Perez ___ -- 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] call file concurrency
IIRC, some early dialler of the pre-AMI era used this technique to control the number of calls placed simoultaneously - they just counted the number of call files in the spool dir. As they are deleted when the call is over, this was a simple way to do the throttling. You could use a similar technique; have call files written to a staging directory and then use a simple process to transfer them to the actual spool dir so that there are never more than N in the spool dir. Thanks l. 2009/2/27 Bill Michaelson b...@cosi.com Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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] building a phone
On Fri, Feb 27, 2009 at 09:39:59PM -0600, Michael Graves wrote: Witness the fact that the old Pingtel phones ran Java, and they were incredibly lame. I think part of what this thread misses is that DSP is a god chunk of what SIP phones need. A general purpose CPU is not the right tool for the task. A cheap DSP is better suited to compression, transcoding, etc. Maybe. But it is also a great limitation on the hackability and thus on the speed of development. I think that general-purpose processors are strong enough for that. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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
