Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime
On Mon, Feb 8, 2010 at 2:52 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Feb 08, 2010 at 02:37:18AM -0500, Steve Totaro wrote: Just start it with safe_asterisk. http://linux.die.net/man/8/safe_asterisk And I take it that the name is intentional. Unless my info is out of date, it will kill two birds with one stone. You're in a lethal mood today :-) -- 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 Just offering up some great info that some brilliant guy documented in a man page. -- _ -- 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] {top|bottom|interleaved} posting, once again
On 6 Feb 2010, at 19:17, Philipp Kempgen wrote: Actually bottom-posting without trimming anything (SCNR) is about as annoying as top-posting. Yup, at least with bottom-posting, you might be reminded to trim down the included text. Top-posters are, IMHO, the worst for adding a single line of text and including reams of previous emails. W-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk going down (Josiah Bryan)
Thanks Josiah Bryan, I do not have any dns server running on my asterisk server, we do have an external DNS server working in the data center; the IP of this dns server is 10.4.1.5... Following you will see my main configuration: /etc/resolv.conf: domain localdomain search localdomain nameserver 10.4.1.5 nameserver 10.4.1.2 /etc/hosts: # Do not remove the following line, or various programs # that require network functionality will fail. 127.0.0.1 localhost.localdomain localhost Thanks in advance... -- _ -- 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] Help with iax.conf {tesco|freshtel} 1.6
I have something going on that I don't fully understand after a weekend
of looking for answers.
I have an iax account with Tesco that works flawlessly with the Zoiper
client - but is giving me trouble with inbound calls in Asterisk 1.6.
After some playing I have ended up with an iax.conf file that looks like
this:
[general]
calltokenoptional = 77.75.0.0/255.255.248.0
maxcallnumbers = 16382
port=4569
bandwidth=low
disallow=all
allow=alaw
allow=ulaw
allow=gsm
jitterbuffer=yes
tos=lowdelay
qualify=80
register = 012:passw...@gateway.tescointernetphone.com
[012]
type=friend
requirecalltoken=no
context=from-iax
host=gateway.tescointernetphone.com
auth=rsa
username=012
secret=PASSWORD
qualify=yes
For testing I've tried various options in extensions.conf, but for the
time being have settled with:
exten = 012,1,NoOp(--- AIX-INBOUND-TESCO DEBUG ---)
exten = 012,2,Wait(10)
exten = 012,3,hangup
Looking at iax2 show peers I have:
Name/UsernameHost Mask Port
Status
012/012 77.75.0.135 (S) 255.255.255.255 4569 OK
(56 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
If I make an INBOUND call to the pstn number associated with this iax
provider, with debugging on I see the call, but it never goes to the
matching extension - it just rings out until I hang up:
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 00013ms SCall: 00219 DCall: 0 [77.75.0.135:4569]
VERSION : 2
CALLED NUMBER : 012
CODEC_PREFS : (ilbc|g729|alaw|ulaw|gsm)
CALLING NUMBER : Unavailable
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME: Unavailable
LANGUAGE: en
FORMAT : 256
CAPABILITY : 58638
ADSICPE : 2
DATE TIME : 2010-02-08 09:04:54
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 00019ms SCall: 00655 DCall: 00219 [77.75.0.135:4569]
AUTHMETHODS : 4
CHALLENGE : \x31\x31\munged
USERNAME: 012
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00019ms SCall: 00219 DCall: 00655 [77.75.0.135:4569]
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 3ms SCall: 02774 DCall: 0 [77.75.0.135:4569]
USERNAME: 012
REFRESH : 60
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REGACK
Timestamp: 00013ms SCall: 00091 DCall: 02774 [77.75.0.135:4569]
USERNAME: 012
DATE TIME : 2010-02-08 09:05:00
REFRESH : 60
APPARENT ADDRES : IPV4 x.x.x.x:4569
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00013ms SCall: 02774 DCall: 00091 [77.75.0.135:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
HANGUP
Timestamp: 08647ms SCall: 00219 DCall: 00655 [77.75.0.135:4569]
CAUSE CODE : 0
However, if I add this to the bottom of iax.conf:
[guest]
type=user
context=from-iax
It works -
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 00018ms SCall: 00423 DCall: 0 [77.75.0.135:4569]
VERSION : 2
CALLED NUMBER : 012
CODEC_PREFS : (ilbc|g729|alaw|ulaw|gsm)
CALLING NUMBER : Unavailable
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME: Unavailable
LANGUAGE: en
FORMAT : 256
CAPABILITY : 58638
ADSICPE : 2
DATE TIME : 2010-02-08 09:18:20
-- Accepting UNAUTHENTICATED call from 77.75.0.135:
requested format = g729,
requested prefs = (ilbc|g729|alaw|ulaw|gsm),
actual format = alaw,
host prefs = (alaw|ulaw|gsm),
priority = mine
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
Timestamp: 00016ms SCall: 03340 DCall: 00423 [77.75.0.135:4569]
FORMAT : 8
stinger2*CLI
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00016ms SCall: 00423 DCall: 03340 [77.75.0.135:4569]
-- Executing [012x...@from-iax:1] NoOp(IAX2/012-3340,
--- AIX-INBOUND-TESCO DEBUG --- ) in new stack
== Begin MixMonitor Recording IAX2/01256510343-3340
-- Executing [01256510...@from-iax:3] Wait(IAX2/01256510343-3340,
1) in new stack
-- Executing [01256510...@from-iax:4] Dial(IAX2/01256510343-3340,
SIP/5050,55,tr) in new stack
{snip}
I am guessing this is either an issue with this:
Accepting UNAUTHENTICATED call from 77.75.0.135:
Or is normal and I'm missing something here
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asterisk-users mailing list
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[asterisk-users] queue with strategy=linear
Hi, Using asterisk 1.6.2.0 I have a queue definition with strategy=linear. How do I jump to the next dialplan item after having tried (unsuccessfully) all queue members? If I use Queue(test,n) then only the first member is contacted. And if I omit the n option then all members are retried indefinitely. 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] Asterisk 1.4.26.2 died after 80 days uptime
8 feb 2010 kl. 08.37 skrev Steve Totaro: On Mon, Feb 8, 2010 at 2:20 AM, Olle E. Johansson o...@edvina.net wrote: 7 feb 2010 kl. 15.09 skrev Per Jessen: Thomas Winter wrote: Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? My asterisk 1.4.23 also dies about once a month. I've never been able to work out why. I haven't seen this, but it is definitely something we should try to catch. It could be a memory leak or another type of leak. Any advice from other developers on how to try to catch this? One thing that would be good would be to get a core dump. There's a document in the /doc directory on how to recompile Asterisk with symbols and force a core dump to happen when we get a crash. /O Just start it with safe_asterisk. http://linux.die.net/man/8/safe_asterisk Unless my info is out of date, it will kill two birds with one stone. Asterisk will restart itself, and you will get a core dump. There was a reason I referred to the documentation ;-) YOu will have to recompile it with the DONT_OPTIMIZE variable set so that the core dump actually has meaningful symbols. /O -- _ -- 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] Not able to compile asterisk, zaptel, libpri in /usr/src
On 8 Feb 2010, at 05:30, Trevor Peirce wrote: aster...@opensourcesolution.in wrote: Not able to compile asterisk,zaptel,libpri in /usr/src Have you tried to run make? Without any information on what you're tried and what error you receive, I can almost guarantee you will not receive any help on this forum. OP is a serial 'shit question' poster unfortunately. If you care to search the archive you'll probably be reduced to tears. Gems such as: 'what is asterisk now' 'i had installed asterisk under /etc. now i want to know by command which version of asterisk i had installed. how to know the version plz tell me.' 'thanks a lot fred for the link.' (new thread) S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime
On Mon, Feb 08, 2010 at 11:03:19AM +0100, Olle E. Johansson wrote: You will have to recompile it with the DONT_OPTIMIZE variable set so that the core dump actually has meaningful symbols. Doing so hurts your performance (and also slightly changes the behaviour of the program). Asterisk build by default with debugging symbols. If you installed from a binary package, chances are that debugging symbols have been stripped but are available in a separate package. -- 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] Can an agent Login to a queue and be paused
I'm not sure if this works for newer versions of Asterisk, but on old ones, you could pause an agent and THEN log him on, and he'd be paused. l. 2010/2/4 Robert Grignon rgrig...@fleetone.com I thought there was an option for this but cant find it We have a busy callcenter and I would like the agents to log in and be in a paused state upon login... Right now they login and they are instantly receiving a call Thanks for the input... -- 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] Asterisk 1.4.26.2 died after 80 days uptime
8 feb 2010 kl. 11.26 skrev Tzafrir Cohen: On Mon, Feb 08, 2010 at 11:03:19AM +0100, Olle E. Johansson wrote: You will have to recompile it with the DONT_OPTIMIZE variable set so that the core dump actually has meaningful symbols. Doing so hurts your performance (and also slightly changes the behaviour of the program). Yes, but if we can get a readable core dump, it is good help. Asterisk build by default with debugging symbols. If you installed from a binary package, chances are that debugging symbols have been stripped but are available in a separate package. Right. /O -- _ -- 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] conferencing without DAHDI
Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. thanks klaus -- _ -- 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] conferencing without DAHDI
8 feb 2010 kl. 12.29 skrev Klaus Darilion: Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. In Asterisk trunk there's a new conference bridge module you can test. There are also some third-party modules out there, like app_conference. /O -- _ -- 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] conferencing without DAHDI
Hi Klaus, The module is app_confbridge, and the application is ConfBridge. I had been using it for a while because it's really easy to use : you don't need any configuration file, and you get cool announcements upon conference events from a playback channel. The options work pretty much like meetme, although I would have liked to have a 'x' option to close the conference when the last marked user leaves. Moreover, I couldn't have the playback channel speak French, from what I've read in the source code, I think that feature would require a configuration file because the playback channel is not a per user option. Philippe On Mon, Feb 8, 2010 at 12:56 PM, Olle E. Johansson o...@edvina.net wrote: 8 feb 2010 kl. 12.29 skrev Klaus Darilion: Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. In Asterisk trunk there's a new conference bridge module you can test. There are also some third-party modules out there, like app_conference. /O -- _ -- 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 -- Philippe Sultan -- _ -- 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] conferencing without DAHDI
And by the way, app_confbridge is included in the 1.6.2 series (at least). On Mon, Feb 8, 2010 at 1:49 PM, Philippe Sultan philippe.sul...@gmail.com wrote: Hi Klaus, The module is app_confbridge, and the application is ConfBridge. I had been using it for a while because it's really easy to use : you don't need any configuration file, and you get cool announcements upon conference events from a playback channel. The options work pretty much like meetme, although I would have liked to have a 'x' option to close the conference when the last marked user leaves. Moreover, I couldn't have the playback channel speak French, from what I've read in the source code, I think that feature would require a configuration file because the playback channel is not a per user option. Philippe On Mon, Feb 8, 2010 at 12:56 PM, Olle E. Johansson o...@edvina.net wrote: 8 feb 2010 kl. 12.29 skrev Klaus Darilion: Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. In Asterisk trunk there's a new conference bridge module you can test. There are also some third-party modules out there, like app_conference. /O -- _ -- 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 -- Philippe Sultan -- Philippe Sultan -- _ -- 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] GSM Gateway
Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign relies on USB and project GSMOPEN. what would you recommend - trunk or usb ? Or there are other possibilities ? Thanks, Peter -- _ -- 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] High codec translation times on x64
Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - -- - - - - - - - - - - - gsm - - 3001 3002 6999 3001 3000 10999 - - 40994 8000 6999 - - 13998 ulaw - 5000 - 1 4000 2 1 8000 - - 37995 5001 4000 - - 10999 alaw - 5000 1 - 4000 2 1 8000 - - 37995 5001 4000 - - 10999 g726aal2 - 8998 4000 4001- 4000 3999 11998 - - 41993 8999 7998 - - 14997 adpcm - 5000 2 3 4000 - 1 8000 - - 37995 5001 4000 - - 10999 slin - 4999 1 2 3999 1 - 7999 - - 37994 5000 3999 - - 10998 lpc10 - 7999 3001 3002 6999 3001 3000 - - - 40994 8000 6999 - - 13998 g729 - - - -- - - - - - - - - - - - speex - - - -- - - - - - - - - - - - ilbc - 11998 7000 700110998 7000 6999 14998 - - - 11999 10998 - - 17997 g726 - 8998 4000 4001 7998 4000 3999 11998 - - 41993 - 7998 - - 14997 g722 - 12998 8000 800111998 8000 7999 15998 - - 45993 12999 - - - 6999 siren7 - - - -- - - - - - - - - - - - siren14 - - - -- - - - - - - - - - - - slin16 - 21996 16998 1699920996 16998 16997 24996 - - 54991 21997 8998 - - - On CentOS 32x rentier*CLI show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- ---- -- -- - --- gsm- -222 21 3- - 162- ulaw- 2-12 21 3- - 162- alaw- 21-2 21 3- - 162- g726aal2- 222- 21 3- - 161- adpcm- 2222 -1 3- - 162- slin- 1111 1- 2- - 151- lpc10- 2222 21 -- - 162- g729- ---- -- -- - --- speex- ---- -- -- - --- ilbc- 3333 32 4- - -3- g726- 2221 21 3- - 16-- g722- ---- -- -- - --- This is strange :-) Chris -- _ -- 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] GSM Gateway
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html We use this one, and it works great.. easy to setup and it works with a normal network connection :) On Mon, Feb 8, 2010 at 1:52 PM, Peter peterp...@aboutsupport.com wrote: Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign relies on USB and project GSMOPEN. what would you recommend - trunk or usb ? Or there are other possibilities ? Thanks, Peter -- _ -- 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 -- Groet // Kind regards, Peter den Hartog -- _ -- 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] High codec translation times on x64
Are you sure you compare apples to apples Here is my output on CentOS 32-bit core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - -- - - - - - - - - - gsm - - 1001 1001 3999 1001 1000 7000 6000 - 3999 3000 1999 ulaw - 3000 - 1 3000 2 1 6001 5001 9000 - 3000 2001 1000 alaw - 3000 1 - 3000 2 1 6001 5001 9000 - 3000 2001 1000 g726aal2 - 5999 3001 3001- 3001 3000 9000 8000 11999 - 1 5000 3999 adpcm - 3000 2 2 3000 - 1 6001 5001 9000 - 3000 2001 1000 slin - 2999 1 1 2999 1 - 6000 5000 8999 - 2999 2000999 lpc10 - 4998 2000 2000 4998 2000 1999 - 6999 10998 - 4998 3999 2998 g729 - 3998 1000 1000 3998 1000 999 6999 - 9998 - 3998 2999 1998 speex - 3999 1001 1001 3999 1001 1000 7000 6000 - - 3999 3000 1999 ilbc - - - -- - - - - - - - - - g726 - 5999 3001 30011 3001 3000 9000 8000 11999 - - 5000 3999 g722 - 6998 4000 4000 6998 4000 3999 8999 12998 - 6998 - 3999 slin16 - 9998 7000 7000 9998 7000 6999 12999 11999 15998 - 9998 6000 - When Asterisk 1.6.x is used AND core show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- ---- -- -- -- -- gsm- -224 21 6-40- 4- ulaw- 4-14 21 6-40- 4- alaw- 41-4 21 6-40- 4- g726aal2- 533- 32 7-41- 1- adpcm- 4224 -1 6-40- 4- slin- 3113 1- 5-39- 3- lpc10- 5335 32 --41- 5- g729- ---- -- -- -- -- speex- 6446 43 8- -- 6- ilbc- ---- -- -- -- -- g726- 5331 32 7-41- -- g722- ---- -- -- -- -- When Asterisk 1.4.x is used -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Brown Sent: Monday, February 08, 2010 7:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] High codec translation times on x64 Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - -- - - - - - - - - - - - gsm - - 3001 3002 6999 3001 3000 10999 - - 40994 8000 6999 - - 13998 ulaw - 5000 - 1 4000 2 1 8000 - - 37995 5001 4000 - - 10999 alaw - 5000 1 - 4000 2 1 8000 - - 37995 5001 4000 - - 10999 g726aal2 - 8998 4000 4001- 4000 3999 11998 - - 41993 8999 7998 - - 14997 adpcm - 5000 2 3 4000 - 1 8000 - - 37995 5001 4000 - - 10999 slin - 4999 1 2 3999 1 - 7999 - - 37994 5000 3999 - - 10998 lpc10 - 7999 3001 3002 6999 3001 3000 - - - 40994 8000 6999 - - 13998 g729 - - - -- - - - - - - - - - - - speex - - - -- - - - - - - - - - - - ilbc - 11998 7000 700110998 7000 6999 14998 - - - 11999 10998 - - 17997 g726 - 8998 4000 4001 7998 4000
Re: [asterisk-users] High codec translation times on x64
i don't think that is high, 64x is at microseconds the 32x is at milliseconds. On 2/8/2010 8:54 PM, Christopher Brown wrote: Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - -- - - - - - - - - - - - gsm - - 3001 3002 6999 3001 3000 10999 - - 40994 8000 6999 - - 13998 ulaw - 5000 - 1 4000 2 1 8000 - - 37995 5001 4000 - - 10999 alaw - 5000 1 - 4000 2 1 8000 - - 37995 5001 4000 - - 10999 g726aal2 - 8998 4000 4001- 4000 3999 11998 - - 41993 8999 7998 - - 14997 adpcm - 5000 2 3 4000 - 1 8000 - - 37995 5001 4000 - - 10999 slin - 4999 1 2 3999 1 - 7999 - - 37994 5000 3999 - - 10998 lpc10 - 7999 3001 3002 6999 3001 3000 - - - 40994 8000 6999 - - 13998 g729 - - - -- - - - - - - - - - - - speex - - - -- - - - - - - - - - - - ilbc - 11998 7000 700110998 7000 6999 14998 - - - 11999 10998 - - 17997 g726 - 8998 4000 4001 7998 4000 3999 11998 - - 41993 - 7998 - - 14997 g722 - 12998 8000 800111998 8000 7999 15998 - - 45993 12999 - - - 6999 siren7 - - - -- - - - - - - - - - - - siren14 - - - -- - - - - - - - - - - - slin16 - 21996 16998 1699920996 16998 16997 24996 - - 54991 21997 8998 - - - On CentOS 32x rentier*CLI show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- ---- -- -- - --- gsm- -222 21 3- - 162- ulaw- 2-12 21 3- - 162- alaw- 21-2 21 3- - 162- g726aal2- 222- 21 3- - 161- adpcm- 2222 -1 3- - 162- slin- 1111 1- 2- - 151- lpc10- 2222 21 -- - 162- g729- ---- -- -- - --- speex- ---- -- -- - --- ilbc- 3333 32 4- - -3- g726- 2221 21 3- - 16-- g722- ---- -- -- - --- This is strange :-) Chris -- _ -- 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] GSM Gateway
On Mon, Feb 08, 2010 at 02:52:33PM +0200, Peter wrote: I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign relies on USB and project GSMOPEN. what would you recommend - trunk or usb ? Or there are other possibilities ? Portech GSM gateways tend to work quite well. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- 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] GSM Gateway
Op 08-02-10 13:52, Peter schreef: Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. Not too fancy, but these work great, and are pretty cheap: http://www.wildix.com/product_info.php?products_id=776cPath= Ron -- NeoNova BV innovatieve internetoplossingen http://www.neonova.nl Science Park 140 1098 XG Amsterdam info: 020-5611300 servicedesk: 020-5611302 fax: 020-5611301 KvK Amsterdam 34151241 Op dit bericht is de volgende disclaimer van toepassing: http://www.neonova.nl/maildisclaimer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime
Box #1 faxserver*CLI core show version Asterisk 1.4.21.2 built by root @ faxserver.localhost on a i686 running Linux on 2008-08-07 20:30:54 UTC faxserver*CLI core show uptime System uptime: 21 weeks, 2 days, 22 hours, 43 minutes, 42 seconds faxserver*CLI this box gets about 200 faxes a day, and does a tone of agi script processing, and network printing. Someday I may upgrade it, but it runs too well for me to want to touch it. Box #2 sip*CLI core show version Asterisk 1.4.26.2 built by root @ ast-two.localhost on a i686 running Linux on 2009-09-05 00:17:05 UTC sip*CLI core show uptime System uptime: 3 weeks, 3 days, 15 hours, 17 minutes, 34 seconds sip*CLI this is my IVR outbound LD box.. Personnel Box for home: localhost*CLI core show version Asterisk 1.4.28 built by root @ localhost.localdomain on a i686 running Linux on 2009-12-20 04:16:08 UTC localhost*CLI core show uptime System uptime: 3 weeks, 1 day, 15 hours, 38 minutes, 5 seconds Last reload: 3 weeks, 1 day, 6 hours, 19 minutes, 54 seconds localhost*CLI Doesn't get many calls at all.. it's just for my house, maybe 10 calls a week.. , and I do a lot of custom network IVR stuff with it.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Per Jessen Sent: Sunday, February 07, 2010 9:09 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime Thomas Winter wrote: Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? My asterisk 1.4.23 also dies about once a month. I've never been able to work out why. /Per Jessen, Zürich -- _ -- 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] Asterisk 1.4.26.2 died after 80 days uptime
After reviewing other emails, you also may want to enable debug logging, and find last log entry before crash.. Also graph cpu load, memory usage, call count.. I had one server that would reboot every few days, turned out the PCI-e bus was not playing nicely with the PRI Card, after switching servers, the crashing went away. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell (Lists) Sent: Monday, February 08, 2010 8:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime Box #1 faxserver*CLI core show version Asterisk 1.4.21.2 built by root @ faxserver.localhost on a i686 running Linux on 2008-08-07 20:30:54 UTC faxserver*CLI core show uptime System uptime: 21 weeks, 2 days, 22 hours, 43 minutes, 42 seconds faxserver*CLI this box gets about 200 faxes a day, and does a tone of agi script processing, and network printing. Someday I may upgrade it, but it runs too well for me to want to touch it. Box #2 sip*CLI core show version Asterisk 1.4.26.2 built by root @ ast-two.localhost on a i686 running Linux on 2009-09-05 00:17:05 UTC sip*CLI core show uptime System uptime: 3 weeks, 3 days, 15 hours, 17 minutes, 34 seconds sip*CLI this is my IVR outbound LD box.. Personnel Box for home: localhost*CLI core show version Asterisk 1.4.28 built by root @ localhost.localdomain on a i686 running Linux on 2009-12-20 04:16:08 UTC localhost*CLI core show uptime System uptime: 3 weeks, 1 day, 15 hours, 38 minutes, 5 seconds Last reload: 3 weeks, 1 day, 6 hours, 19 minutes, 54 seconds localhost*CLI Doesn't get many calls at all.. it's just for my house, maybe 10 calls a week.. , and I do a lot of custom network IVR stuff with it.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Per Jessen Sent: Sunday, February 07, 2010 9:09 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime Thomas Winter wrote: Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? My asterisk 1.4.23 also dies about once a month. I've never been able to work out why. /Per Jessen, Zürich -- _ -- 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] Can an agent Login to a queue and be paused
What Id did was on the dialplan, create an specifica extension for login
agents. Lets say Agent/10017, then
When dial 2110017 the agents is promts for Agent passwd.Then I have a macro
only for pausing agents depending on the meaning.
So if the agent is successfully granted on the Login Context, that same
context goto pause macro.
Quick example:
[queues_logon]
; Agent Login Procedure
exten = _211,1,Answer()
exten = _211,n,NoCDR()
exten = _211,n,GotoIf($[${LEN(${AGENTBYCALLERID_${CALLERID(number)}})}
1 ]?4:5) ; Check that the physical extension is free
exten = _211,n,AgentCallbackLogin(${EXTEN:2},,${CALLERID(number)}) ;
Ask for agent password and log the agent on
exten = _211,n,Macro(agent_pause_reason,${EXTEN:2},30) ; Put Agents
into Initial Paused State on the Queue
exten = _211,n,Hangup()
[macro-agent_pause]
; ${ARG1} - Agent_nro
exten = s,1,PauseQueueMember(|Agent/${ARG1})
exten = s,n,MacroExit
2010/2/8 Lenz Emilitri lenz.lo...@gmail.com
I'm not sure if this works for newer versions of Asterisk, but on old ones,
you could pause an agent and THEN log him on, and he'd be paused.
l.
2010/2/4 Robert Grignon rgrig...@fleetone.com
I thought there was an option for this but cant find it
We have a busy callcenter and I would like the agents to log in and be
in a paused state upon login... Right now they login and they are
instantly receiving a call
Thanks for the input...
--
Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] large scale paging
On 2010-02-06 21:42, C F wrote: For a case like this I would go with overhead paging. On Fri, Feb 5, 2010 at 4:50 PM, Mark Willismarksli...@markwillis.net wrote: Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. They are trying to get away from overhead paging. -- _ -- 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] GSM Gateway
On Mon, 8 Feb 2010, Peter wrote: Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign relies on USB and project GSMOPEN. I've used a Portech unit (2 sims) in recent times. Did what it said it was supposed to do (and still is as far as I'm aware!) Relatively easy to setup and just had a few extralines of dialplan to detect the dialled number and it was was to a UK mobile, then dial it via the Portech, or fail-back to their ISDN lines.. Presumab'y you're using it for LCR type of things - relatively easy to do in the UK, but your country's dialling codes might vary ... 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] Still on spandsp/app_fax and T.38
Unfortunely it didn't solve the problem. Here's the session packet capture after editing app_fax.c. http://www.canall.com.br/wireshark_t38_jbig.gz Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Kevin P. Fleming kpflem...@digium.com escreveu: Vinícius Fontes wrote: Will do. You guys will have my feedback on monday. If everything goes okay with that change, I'll post a patch on Mantis. No need for the patch; it's already on my radar, and if you confirm that it actually solves an interop problem, I'll commit the update to the various branches it belongs in. I'd still like to hear from Steve Underwood if I misinterpreted the MMR/JBIG transcoding function calls in spandsp that led me to enabling these features in the first place... -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] High codec translation times on x64
On Mon, Feb 08, 2010 at 01:54:55PM +0100, Christopher Brown wrote: Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - -- - - - - - - - - - - - gsm - - 3001 3002 6999 3001 3000 10999 - - 40994 8000 6999 - - 13998 Those numbers are the time of translating a relatively short period of time. Thus you hit granularity issues. If you want to get more accurate timing, recalculate it with a longer sample: core show translation recalc length You should probably use at least 10, and preferably 100 or 1000, to get more accurate timing. -- 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] Can an agent Login to a queue and be paused
The agents tried that but it did not work... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Monday, February 08, 2010 4:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can an agent Login to a queue and be paused I'm not sure if this works for newer versions of Asterisk, but on old ones, you could pause an agent and THEN log him on, and he'd be paused. l. 2010/2/4 Robert Grignon rgrig...@fleetone.com I thought there was an option for this but cant find it We have a busy callcenter and I would like the agents to log in and be in a paused state upon login... Right now they login and they are instantly receiving a call Thanks for the input... -- 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] Can an agent Login to a queue and be paused
Not a bad idea... We use queuemetrics and the login is done via Web GUI.
I could easily just send it to pause upon login...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mariano
Lecuona
Sent: Monday, February 08, 2010 8:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can an agent Login to a queue and be
paused
What Id did was on the dialplan, create an specifica extension for login
agents. Lets say Agent/10017, then
When dial 2110017 the agents is promts for Agent passwd.Then I have a
macro only for pausing agents depending on the meaning.
So if the agent is successfully granted on the Login Context, that same
context goto pause macro.
Quick example:
[queues_logon]
; Agent Login Procedure
exten = _211,1,Answer()
exten = _211,n,NoCDR()
exten =
_211,n,GotoIf($[${LEN(${AGENTBYCALLERID_${CALLERID(number)}})} 1
]?4:5) ; Check that the physical extension is free
exten = _211,n,AgentCallbackLogin(${EXTEN:2},,${CALLERID(number)})
; Ask for agent password and log the agent on
exten = _211,n,Macro(agent_pause_reason,${EXTEN:2},30) ; Put
Agents into Initial Paused State on the Queue
exten = _211,n,Hangup()
[macro-agent_pause]
; ${ARG1} - Agent_nro
exten = s,1,PauseQueueMember(|Agent/${ARG1})
exten = s,n,MacroExit
2010/2/8 Lenz Emilitri lenz.lo...@gmail.com
I'm not sure if this works for newer versions of Asterisk, but
on old ones, you could pause an agent and THEN log him on, and he'd be
paused.
l.
2010/2/4 Robert Grignon rgrig...@fleetone.com
I thought there was an option for this but cant find
it
We have a busy callcenter and I would like the agents to
log in and be
in a paused state upon login... Right now they login and
they are
instantly receiving a call
Thanks for the input...
--
Loway - home of QueueMetrics - http://queuemetrics.com
--
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[asterisk-users] Asterisk how install speex support
Hi, I would like to add support for speex codec in Asterisk. In Ubuntu 9.10 the procedure is the following: 1) sudo apt-get install speex libspeex-dev 2) install Asterisk that enable speex support in configure procedure 3) core show translation I can't see the translation time. Where is the problem? Thank -- BOINC manager http://boincmanager.altervista.org -- _ -- 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] billsec is set to duration if call is not answered
The behaviour of my Asterisk appears to have changed suddenly without any apparent cause. The version is use is 1.4.27.1 When a call is not answered billsec is set to duration, and calls are charged. I can't see any change I could have made to cause this problem. Is it something already known in Asterisk? /voipfc -- _ -- 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] Can an agent Login to a queue and be paused
YEas actually a Web Gui is the best idea. My dialplan gives a backup
solution for login
2010/2/8 Robert Grignon rgrig...@fleetone.com
Not a bad idea... We use queuemetrics and the login is done via Web GUI.
I could easily just send it to pause upon login...
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mariano Lecuona
*Sent:* Monday, February 08, 2010 8:20 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Can an agent Login to a queue and be
paused
What Id did was on the dialplan, create an specifica extension for login
agents. Lets say Agent/10017, then
When dial 2110017 the agents is promts for Agent passwd.Then I have a macro
only for pausing agents depending on the meaning.
So if the agent is successfully granted on the Login Context, that same
context goto pause macro.
Quick example:
[queues_logon]
; Agent Login Procedure
exten = _211,1,Answer()
exten = _211,n,NoCDR()
exten = _211,n,GotoIf($[${LEN(${AGENTBYCALLERID_${CALLERID(number)}})}
1 ]?4:5) ; Check that the physical extension is free
exten = _211,n,AgentCallbackLogin(${EXTEN:2},,${CALLERID(number)}) ;
Ask for agent password and log the agent on
exten = _211,n,Macro(agent_pause_reason,${EXTEN:2},30) ; Put Agents
into Initial Paused State on the Queue
exten = _211,n,Hangup()
[macro-agent_pause]
; ${ARG1} - Agent_nro
exten = s,1,PauseQueueMember(|Agent/${ARG1})
exten = s,n,MacroExit
2010/2/8 Lenz Emilitri lenz.lo...@gmail.com
I'm not sure if this works for newer versions of Asterisk, but on old
ones, you could pause an agent and THEN log him on, and he'd be paused.
l.
2010/2/4 Robert Grignon rgrig...@fleetone.com
I thought there was an option for this but cant find it
We have a busy callcenter and I would like the agents to log in and be
in a paused state upon login... Right now they login and they are
instantly receiving a call
Thanks for the input...
--
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[asterisk-users] IVR Demo / Create file / Move file / Demo all
Do you see any syntax errors?
Positive comments welcomed.
The short version of the logic is as follows:
create a file based on the NUMBER
create a an audio file
move to a new extension (label) and play the results
exten = 621,1,Answer()
exten = 621,n,Read(NUMBER,enteryournumberstartingwithaone,12,,5)
; create a variable from a DTMF entry / 12 characters long
exten = 621,n,System{(/tmp touch($NUMBER)}
; create the file based on the variable entered
exten = 621,n,Set(audioscript=$[${NUMBER} + 1])
; set a channel variable in advance of recording to it
exten = 621,n,SayDigits(${NUMBER})
; say the NUMBER that was entered
exten = 621,n,System{/tmp touch($(audioscript)}
; create a file
exten = 621,n,Record(${audioscript})
; record a file based on the NUMBER + 1
exten = 621,n,Playback(audioscript)
; listen to the recording - it changes for each
Demo
exten = 621,n,System(mv audioscript /var/lib/asterisk/sounds/en)
; move the recording to the sounds directory
exten = 621,n,Goto(${audioscript},1)
; goto the label/alias to hear it all together
exten = audioscript,1,Answer()
; Nothing
exten = audioscript,n,Playback(audioscript)
; plays audioscript
exten = audioscript,n,Playback(staticIVR_sample)
; adds some boring IVR lingo
exten = audioscript,n,Hangup()
; drops the call
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Re: [asterisk-users] IVR Demo / Create file / Move file / Demo all
On Mon, 8 Feb 2010, Thomas Perron wrote: Do you see any syntax errors? Yes. Lots. Can I please have the last 5 minutes of my life back? Positive comments welcomed. Please don't bother the list to syntax check your code if you are too lazy to type it into your dialplan and see if Asterisk likes it. -- 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] IVR Demo / Create file / Move file / Demo all
Steve, if he had that kind of power, he wouldn't have made the OP. BTW, I doubt it took you 5 min to actually figure out that the syntax wasn't correct. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, February 08, 2010 11:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR Demo / Create file / Move file / Demo all On Mon, 8 Feb 2010, Thomas Perron wrote: Do you see any syntax errors? Yes. Lots. Can I please have the last 5 minutes of my life back? Positive comments welcomed. Please don't bother the list to syntax check your code if you are too lazy to type it into your dialplan and see if Asterisk likes it. -- 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] IVR Demo / Create file / Move file / Demo all
what is OP please? can you just simply comment on the technical work please? On Mon, Feb 8, 2010 at 12:24 PM, Danny Nicholas da...@debsinc.com wrote: Steve, if he had that kind of power, he wouldn't have made the OP. BTW, I doubt it took you 5 min to actually figure out that the syntax wasn't correct. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, February 08, 2010 11:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR Demo / Create file / Move file / Demo all On Mon, 8 Feb 2010, Thomas Perron wrote: Do you see any syntax errors? Yes. Lots. Can I please have the last 5 minutes of my life back? Positive comments welcomed. Please don't bother the list to syntax check your code if you are too lazy to type it into your dialplan and see if Asterisk likes it. -- 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 -- _ -- 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] conferencing without DAHDI
Hi Philippe! Am 08.02.2010 13:49, schrieb Philippe Sultan: Hi Klaus, The module is app_confbridge, and the application is ConfBridge. I had been using it for a while because it's really easy to use : you don't need any configuration file, and you get cool announcements upon conference events from a playback channel. That sounds good. thanks klaus The options work pretty much like meetme, although I would have liked to have a 'x' option to close the conference when the last marked user leaves. Moreover, I couldn't have the playback channel speak French, from what I've read in the source code, I think that feature would require a configuration file because the playback channel is not a per user option. Philippe On Mon, Feb 8, 2010 at 12:56 PM, Olle E. Johanssono...@edvina.net wrote: 8 feb 2010 kl. 12.29 skrev Klaus Darilion: Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. In Asterisk trunk there's a new conference bridge module you can test. There are also some third-party modules out there, like app_conference. /O -- _ -- 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] IVR Demo / Create file / Move file / Demo all
Thomas, you are the OP - Original Poster - that is technical lingo; as for
the script you submitted, here are a few of the problems;
exten = 621,1,Answer()
exten = 621,n,Read(NUMBER,enteryournumberstartingwithaone,12,,5)
** the above name is very wordy and could be replaced with the standard
enter-phone-number10 (in quotes because of outlook)
; create a variable from a DTMF entry / 12 characters long
exten = 621,n,System{(/tmp touch($NUMBER)}
** you don't need to create the file, Record will do that **
; create the file based on the variable entered
exten = 621,n,Set(audioscript=$[${NUMBER} + 1])
; set a channel variable in advance of recording to it
exten = 621,n,SayDigits(${NUMBER})
; say the NUMBER that was entered
exten = 621,n,System{/tmp touch($(audioscript)}
** See error #2 **
; create a file
exten = 621,n,Record(${audioscript})
** this would have to be
exten = 621,n,Record(${audioscript}.gsm)
**
; record a file based on the NUMBER + 1
exten = 621,n,Playback(audioscript)
** audioscript would be a fixed name like
/var/lib/asterisk/sounds/audioscript.gsm - correct would be
exten = 621,n,Playback(${audioscript}) **
; listen to the recording - it changes for each Demo
exten = 621,n,System(mv audioscript /var/lib/asterisk/sounds/en)
** same error as above - correct is
exten = 621,n,System(mv ${audioscript}.gsm /var/lib/asterisk/sounds/en)
**
; move the recording to the sounds directory
exten = 621,n,Goto(${audioscript},1)
** the below should be googled for a correct solution - perhaps
voip-info.org ?? **
; goto the label/alias to hear it all together
exten = audioscript,1,Answer()
; Nothing
exten = audioscript,n,Playback(audioscript)
; plays audioscript
exten = audioscript,n,Playback(staticIVR_sample)
; adds some boring IVR lingo
exten = audioscript,n,Hangup()
; drops the call
--
Danny Nicholas
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
Sent: Monday, February 08, 2010 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IVR Demo / Create file / Move file / Demo all
Do you see any syntax errors?
Positive comments welcomed.
The short version of the logic is as follows:
create a file based on the NUMBER
create a an audio file
move to a new extension (label) and play the results
exten = 621,1,Answer()
exten = 621,n,Read(NUMBER,enteryournumberstartingwithaone,12,,5)
; create a variable from a DTMF entry / 12 characters long
exten = 621,n,System{(/tmp touch($NUMBER)}
; create the file based on the variable entered
exten = 621,n,Set(audioscript=$[${NUMBER} + 1])
; set a channel variable in advance of recording to it
exten = 621,n,SayDigits(${NUMBER})
; say the NUMBER that was entered
exten = 621,n,System{/tmp touch($(audioscript)}
; create a file
exten = 621,n,Record(${audioscript})
; record a file based on the NUMBER + 1
exten = 621,n,Playback(audioscript)
; listen to the recording - it changes for each
Demo
exten = 621,n,System(mv audioscript /var/lib/asterisk/sounds/en)
; move the recording to the sounds directory
exten = 621,n,Goto(${audioscript},1)
; goto the label/alias to hear it all together
exten = audioscript,1,Answer()
; Nothing
exten = audioscript,n,Playback(audioscript)
; plays audioscript
exten = audioscript,n,Playback(staticIVR_sample)
; adds some boring IVR lingo
exten = audioscript,n,Hangup()
; drops the call
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Re: [asterisk-users] GSM Gateway
Another vote for the Voiceblues. Rock solid equipment. Peter den Hartog wrote: http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html We use this one, and it works great.. easy to setup and it works with a normal network connection :) On Mon, Feb 8, 2010 at 1:52 PM, Peter peterp...@aboutsupport.com mailto:peterp...@aboutsupport.com wrote: Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign relies on USB and project GSMOPEN. what would you recommend - trunk or usb ? Or there are other possibilities ? Thanks, Peter -- _ -- 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 -- Groet // Kind regards, Peter den Hartog -- _ -- 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] queue with strategy=linear
Be sure to set a timeout on your Queue() command in the dialplan. exten = 100,1,Answer() exten = 100,n,Queue(test1800) exten = 100,n,Voicemail(100,u) exten = 100,n,Hangup() Thanks, --Warren Selby On Feb 8, 2010, at 3:27 AM, Louis-David Mitterrand vindex+lists-asterisk-us...@apartia.org wrote: Hi, Using asterisk 1.6.2.0 I have a queue definition with strategy=linear. How do I jump to the next dialplan item after having tried (unsuccessfully) all queue members? If I use Queue(test,n) then only the first member is contacted. And if I omit the n option then all members are retried indefinitely. 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 -- _ -- 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] conferencing without DAHDI
Am 08.02.2010 13:49, schrieb Philippe Sultan: Hi Klaus, The module is app_confbridge, and the application is ConfBridge. I had been using it for a while because it's really easy to use : you don't need any configuration file, and you get cool announcements upon conference events from a playback channel. Philippe, what exactly is a playback channel? Is it a pseudo participant playing back the announcements? thanks klaus Further, is there somewhere a documentation The options work pretty much like meetme, although I would have liked to have a 'x' option to close the conference when the last marked user leaves. Moreover, I couldn't have the playback channel speak French, from what I've read in the source code, I think that feature would require a configuration file because the playback channel is not a per user option. Philippe On Mon, Feb 8, 2010 at 12:56 PM, Olle E. Johanssono...@edvina.net wrote: 8 feb 2010 kl. 12.29 skrev Klaus Darilion: Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. In Asterisk trunk there's a new conference bridge module you can test. There are also some third-party modules out there, like app_conference. /O -- _ -- 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] IVR Demo / Create file / Move file / Demo all
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
Do you see any syntax errors?
[snip]
On Mon, 8 Feb 2010, Danny Nicholas wrote:
exten = 621,n,Read(NUMBER,enteryournumberstartingwithaone,12,,5)
; create a variable from a DTMF entry / 12 characters long
** the above name is very wordy and could be replaced with the standard
enter-phone-number10 (in quotes because of outlook)
I would suggest creating your prompts outside of Asterisk's directory
for organization and to avoid potential name conflicts. I store my
prompts in a subdirectory of .../sounds/ based on project or client so
I can reference them easily.
exten = 621,n,System{(/tmp touch($NUMBER)}
** you don't need to create the file, Record will do that **
; create the file based on the variable entered
You missed a bunch here:
1) Application parameters belong in parentheses, not curly-braces.
2) Variables are de-referenced by curly-braces, not parentheses.
3) The $ identifying a variable reference belongs outside of the
curly-braces, not inside.
4) The parameter to the system() application is a command line. Maybe you
meant touch /tmp/${NUMBER}.
Positive comments welcomed.
And most importantly, do you (Thomas) really value the time of members of
this list so little that you waste it with issues you obviously have put
close to zero effort into?
On Sun, 7 Feb 2010, Thomas Perron wrote:
works like magic. thank you. I love this list. when you get stumped
you can always (almost!) count on some great input!
I guess we have different definitions of stumped.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] conferencing without DAHDI
Philippe, what exactly is a playback channel? Is it a pseudo participant playing back the announcements? Yes. Announcements are played to the conference members by creating a channel of type 'Bridge' which streams the sound files. thanks klaus Further, is there somewhere a documentation Well, there is no sample configuration in the tarball because ConfBridge does require any configuration file. 'core show application ConfBridge' in the CLI will give you the options list. You'd probably also want to take a look at the app_confbridge.c file. Very short and readable for such a powerful app. Philippe -- _ -- 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] moving a bridged call to a conference room
I'm just figuring out conferencing. I have a super-simple setup with one room: exten = 600,1,Answer exten = 600,2,ConfBridge(1234,c|M|s) exten = 600,3,hangup If two people want to take their (bridged) call to the conference room, the local user has to do a transfer (to 600), and then dial 600 themselves. Is there an easier way to transfer both ends of a bridged call to the conference room? - Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 Asterisk Boxes, Single Voicemail
I tried NFS, but must be doing something wrong, as lag times between the two are unacceptably high -- as high as 10 to 15 seconds. If you have any hints about this problem, please let me know. Meantime, I'll pursue the rsync angle. Thanks, G -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Friday, February 05, 2010 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 2 Asterisk Boxes, Single Voicemail On 5 Feb 2010, at 16:55, Greg Blakely wrote: If so, how? NFS or rsync? S -- _ -- 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] Strange Problem
Hello list! I've run into a strange problem today and I was hoping that someone here has seen this before and maybe can give me a hand: I'm using asterisk 1.6.0.22 in this config: (A)PATTON ISDN -(B) ASTERISK - (C)PATTON PRI - PSTN - (D)OTHER PBX Strange Problem: USER A calls makes a call to a PBX over the PSTN and ends into an IVR. When the user makes a selection and gets his call passed to an extension of that PBX (USER D), USER D has no sound while USER A hears the voice just fine. If USER A makes a direct call to USER D, calling directly his extension, the call has audio both ways and its all working fine. The same thing if USER A calls directly mobile phones or numbers that aren't managed by IVRs. I've verified this with a few PBXs(different manufacturers), and the problem is there every time an IVR gets the control of the call. A sip debug in asterisk confirmed that the SIP Session is not renegotiated when the call exits USER's D IVR and ends up to his extension. Any idea what might be causing this? Thank you in advance! Alex -- _ -- 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] IVR Demo / Create file / Move file / Demo all
On Mon, Feb 08, 2010 at 12:36:18PM -0500, Thomas Perron wrote: what is OP please? can you just simply comment on the technical work please? Original Poster. The one who started the thread. In this case it's you. -- 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] Strange Problem
Monitor the successful and failing calls from a CLI session with core set verbose 5. This should show you what is different between the two calls. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alexandru Oniciuc Sent: Monday, February 08, 2010 3:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Strange Problem Hello list! I've run into a strange problem today and I was hoping that someone here has seen this before and maybe can give me a hand: I'm using asterisk 1.6.0.22 in this config: (A)PATTON ISDN -(B) ASTERISK - (C)PATTON PRI - PSTN - (D)OTHER PBX Strange Problem: USER A calls makes a call to a PBX over the PSTN and ends into an IVR. When the user makes a selection and gets his call passed to an extension of that PBX (USER D), USER D has no sound while USER A hears the voice just fine. If USER A makes a direct call to USER D, calling directly his extension, the call has audio both ways and its all working fine. The same thing if USER A calls directly mobile phones or numbers that aren't managed by IVRs. I've verified this with a few PBXs(different manufacturers), and the problem is there every time an IVR gets the control of the call. A sip debug in asterisk confirmed that the SIP Session is not renegotiated when the call exits USER's D IVR and ends up to his extension. Any idea what might be causing this? Thank you in advance! Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk how install speex support
check the output of running configure for any mentions of problems with libspeex On Mon, Feb 8, 2010 at 8:09 AM, nedo nodo nedo.n...@gmail.com wrote: Hi, I would like to add support for speex codec in Asterisk. In Ubuntu 9.10 the procedure is the following: 1) sudo apt-get install speex libspeex-dev 2) install Asterisk that enable speex support in configure procedure 3) core show translation I can't see the translation time. Where is the problem? Thank -- BOINC manager http://boincmanager.altervista.org -- _ -- 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] Losing local SIP phones when internet goes down?
On 6/02/10 4:06 AM, Dave Cotton wrote: On 05/02/10 16:01, Jeff LaCoursiere wrote: On Fri, 5 Feb 2010, Vinícius Fontes wrote: I solved similar issues by setting srvlookup=no, having bind running locally and just the line nameserver 127.0.0.1 on /etc/resolv.conf. Your local bind is what solved the problem. The srvlookup=no didn't actually help IMO. Given the choice between configuring bind and dnsmasq I know which I'd go for. They're both pretty easy - bind9 easier I reckon. To set up on debian do: apt-get install bind9 add to the top of /etc/resolv.conf nameserver 127.0.0.1 Then it's done. Dnsmasq is probably overkill for this type of thing, though some people in the office prefer it to bind. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] Losing local SIP phones when inte rnet goes down?
On Monday 08 February 2010 17:11:34 Matt Riddell wrote: On 6/02/10 4:06 AM, Dave Cotton wrote: On 05/02/10 16:01, Jeff LaCoursiere wrote: On Fri, 5 Feb 2010, Vinícius Fontes wrote: I solved similar issues by setting srvlookup=no, having bind running locally and just the line nameserver 127.0.0.1 on /etc/resolv.conf. Your local bind is what solved the problem. The srvlookup=no didn't actually help IMO. Given the choice between configuring bind and dnsmasq I know which I'd go for. They're both pretty easy - bind9 easier I reckon. To set up on debian do: apt-get install bind9 add to the top of /etc/resolv.conf nameserver 127.0.0.1 If you're using DHCP on any of your interfaces, you'll need to configure dhclient (or whatever dhcp client you're using) to prepend in the configuration with (e.g. /etc/dhcp3/dhclient.conf): prepend domain-name-servers 127.0.0.1; Otherwise, your entry in resolv.conf will be overwritten on each DHCP lease renewal. -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime
I may be late to this thread, but my own restarted every 3-5 days until I upgraded to 1.4.29 (I skipped 1.4.28). It`s been running for 8 days now, which isn't long enough for me to declare whatever-it-is fixed, but enough to say it's at least better with 1.4.29 stability wise. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of William Stillwell (Lists) Sent: Monday, February 08, 2010 9:13 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime After reviewing other emails, you also may want to enable debug logging, and find last log entry before crash.. Also graph cpu load, memory usage, call count.. I had one server that would reboot every few days, turned out the PCI-e bus was not playing nicely with the PRI Card, after switching servers, the crashing went away. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of William Stillwell (Lists) Sent: Monday, February 08, 2010 8:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime Box #1 faxserver*CLI core show version Asterisk 1.4.21.2 built by root @ faxserver.localhost on a i686 running Linux on 2008-08-07 20:30:54 UTC faxserver*CLI core show uptime System uptime: 21 weeks, 2 days, 22 hours, 43 minutes, 42 seconds faxserver*CLI this box gets about 200 faxes a day, and does a tone of agi script processing, and network printing. Someday I may upgrade it, but it runs too well for me to want to touch it. Box #2 sip*CLI core show version Asterisk 1.4.26.2 built by root @ ast-two.localhost on a i686 running Linux on 2009-09-05 00:17:05 UTC sip*CLI core show uptime System uptime: 3 weeks, 3 days, 15 hours, 17 minutes, 34 seconds sip*CLI this is my IVR outbound LD box.. Personnel Box for home: localhost*CLI core show version Asterisk 1.4.28 built by root @ localhost.localdomain on a i686 running Linux on 2009-12-20 04:16:08 UTC localhost*CLI core show uptime System uptime: 3 weeks, 1 day, 15 hours, 38 minutes, 5 seconds Last reload: 3 weeks, 1 day, 6 hours, 19 minutes, 54 seconds localhost*CLI Doesn't get many calls at all.. it's just for my house, maybe 10 calls a week.. , and I do a lot of custom network IVR stuff with it.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Per Jessen Sent: Sunday, February 07, 2010 9:09 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime Thomas Winter wrote: Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? My asterisk 1.4.23 also dies about once a month. I've never been able to work out why. /Per Jessen, Z�rich -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime
I have a box with four PRIs running Asterisk 1.2.something with approx two years of uptime, not so much as a reload. It is an IBM x305. I am not sure why people want the latest and Greatest unless there is some killer app you want. Thanks, Steve T On Mon, Feb 8, 2010 at 8:00 PM, Mike l...@virtutel.ca wrote: I may be late to this thread, but my own restarted every 3-5 days until I upgraded to 1.4.29 (I skipped 1.4.28). It`s been running for 8 days now, which isn't long enough for me to declare whatever-it-is fixed, but enough to say it's at least better with 1.4.29 stability wise. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of William Stillwell (Lists) Sent: Monday, February 08, 2010 9:13 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime After reviewing other emails, you also may want to enable debug logging, and find last log entry before crash.. Also graph cpu load, memory usage, call count.. I had one server that would reboot every few days, turned out the PCI-e bus was not playing nicely with the PRI Card, after switching servers, the crashing went away. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of William Stillwell (Lists) Sent: Monday, February 08, 2010 8:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime Box #1 faxserver*CLI core show version Asterisk 1.4.21.2 built by root @ faxserver.localhost on a i686 running Linux on 2008-08-07 20:30:54 UTC faxserver*CLI core show uptime System uptime: 21 weeks, 2 days, 22 hours, 43 minutes, 42 seconds faxserver*CLI this box gets about 200 faxes a day, and does a tone of agi script processing, and network printing. Someday I may upgrade it, but it runs too well for me to want to touch it. Box #2 sip*CLI core show version Asterisk 1.4.26.2 built by root @ ast-two.localhost on a i686 running Linux on 2009-09-05 00:17:05 UTC sip*CLI core show uptime System uptime: 3 weeks, 3 days, 15 hours, 17 minutes, 34 seconds sip*CLI this is my IVR outbound LD box.. Personnel Box for home: localhost*CLI core show version Asterisk 1.4.28 built by root @ localhost.localdomain on a i686 running Linux on 2009-12-20 04:16:08 UTC localhost*CLI core show uptime System uptime: 3 weeks, 1 day, 15 hours, 38 minutes, 5 seconds Last reload: 3 weeks, 1 day, 6 hours, 19 minutes, 54 seconds localhost*CLI Doesn't get many calls at all.. it's just for my house, maybe 10 calls a week.. , and I do a lot of custom network IVR stuff with it.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Per Jessen Sent: Sunday, February 07, 2010 9:09 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime Thomas Winter wrote: Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? My asterisk 1.4.23 also dies about once a month. I've never been able to work out why. /Per Jessen, Z�rich -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime
Steve Totaro wrote: I am not sure why people want the latest and Greatest unless there is some killer app you want. It is an illness, to be sure. I have read of so many issues with various 1.4 versions, and 1.2 was working for me as well, that I left well enough alone. I finally migrated, when I built a new box, to CentOS 5 and 1.4.28 An IBM x330 Only been up since October though. Simple flow chart. Does it work? don't f**k with it John Novack -- Dog is my co-pilot -- _ -- 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] IVR Demo / Create file / Move file / Demo all
this solution works. thanks for the helpful comments.
exten = 621,n,Read(NUMBER,snowday,12,,10)
; create a variable from a DTMF entry / 12 characters long
;exten = 621,n,System{(/tmp touch($NUMBER)}
; create the file based on the variable entered
exten = 621,n,Set(audioscript=$[${NUMBER} + 1])
; set a channel variable in advance of recording to it
exten = 621,n,SayDigits(${NUMBER})
; say the NUMBER that was entered
exten = 621,n,SendDTMF(${NUMBER})
;exten = 621,n,System{/tmp touch($(audioscript)}
; create a file
exten = 621,n,Record(${audioscript}.gsm)
; record a file based on the NUMBER + 1
exten = 621,n,Playback(${audioscript})
; listen to the recording , etc.
exten = 621,n,System(mv ${audioscript}.gsm
/var/lib/asterisk/sounds/en) ; move the recording to the
sounds directory
exten = 621,n,Playback(dir-welcome)
exten = 621,n,Playback(${audioscript})
exten = 621,n,Playback(snowday2)
exten = 621,n,Goto(s,1)
On Mon, Feb 8, 2010 at 2:00 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Mon, Feb 08, 2010 at 12:36:18PM -0500, Thomas Perron wrote:
what is OP please?
can you just simply comment on the technical work please?
Original Poster. The one who started the thread. In this case it's you.
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Re: [asterisk-users] E71
hi, I'm been successful in making calls to another local extension using Nokia E71. However calling the E71 from another ext. (X-lite) is not successful. There is a ringing tone from the caller side but the E71 is silent. Tried disabling the NAT (dunno whether that helps). Instructions where from http://www.geek.com/articles/mobile/feature-voip-with-nokia-e71-how-to-2008095/ Am i missing something? -- _ -- 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] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
Hi Team Can someone advice me on how i can lower the load average on my asterisk server? dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10.1 asterisk-1.4.25.1 2 X TE412P Digium cards on ISDN PRI Im using the system as an IVR without any transcoding or bridging ** top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75 Tasks: 149 total, 1 running, 148 sleeping, 0 stopped, 0 zombie Cpu0 : 10.3%us, 32.0%sy, 0.0%ni, 57.3%id, 0.0%wa, 0.0%hi, 0.3%si, 0.0%st Cpu1 : 10.6%us, 34.6%sy, 0.0%ni, 54.8%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu2 : 13.3%us, 36.5%sy, 0.0%ni, 49.8%id, 0.0%wa, 0.0%hi, 0.3%si, 0.0%st Cpu3 : 8.6%us, 39.5%sy, 0.0%ni, 51.8%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu4 : 7.3%us, 38.0%sy, 0.0%ni, 54.7%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu5 : 17.9%us, 37.5%sy, 0.0%ni, 44.5%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu6 : 13.3%us, 37.2%sy, 0.0%ni, 49.5%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu7 : 12.7%us, 37.3%sy, 0.0%ni, 50.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 3961100k total, 3837920k used, 123180k free, 108944k buffers Swap: 779144k total, 56k used, 779088k free, 3602540k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 683 root 15 0 97968 36m 5616 S 307.7 0.9 41457:34 asterisk 17176 root 15 0 2196 1052 800 R 0.7 0.0 0:00.32 top 1 root 15 0 2064 592 512 S 0.0 0.0 0:13.96 init 2 root RT -5 000 S 0.0 0.0 5:27.80 migration/0 3 root 34 19 000 S 0.0 0.0 0:00.11 ksoftirqd/0 4 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/0 5 root RT -5 000 S 0.0 0.0 1:07.67 migration/1 6 root 34 19 000 S 0.0 0.0 0:00.09 ksoftirqd/1 7 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/1 8 root RT -5 000 S 0.0 0.0 1:16.92 migration/2 9 root 34 19 000 S 0.0 0.0 0:00.03 ksoftirqd/2 10 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/2 11 root RT -5 000 S 0.0 0.0 1:34.54 migration/3 12 root 34 19 000 S 0.0 0.0 0:00.15 ksoftirqd/3 13 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/3 14 root RT -5 000 S 0.0 0.0 0:54.66 migration/4 15 root 34 19 000 S 0.0 0.0 0:00.01 ksoftirqd/4 16 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/4 17 root RT -5 000 S 0.0 0.0 1:39.64 migration/5 18 root 39 19 000 S 0.0 0.0 0:00.21 ksoftirqd/5 19 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/5 20 root RT -5 000 S 0.0 0.0 1:06.27 migration/6 21 root 34 19 000 S 0.0 0.0 0:00.03 ksoftirqd/6 22 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/6 23 root RT -5 000 S 0.0 0.0 1:23.24 migration/7 24 root 34 19 000 S 0.0 0.0 0:00.17 ksoftirqd/7 25 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/7 26 root 10 -5 000 S 0.0 0.0 0:25.70 events/0 27 root 10 -5 000 S 0.0 0.0 0:37.83 events/1 28 root 10 -5 000 S 0.0 0.0 0:15.67 events/2 29 root 10 -5 000 S 0.0 0.0 0:40.36 events/3 30 root 10 -5 000 S 0.0 0.0 0:16.45 events/4 * Thanks Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
Do you want the advice in ALL CAPS? On 02/08/2010 11:42 PM, Muro, Sam wrote: Hi Team Can someone advice me on how i can lower the load average on my asterisk server? dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10.1 asterisk-1.4.25.1 2 X TE412P Digium cards on ISDN PRI Im using the system as an IVR without any transcoding or bridging ** top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75 Tasks: 149 total, 1 running, 148 sleeping, 0 stopped, 0 zombie Cpu0 : 10.3%us, 32.0%sy, 0.0%ni, 57.3%id, 0.0%wa, 0.0%hi, 0.3%si, 0.0%st Cpu1 : 10.6%us, 34.6%sy, 0.0%ni, 54.8%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu2 : 13.3%us, 36.5%sy, 0.0%ni, 49.8%id, 0.0%wa, 0.0%hi, 0.3%si, 0.0%st Cpu3 : 8.6%us, 39.5%sy, 0.0%ni, 51.8%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu4 : 7.3%us, 38.0%sy, 0.0%ni, 54.7%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu5 : 17.9%us, 37.5%sy, 0.0%ni, 44.5%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu6 : 13.3%us, 37.2%sy, 0.0%ni, 49.5%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu7 : 12.7%us, 37.3%sy, 0.0%ni, 50.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 3961100k total, 3837920k used, 123180k free, 108944k buffers Swap: 779144k total, 56k used, 779088k free, 3602540k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 683 root 15 0 97968 36m 5616 S 307.7 0.9 41457:34 asterisk 17176 root 15 0 2196 1052 800 R 0.7 0.0 0:00.32 top 1 root 15 0 2064 592 512 S 0.0 0.0 0:13.96 init 2 root RT -5 000 S 0.0 0.0 5:27.80 migration/0 3 root 34 19 000 S 0.0 0.0 0:00.11 ksoftirqd/0 4 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/0 5 root RT -5 000 S 0.0 0.0 1:07.67 migration/1 6 root 34 19 000 S 0.0 0.0 0:00.09 ksoftirqd/1 7 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/1 8 root RT -5 000 S 0.0 0.0 1:16.92 migration/2 9 root 34 19 000 S 0.0 0.0 0:00.03 ksoftirqd/2 10 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/2 11 root RT -5 000 S 0.0 0.0 1:34.54 migration/3 12 root 34 19 000 S 0.0 0.0 0:00.15 ksoftirqd/3 13 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/3 14 root RT -5 000 S 0.0 0.0 0:54.66 migration/4 15 root 34 19 000 S 0.0 0.0 0:00.01 ksoftirqd/4 16 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/4 17 root RT -5 000 S 0.0 0.0 1:39.64 migration/5 18 root 39 19 000 S 0.0 0.0 0:00.21 ksoftirqd/5 19 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/5 20 root RT -5 000 S 0.0 0.0 1:06.27 migration/6 21 root 34 19 000 S 0.0 0.0 0:00.03 ksoftirqd/6 22 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/6 23 root RT -5 000 S 0.0 0.0 1:23.24 migration/7 24 root 34 19 000 S 0.0 0.0 0:00.17 ksoftirqd/7 25 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/7 26 root 10 -5 000 S 0.0 0.0 0:25.70 events/0 27 root 10 -5 000 S 0.0 0.0 0:37.83 events/1 28 root 10 -5 000 S 0.0 0.0 0:15.67 events/2 29 root 10 -5 000 S 0.0 0.0 0:40.36 events/3 30 root 10 -5 000 S 0.0 0.0 0:16.45 events/4 * Thanks Sam -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.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] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
On Mon, Feb 8, 2010 at 11:42 PM, Muro, Sam resea...@businesstz.com wrote: Hi Team Can someone advice me on how i can lower the load average on my asterisk server? dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10.1 asterisk-1.4.25.1 2 X TE412P Digium cards on ISDN PRI Im using the system as an IVR without any transcoding or bridging ** top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75 Tasks: 149 total, 1 running, 148 sleeping, 0 stopped, 0 zombie Cpu0 : 10.3%us, 32.0%sy, 0.0%ni, 57.3%id, 0.0%wa, 0.0%hi, 0.3%si, 0.0%st Cpu1 : 10.6%us, 34.6%sy, 0.0%ni, 54.8%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu2 : 13.3%us, 36.5%sy, 0.0%ni, 49.8%id, 0.0%wa, 0.0%hi, 0.3%si, 0.0%st Cpu3 : 8.6%us, 39.5%sy, 0.0%ni, 51.8%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu4 : 7.3%us, 38.0%sy, 0.0%ni, 54.7%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu5 : 17.9%us, 37.5%sy, 0.0%ni, 44.5%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu6 : 13.3%us, 37.2%sy, 0.0%ni, 49.5%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu7 : 12.7%us, 37.3%sy, 0.0%ni, 50.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 3961100k total, 3837920k used, 123180k free, 108944k buffers Swap: 779144k total, 56k used, 779088k free, 3602540k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 683 root 15 0 97968 36m 5616 S 307.7 0.9 41457:34 asterisk 17176 root 15 0 2196 1052 800 R 0.7 0.0 0:00.32 top 1 root 15 0 2064 592 512 S 0.0 0.0 0:13.96 init 2 root RT -5 0 0 0 S 0.0 0.0 5:27.80 migration/0 3 root 34 19 0 0 0 S 0.0 0.0 0:00.11 ksoftirqd/0 4 root RT -5 0 0 0 S 0.0 0.0 0:00.00 watchdog/0 5 root RT -5 0 0 0 S 0.0 0.0 1:07.67 migration/1 6 root 34 19 0 0 0 S 0.0 0.0 0:00.09 ksoftirqd/1 7 root RT -5 0 0 0 S 0.0 0.0 0:00.00 watchdog/1 8 root RT -5 0 0 0 S 0.0 0.0 1:16.92 migration/2 9 root 34 19 0 0 0 S 0.0 0.0 0:00.03 ksoftirqd/2 10 root RT -5 0 0 0 S 0.0 0.0 0:00.00 watchdog/2 11 root RT -5 0 0 0 S 0.0 0.0 1:34.54 migration/3 12 root 34 19 0 0 0 S 0.0 0.0 0:00.15 ksoftirqd/3 13 root RT -5 0 0 0 S 0.0 0.0 0:00.00 watchdog/3 14 root RT -5 0 0 0 S 0.0 0.0 0:54.66 migration/4 15 root 34 19 0 0 0 S 0.0 0.0 0:00.01 ksoftirqd/4 16 root RT -5 0 0 0 S 0.0 0.0 0:00.00 watchdog/4 17 root RT -5 0 0 0 S 0.0 0.0 1:39.64 migration/5 18 root 39 19 0 0 0 S 0.0 0.0 0:00.21 ksoftirqd/5 19 root RT -5 0 0 0 S 0.0 0.0 0:00.00 watchdog/5 20 root RT -5 0 0 0 S 0.0 0.0 1:06.27 migration/6 21 root 34 19 0 0 0 S 0.0 0.0 0:00.03 ksoftirqd/6 22 root RT -5 0 0 0 S 0.0 0.0 0:00.00 watchdog/6 23 root RT -5 0 0 0 S 0.0 0.0 1:23.24 migration/7 24 root 34 19 0 0 0 S 0.0 0.0 0:00.17 ksoftirqd/7 25 root RT -5 0 0 0 S 0.0 0.0 0:00.00 watchdog/7 26 root 10 -5 0 0 0 S 0.0 0.0 0:25.70 events/0 27 root 10 -5 0 0 0 S 0.0 0.0 0:37.83 events/1 28 root 10 -5 0 0 0 S 0.0 0.0 0:15.67 events/2 29 root 10 -5 0 0 0 S 0.0 0.0 0:40.36 events/3 30 root 10 -5 0 0 0 S 0.0 0.0 0:16.45 events/4 * Thanks Sam Even though you shouldn't have to, have your rebooted? 200 days of uptime and this just started? Have you recently updated the box? ksoftirqd seems to have issues in some kernels. That is where I would start after restarting Asterisk and or the server. http://tinyurl.com/ygd2eha Thanks, Steve T -- _ -- 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] GSM Gateway
Hi, I can not find pricing and shipping information for these. I tried to contact their sales for these. We will see, but most likely we will go with portech. Peter On 08.2.2010 15:15, Peter den Hartog wrote: http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html We use this one, and it works great.. easy to setup and it works with a normal network connection :) On Mon, Feb 8, 2010 at 1:52 PM, Peter peterp...@aboutsupport.com mailto:peterp...@aboutsupport.com wrote: Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign relies on USB and project GSMOPEN. what would you recommend - trunk or usb ? Or there are other possibilities ? Thanks, Peter -- _ -- 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 -- Groet // Kind regards, Peter den Hartog -- _ -- 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
