Re: [asterisk-users] Phone Inventory
Thank you all
You are a life saver
Sam
Dale Noll wrote:
On 02/23/2012 08:49 AM, Danny Nicholas wrote:
Here is a snippet that somebody smarter than I am can improve upon
for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx
sip
show peer $a;done|grep Useragent
for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx
sip
show peer $a;done|grep Contact
Thanks for the inspiration!!
Here is my version, done with a single loop and gets Useragent and
Contact together with a visual separation between peers.
asterisk -rx sip show peers|
cut -f1 -d/ | grep -P '\d\d\d\d' |
grep -vP '(UNKNOWN|Unmonitored)' |
while read PEER
do
asterisk -rx sip show peer ${PEER} |
grep -P (Useragent|Contact)
echo
done
I hope others find it useful.
Dale
PS. I by no means claim to be smarter than thou. I just happen to
really like grep and the -P option ;-)
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[asterisk-users] Replicating SIP registration Info between active to standby
I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a time. How can I copy sip registration information from Active Server to Standby Server Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replicating SIP registration Info between active to standby
Hi Takehiro Are you suggesting sharing the AstDB ? Sam Takehiro Matsushima wrote: Hi. How about place backend DB on shared disk, or make replication between them? 2012/02/24 13:58 Muro, Sam resea...@businesstz.com: I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a time. How can I copy sip registration information from Active Server to Standby Server Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phone Inventory
Hi there I have just took a support of a customer with hundreds of IP phones, mostly Polycom with mixed models. Is there a way to query asterisk or any other command to retrieve the inventory of all connected phones. i.e. Phone Type and Phone Model, say Polycom SPIP331 or so Thanks Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Security: Allow only one phone per sip registration
Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those accounts. 200 being one very sensitive individual. Lets say, an insider, get a new phone or perhaps an xlite and configure it with the same extension, 200. Asterisk will register it as 200 to the new IP address. Now extension 202 call 200. The hacker answers it and pretend is the same person. Do what he want to do and thats it. Question; How can i stop this type of threat Regads Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration
Terry Wilson wrote: - Original Message - From: Sam Muro resea...@businesstz.com To: asterisk-users@lists.digium.com Sent: Friday, October 14, 2011 2:02:01 AM Subject: [asterisk-users] Asterisk Security: Allow only one phone per sip registration Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those accounts. 200 being one very sensitive individual. Lets say, an insider, get a new phone or perhaps an xlite and configure it with the same extension, 200. Asterisk will register it as 200 to the new IP address. Now extension 202 call 200. The hacker answers it and pretend is the same person. Do what he want to do and thats it. Question; How can i stop this type of threat I would recommend actually setting a different secret field in sip.conf for each device so that your would-be attacker isn't able to register as someone else. Is there a way one can bind sip account to specific mac-address (assume on the same subnet). In this way, even if you know the username/secret, you will still have to use the same physical phone, unless you play with mac-address. Or you could buy a gun. I bet the insider would be very afraid of the gun and would therefore avoid any shenanigans while you were around. This would especially be true if you randomly shot items like coffee cups and plants whenever you thought they were looking at you funny. That'll show 'em. Lol! Here they will name you a terrorist -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration
Terry Wilson wrote: Is there a way one can bind sip account to specific mac-address (assume on the same subnet). In this way, even if you know the username/secret, you will still have to use the same physical phone, unless you play with mac-address. No. And mac addresses are easily spoofed so it would not help. Use passwords. Keep them safe. Thanks. Let me see how best i can complicate them per phone. Ooops, 1000 sip phones -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration
Thanks Terry! Let me think of all possibilities and shall holla. Can you be one? Terry Wilson wrote: Thanks. Let me see how best i can complicate them per phone. Ooops, 1000 sip phones If it were me, I would look into Asterisk Realtime for handling the SIP phones. I would then write a script to generate the configs for the phones into the SIP realtime database with random passwords. Match up the phones with the accounts and provision the phones. You would most likely use a provisioning server of some kind to generate the actual phone configurations. You can check out the res_phoneprov module in Asterisk, find another one somewhere, or write your own. Many people tend to write their own for large installations. I did. If you have a big installation like this and are wondering about things like whether mac addresses should be used for security, it might also be a good idea to hire a consultant. Check out the asterisk-biz mailing list. Terry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration
Thanks A.J I know and I can assure you no one will get that physical access to the system. A J Stiles wrote: On Friday 14 October 2011, Muro, Sam wrote: Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those accounts. 200 being one very sensitive individual. Lets say, an insider, get a new phone or perhaps an xlite and configure it with the same extension, 200. Asterisk will register it as 200 to the new IP address. Now extension 202 call 200. The hacker answers it and pretend is the same person. Do what he want to do and thats it. Question; How can i stop this type of threat Be careful who you employ and how you treat them :) Once someone has physical access to your equipment, all bets are off . -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Passing parameter from executable program to asterisk dialplan
I am having a problem understanding the way to retrieve some parameters to asterisk via AGI or what ever method that fits. I have an executable program that accept one parameter (CALLERID) and return customer status from the database server which can be printed in the console. #./retrive 0117473789 NAME: Franklin John STATUS: Active Can someone advice on how i can catch this values from AGI or directly on dialplan. Thanks Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing parameter from executable program to asterisk dialplan
Kyle Kienapfel wrote: On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam resea...@businesstz.com wrote: I am having a problem understanding the way to retrieve some parameters to asterisk via AGI or what ever method that fits. I have an executable program that accept one parameter (CALLERID) and return customer status from the database server which can be printed in the console. #./retrive 0117473789 NAME: Franklin John STATUS: Active Can someone advice on how i can catch this values from AGI or directly on dialplan. Thanks Sam -- Hopefully you can modify the executable #./retrieve 8675309 SET VARIABLE name Jenny SET VARIABLE status Active When running an AGI asterisk expects to have a conversation with the application, so when the AGI does a command asterisk reports back with whether or not it worked. I know a person can set one variable that way, but when I got a need to set two variables I finally broke down and read the documentation on AGI's :) Start Readlines from input until line is blank print SET VARIABLE name Jenny readline print SET VARIABLE status Active End -- Thanks, So I you suggesting that the executable to changed to output say cout SET VARIABLE name Jenny; and let the AGI retrieve them as per the pseudo you mentioned? Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing parameter from executable program to asterisk dialplan
Steve Edwards wrote: On Sun, 25 Jul 2010, Muro, Sam wrote: I am having a problem understanding the way to retrieve some parameters to asterisk via AGI or what ever method that fits. I have an executable program that accept one parameter (CALLERID) and return customer status from the database server which can be printed in the console. #./retrive 0117473789 NAME: Franklin John STATUS: Active Can someone advice on how i can catch this values from AGI or directly on dialplan. AGI is a protocol used to interact with Asterisk. An AGI is a separate process created by Asterisk when you execute agi() in the dialplan. From best to worst... 1) You could recode your retrieve application so it uses the AGI protocol. Then, you could set channel variables to make these values accessible to the rest of your dialplan. 2) You could cobble up an AGI to execute your retrieve application using a pipe (popen() in c), parse the output and set channel variables. 3) You could cobble up something to execute your retrieve application, redirecting the output to a file and then use the FILE function read the text file and then parse the output using dialplan functions. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- Thanks Steve Option one and two looks more ideal. Let stick on option one, the program is written in C++ (Actually is a corba interface). I have tried looking on how to write AGI script using C++ in vain. I am used to perl/php for scripting.. Can you post a snippet of c++ agi script. Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing parameter from executable program to asterisk dialplan
Kyle Kienapfel wrote: On Sun, Jul 25, 2010 at 9:04 AM, Muro, Sam resea...@businesstz.com wrote: Kyle Kienapfel wrote: On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam resea...@businesstz.com wrote: I am having a problem understanding the way to retrieve some parameters to asterisk via AGI or what ever method that fits. I have an executable program that accept one parameter (CALLERID) and return customer status from the database server which can be printed in the console. #./retrive 0117473789 NAME: Franklin John STATUS: Active Can someone advice on how i can catch this values from AGI or directly on dialplan. Thanks Sam -- Hopefully you can modify the executable #./retrieve 8675309 SET VARIABLE name Jenny SET VARIABLE status Active When running an AGI asterisk expects to have a conversation with the application, so when the AGI does a command asterisk reports back with whether or not it worked. I know a person can set one variable that way, but when I got a need to set two variables I finally broke down and read the documentation on AGI's :) Start Readlines from input until line is blank print SET VARIABLE name Jenny readline print SET VARIABLE status Active End -- Thanks, So I you suggesting that the executable to changed to output say cout SET VARIABLE name Jenny; and let the AGI retrieve them as per the pseudo you mentioned? Sam Does doing that output a newline at the end of the line? If it doesn't you might want something more like (i am just guessing syntax here btw) cout SET VARIABLE name Jenny ENDL; or cout SET VARIABLE name Jenny\n; -- You are right. Both of them are correct syntax I will give it a try and revert Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing parameter from executable program to asterisk dialplan
Steve Edwards wrote: On Sun, 25 Jul 2010, Muro, Sam wrote: I am having a problem understanding the way to retrieve some parameters to asterisk via AGI or what ever method that fits. I have an executable program that accept one parameter (CALLERID) and return customer status from the database server which can be printed in the console. #./retrive 0117473789 NAME: Franklin John STATUS: Active Can someone advice on how i can catch this values from AGI or directly on dialplan. Steve Edwards wrote: AGI is a protocol used to interact with Asterisk. An AGI is a separate process created by Asterisk when you execute agi() in the dialplan. From best to worst... 1) You could recode your retrieve application so it uses the AGI protocol. Then, you could set channel variables to make these values accessible to the rest of your dialplan. 2) You could cobble up an AGI to execute your retrieve application using a pipe (popen() in c), parse the output and set channel variables. 3) You could cobble up something to execute your retrieve application, redirecting the output to a file and then use the FILE function read the text file and then parse the output using dialplan functions. On Sun, 25 Jul 2010, Muro, Sam wrote: Option one and two looks more ideal. Let stick on option one, the program is written in C++ (Actually is a corba interface). I have tried looking on how to write AGI script using C++ in vain. I am used to perl/php for scripting.. Can you post a snippet of c++ agi script. I'm a c weenie myself. I coded my own library way too long ago and remember the scars :) If you google for asterisk agi c++ library you'll find links to cagi, quivr, and probably a couple more. I don't know if any are c++ specific. The basic outline is: call a function to read the AGI environment from stdin. (Mine is named agi_read_environment()) get your customer ID number either from the command line or from a channel variable. (argv[] or agi_get_variable(CUSTOMER-ID, customer_id.) retrieve your values from your database. set your channel variables. (agi_set_variable(CUSTOMER-NAME, mysql_row[CUSTOMER_NAME])) -- Thanks Steve Let me check them out and give them a try. I really appreciate your input Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Corba interface
Hi there Has anyone know how to configure asterisk to be able to query Corba interface directly from the dialplan Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Corba interface
Hi there Does anyone know how to configure asterisk to be able to query Corba interface directly from the dialplan Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as avaya definity recordingserver
Moises Silva wrote: On Mon, Mar 22, 2010 at 7:56 PM, Rafael Prado Rocchi pr...@practis.com.brwrote: Hi, it's not that simple. It requires deep modification on asterisk and dahdi sources to work the way you want. Why? I must confess I still don't quite understand what he wants, from what I've read the legacy pbx will place a secondary call via ISDN ( did he mean PRI? ) therefore Asterisk will just Record(), what is it that is not so simple about that? Hi Moses Task: Recording phone calls Here is the scenario; - A legacy system is connected back to back to asterisk pbx with PRI connection and asterisk is connected to the telco via PRI Users(Analog/Digital) Legacy (PRI)-Asterisk---(PRI)---Telco - Telco to users (vise versa) need to be recorded on asterisk - Easily Done - Internal calls (extension to extension) on legacy need to be recording (currently is done via Nice) on asterisk - This's the problem Sam -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as avaya definity recording server
Hi there
Looks like someone hasnt done this!! I have been thinking and find out
that Monitor/Spy and the likes wont help me as the call need to be bridged
with the asterisk core or via channel drivers.
My final shot now is on Record() function. Since the legacy system will
forward the call to the monitoring interfaces when bridged within itself,
it the interface in on Asterisk, then we can capture the pattern and use
exten =
#CALLER_NUMBER#CALLED_NUMBER,1,Record(/var/spool/asterisk/monitor/avaya-${EXTEN:1:4}-${EXTEN:4:4}:wav)
This assume that Len(CALLER_NUMBER) = 4
Anyone with alternative solution?
Muro, Sam wrote:
Oh.. I didnt know that.
Thanks
Sam
Muro, Sam escribió:
What do you mean chief? What am looking at is ability for asterisk to
receive a call and recording until it tier down without bridging it to
the
physical device
Sam
Would you like the advice in all caps?
He means that you put the subject in all caps. He normally gets upset
with everyone that does this on the subject or in the body. I've
corrected the caps in the subject to avoid further upsetting.
Cheers,
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Re: [asterisk-users] asterisk-users Digest, Vol 68, Issue 33
What do you mean chief? What am looking at is ability for asterisk to receive a call and recording until it tier down without bridging it to the physical device Sam Would you like the advice in all caps? On 03/15/2010 01:20 AM, RESEARCH wrote: Hi there I remember to ask this question in the past but now I have thought of something little bit difference. While I understand that asterisk dialplan accept the call to be answered[ Answer() ] in the dialplan, I wanna know if this is possible; i. A call on legacy PBX, extension to extension is made. ii. On call bridging, the legacy PBX initiate a third bridging to the recording system via an ISDN interface. iii. Conversation on Legacy continue but asterisk record this call until hangup is issued Please advice if this is possible. 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USING ASTERISK AS AVAYA DEFINITY RECORDING SERVER
What do you mean chief? What am looking at is ability for asterisk to receive a call and recording until it tier down without bridging it to the physical device Sam Would you like the advice in all caps? On 03/15/2010 01:20 AM, RESEARCH wrote: Hi there I remember to ask this question in the past but now I have thought of something little bit difference. While I understand that asterisk dialplan accept the call to be answered[ Answer() ] in the dialplan, I wanna know if this is possible; i. A call on legacy PBX, extension to extension is made. ii. On call bridging, the legacy PBX initiate a third bridging to the recording system via an ISDN interface. iii. Conversation on Legacy continue but asterisk record this call until hangup is issued Please advice if this is possible. 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as avaya definity recording server
Oh.. I didnt know that. Thanks Sam Muro, Sam escribió: What do you mean chief? What am looking at is ability for asterisk to receive a call and recording until it tier down without bridging it to the physical device Sam Would you like the advice in all caps? He means that you put the subject in all caps. He normally gets upset with everyone that does this on the subject or in the body. I've corrected the caps in the subject to avoid further upsetting. Cheers, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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
Hi Steve Even though you shouldn't have to, have your rebooted? 200 days of uptime and this just started? It seems this problem is common as i have three boxes of the same capacity with exactly the same problem. So reboot should only solve the problem for a while Have you recently updated the box? No. ksoftirqd seems to have issues in some kernels. That is where I would start after restarting Asterisk and or the server. Allow me to look at it and revert 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] 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 System is fairly loaded, but there's still plenty of idle CPU cycles. If we were in a storm of CPU-intensive processes, we would have expected many more running processes. Right now we have none (the single process is 'top' itself). 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 Processes seem to be sorted by size. You should have pressed 'p' to go back to sorting by CPU. Now we don't even see the worst offenders. Tried option 'p' but doesnt seems to exist. Centos 5.3 kernel 2.6.18-128 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 Those are all kernel threads rather than real processes. So I suspect one of two things: 1. You're right after such a storm. The load average will decreases sharply. What do you mean Trafrir Its obvious that the effect increases with increase number of active channels. e.g. @channels=90, load average = 4 but @channels =235, load average= 60+ 2. There are many processes hung in state 'D' (uninterruptable system call). If a process is hung in such a system call for long, it normally means a problem. E.g. disk-access issues which causes all processes trying to acess a certain file to hang. I presume this should happen if there is irq sharing between disks and cards which isnt my case. -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
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
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
Hi Sam!
Hello Steve!
Are there any side-effects from the high load average? The system
doesn't seem to be CPU or disk bound from the look of the CPU stats.
System %age is
high by way - software echo cancellaton?, and Asterisk is using a lot of
cpu
which isn't suprising.
Yes. Audio quality issues. I have enabled the hardware echo cancellation
and configured
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
I'm guessing you are running 8 spans and 200+ calls into your IVR?
You are correct. 8 span which process up to 240 calls at pick time
If the system is actually performing fine then I'd just say that there
is something about the Asterisk threads that makes them look runnable
and that
accounts for the high load average. Is the IVR an agi or fastagi or
what? -
I have the agi scripts not as ivr but to help populate the required
information into mysql db. Probably here is where the problem lies i have
to connect and disconnect to mysql each time a call is made or a specific
menu is selected
Here is the script
*
#!/usr/bin/perl -w
use strict;
use DBI();
use Scalar::Util qw/weaken/;
my $cdr_log_file = /var/log/asterisk/ivr_log;
my $mysql_host = cdr01;
my $mysql_db = ivrcdrdb;
my $mysql_table = tbl_ivrcdr_details;
my $mysql_user = ivruser;
my $mysql_pwd = a09876a;
my $sth;
my $data0= $ARGV[0];
my $data1= $ARGV[1];
my $data2= $ARGV[2];
my $data3= $ARGV[3];
my $data4= $ARGV[4];
my $data5= $ARGV[5];
my $data6= $ARGV[6];
my $data7= $ARGV[7];
# Connect to database
# print Connecting to database...\n\n;
my $dbh =
DBI-connect(DBI:mysql:database=$mysql_db;host=$mysql_host,$mysql_user,$mysql_pwd,{'RaiseError'
= 1});
my $insert_str = insert into $mysql_table (calldate, language, src,
duration, accountcode, uniqueid, currentmenu, nextmenu) values
(\$data0\, \$data1\, \$data2\, \$data3\, \$data4\, \$data5\,
\$data6\, \$data7\);\n;
$sth = $dbh-prepare($insert_str);
$sth-execute();
# print \n\nOK.\n;
$sth-finish();
$dbh-disconnect();
# Trying to resolve memory leak should it happen
delete($ARGV[0]);
delete($ARGV[1]);
delete($ARGV[2]);
delete($ARGV[3]);
delete($ARGV[4]);
delete($ARGV[5]);
delete($ARGV[6]);
delete($ARGV[7]);
exit;
*
the code path may have a spinlock logic to it that means that many
threads
are runnable but when scheduled just go back to sleep. That would
account for high load average with lots of spare CPU. If that's what is
happening then I wouldn't worry much more about it.
Regards,
Steve
Regards
Sam
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[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] asterisk-users Digest, Vol 66, Issue 75
Hi Shahnawaz Have you considered how you are going to address location issue for Mobile users calling 911. You should think of SS7 MAP/TCAP to atleast know their Cell ID Regards Sam Thanks very much everybody who contributed their thoughts. I would try to get some DID's so that each physical location can be identified by 911 call Center. Regards Shahnawaz On 2010-01-29, at 2:41 PM, Kevin P. Fleming wrote: Leif Neland wrote: 2: Often callers are answered with an automated message This is 911, please hold, just to give pranksters/misdiallers a chance to hang up before disturbing the operator. Unless 911 records the incoming call right from the start, they will never hear the im-at message. And even if they do, they have to know the message is there to seek on the recording. In the US at least, calls to PSAPs are recorded from the instant the last digit is dialed, before the call is even routed and ringing (on wireline networks where this is possible, anyway). -- 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
Re: [asterisk-users] 911, Location
Hi Shahnawaz Have you considered how you are going to address location issue for Mobile users calling 911. You should think of SS7 MAP/TCAP to atleast know their Cell ID Regards Sam Thanks very much everybody who contributed their thoughts. I would try to get some DID's so that each physical location can be identified by 911 call Center. Regards Shahnawaz On 2010-01-29, at 2:41 PM, Kevin P. Fleming wrote: Leif Neland wrote: 2: Often callers are answered with an automated message This is 911, please hold, just to give pranksters/misdiallers a chance to hang up before disturbing the operator. Unless 911 records the incoming call right from the start, they will never hear the im-at message. And even if they do, they have to know the message is there to seek on the recording. In the US at least, calls to PSAPs are recorded from the instant the last digit is dialed, before the call is even routed and ringing (on wireline networks where this is possible, anyway). -- 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
Re: [asterisk-users] Asterisk as the recording server for Avaya Definity
Research wrote: . I saw a nice article on voip-info.org on how to replace voicemail server for Avaya Definity with asterisk. Could you send me the link of the article? I'll be looking into doing this within the next year. Thanks, Doug Hi Doug See: http://www.voip-info.org/wiki/view/Asterisk-Partner+ACS+for+Voicemail The problem i have is how to use this info to replace nice/witness recording server 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
