Re: [asterisk-users] Delay Before Join announcement
Logs attached. Thanks in advance! On Fri, May 30, 2014 at 11:54 PM, Prakash N prakas...@tevatel.com wrote: Hi , Can you post cli log With regards N.Prakash -- From: RSCL Mumbai rscl.mum...@gmail.com Sent: 30-05-2014 11:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Delay Before Join announcement Hi, I am using Asterisk 1.4 along with FreePBX. My call flow is as follows: Inbound DID Inbound Route Time Condition Queue. My welcome greeting MP3 is setup under System Recordings its called under Queue Join Announcement. Not sure why, the MP3 audio file starts to play after a 5 sec pause. Any thoughts or pointers are appreciated. Thanks in advance. Vai e4blra*CLI e4blra*CLI e4blra*CLI e4blra*CLI e4blra*CLI e4blra*CLI e4blra*CLI -- Remote UNIX connection -- Remote UNIX connection disconnected == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [2097205030@from-sip-external:1] NoOp(SIP/66.241.111.30-08d6, Received incoming SIP connection from unknown peer to 2097205030) in new stack -- Executing [2097205030@from-sip-external:2] Set(SIP/66.241.111.30-08d6, CDR(dnid)=2097205030) in new stack -- Executing [2097205030@from-sip-external:3] Set(SIP/66.241.111.30-08d6, __DNID=2097205030) in new stack -- Executing [2097205030@from-sip-external:4] Set(SIP/66.241.111.30-08d6, CALLERID(name)=2097205030) in new stack -- Executing [2097205030@from-sip-external:5] NoOp(SIP/66.241.111.30-08d6, USERFIELD TRIGGERED) in new stack -- Executing [2097205030@from-sip-external:6] Set(SIP/66.241.111.30-08d6, DID=2097205030) in new stack -- Executing [2097205030@from-sip-external:7] Goto(SIP/66.241.111.30-08d6, s,1) in new stack -- Goto (from-sip-external,s,1) -- Executing [s@from-sip-external:1] GotoIf(SIP/66.241.111.30-08d6, 1?checklang:noanonymous) in new stack -- Goto (from-sip-external,s,2) -- Executing [s@from-sip-external:2] GotoIf(SIP/66.241.111.30-08d6, 0?setlanguage:from-trunk,2097205030,1) in new stack -- Goto (from-trunk,2097205030,1) -- Executing [2097205030@from-trunk:1] Set(SIP/66.241.111.30-08d6, __FROM_DID=2097205030) in new stack -- Executing [2097205030@from-trunk:2] Gosub(SIP/66.241.111.30-08d6, app-blacklist-check,s,1) in new stack -- Executing [s@app-blacklist-check:1] GotoIf(SIP/66.241.111.30-08d6, 0?blacklisted) in new stack -- Executing [s@app-blacklist-check:2] Set(SIP/66.241.111.30-08d6, CALLED_BLACKLIST=1) in new stack -- Executing [s@app-blacklist-check:3] Return(SIP/66.241.111.30-08d6, ) in new stack -- Executing [2097205030@from-trunk:3] ExecIf(SIP/66.241.111.30-08d6, 0 ?Set(CALLERID(name)=6617480240)) in new stack -- Executing [2097205030@from-trunk:4] Set(SIP/66.241.111.30-08d6, __CALLINGPRES_SV=allowed_not_screened) in new stack -- Executing [2097205030@from-trunk:5] Set(SIP/66.241.111.30-08d6, CALLERPRES()=allowed_not_screened) in new stack -- Executing [2097205030@from-trunk:6] Goto(SIP/66.241.111.30-08d6, timeconditions,12,1) in new stack -- Goto (timeconditions,12,1) -- Executing [12@timeconditions:1] GotoIfTime(SIP/66.241.111.30-08d6, 00:00-23:59,mon-sun,*,*?ext-queues,5554,1) in new stack -- Goto (ext-queues,5554,1) -- Executing [5554@ext-queues:1] Macro(SIP/66.241.111.30-08d6, user-callerid,) in new stack -- Executing [s@macro-user-callerid:1] Set(SIP/66.241.111.30-08d6, AMPUSER=6617480240) in new stack -- Executing [s@macro-user-callerid:2] GotoIf(SIP/66.241.111.30-08d6, 0?report) in new stack -- Executing [s@macro-user-callerid:3] ExecIf(SIP/66.241.111.30-08d6, 1?Set(REALCALLERIDNUM=6617480240)) in new stack -- Executing [s@macro-user-callerid:4] Set(SIP/66.241.111.30-08d6, AMPUSER=) in new stack -- Executing [s@macro-user-callerid:5] Set(SIP/66.241.111.30-08d6, AMPUSERCIDNAME=) in new stack -- Executing [s@macro-user-callerid:6] GotoIf(SIP/66.241.111.30-08d6, 1?report) in new stack -- Goto (macro-user-callerid,s,10) -- Executing [s@macro-user-callerid:10] GotoIf(SIP/66.241.111.30-08d6, 0?continue) in new stack -- Executing [s@macro-user-callerid:11] Set(SIP/66.241.111.30-08d6, __TTL=64) in new stack -- Executing [s@macro-user-callerid:12] GotoIf(SIP/66.241.111.30-08d6, 1?continue) in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] NoOp(SIP/66.241.111.30-08d6, Using CallerID 2097205030 6617480240) in new stack -- Executing [5554@ext-queues:2] Answer(SIP/66.241.111.30-08d6, ) in new stack -- Executing [5554@ext-queues:3] Set(SIP/66.241.111.30-08d6, __BLKVM_OVERRIDE=BLKVM/5554/SIP/66.241.111.30-08d6) in new stack -- Executing [5554@ext-queues:4] Set(SIP
[asterisk-users] Delay Before Join announcement
Hi, I am using Asterisk 1.4 along with FreePBX. My call flow is as follows: Inbound DID Inbound Route Time Condition Queue. My welcome greeting MP3 is setup under System Recordings its called under Queue Join Announcement. Not sure why, the MP3 audio file starts to play after a 5 sec pause. Any thoughts or pointers are appreciated. Thanks in advance. Vai -- _ -- 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] Need help understanding CDR
Thank you every one. Now I understand why I was confused. I have always been using Asterisk in an Inbound environment. Hence my thought were misaligned wrt answered billed. Now I understand. Thank you all!! Is there anyway to capture the time for conversation, IVR, hold etc etc. If not inbuilt into AsteriskCDR, by way of any patch or Dialplan or any 3rd party application, more suitable for an Inbound environment. It would help a lot if I could capture fragmented distribution of time per call -- time in IVR, Queue, Call etc. Regards, Sans On Mon, Mar 18, 2013 at 4:33 PM, Asghar Mohammad asghar...@gmail.comwrote: hi, 00:00 -- Call Connected to asterisk - duration start here 00:01 -- welcome greeting starts billisec start here 00:11 -- welcome greeting ends (10 sec wav file) 00:12 -- Call enters queue and at the same time rings on first available extension 00:15 -- Call is answered by an agent 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec --- both end here duration = 01:15 bilsec = 01:14 duration start as soon as call arrived in asterisk. bilsec start as soon as call answered. exten s,1,Answer() duration and bilsec start at same time because you answered the call immidataly exten s,n,Plaback(something) exten s,n,Dial(agent) exten s,n,Hangup duration and billsec are same exten s,1,Ringing(10) -- duration start here exten s,n,Answer() bilsec start here exten s,n,Plaback(something) exten s,n,Dial(agent) exten s,n,Hangup duration and billsec end here so billsec is 10 seconds less then duration hope this will help you. On Mon, Mar 18, 2013 at 6:29 AM, RSCL Mumbai rscl.mum...@gmail.comwrote: I am using SIP. I am still a bit confused about answered billed time. For example: 00:00 -- Call Connected to asterisk 00:01 -- welcome greeting starts 00:11 -- welcome greeting ends (10 sec wav file) 00:12 -- Call enters queue and at the same time rings on first available extension 00:15 -- Call is answered by an agent 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec. In the given schematic what will be the Answered time and billed time. Thank you for the help in advance!! On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad asghar...@gmail.comwrote: If you have analog FXO ports then the call is considered answered as soon as dialing is completed not always true if FXO configured properly it should not send back answered as soon as dialed. On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.comwrote: If you have analog FXO ports then the call is considered answered as soon as dialing is completed. This does not apply to SIP, PRI, or other technologies which support far end answer detection. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Sunday, March 17, 2013 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Need help understanding CDR Hi, Attached is a sample CDR. I need some help to understand the billsec column. PS: the time value in billsec duration is same. With reference to the attached log, what does the 10 sec / 6 sec / 2 sec correspond to: (a) Time between call connection to asterisk and disconnection from asterisk? (b) Time after welcome greeting and before hangup -- the time the call rang on the extension? (c) Or any other scenario Thank you in advance. Best regards, Sans -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory
[asterisk-users] Need help understanding CDR
Hi, Attached is a sample CDR. I need some help to understand the billsec column. PS: the time value in billsec duration is same. With reference to the attached log, what does the 10 sec / 6 sec / 2 sec correspond to: (a) Time between call connection to asterisk and disconnection from asterisk? (b) Time after welcome greeting and before hangup -- the time the call rang on the extension? (c) Or any other scenario Thank you in advance. Best regards, Sans +-+--+--+-++-+---+---+-+-+-+ | calldate| clid | src | dst | dcontext | dstchannel | lastapp | lastdata | billsec | disposition | dnid| +-+--+--+-++-+---+---+-+-+-+ | 2013-03-15 17:52:53 | 19170002018 8130006555 | 8130006555 | s | app-announcement-4 | | Playback | custom/Welcome,noanswer | 10 | ANSWERED| 19170002018 | | 2013-03-12 16:32:05 | 19170002018 2810007178 | 2810007178 | s | app-announcement-4 | | Playback | custom/Welcome,noanswer | 6 | ANSWERED| 19170002018 | | 2013-03-12 16:31:55 | 19170002018 2810007178 | 2810007178 | s | app-announcement-4 | | Playback | custom/Welcome,noanswer | 2 | ANSWERED| 19170002018 | +-+--+--+-++-+---+---+-+-+-+-- _ -- 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] Need help understanding CDR
I am using SIP. I am still a bit confused about answered billed time. For example: 00:00 -- Call Connected to asterisk 00:01 -- welcome greeting starts 00:11 -- welcome greeting ends (10 sec wav file) 00:12 -- Call enters queue and at the same time rings on first available extension 00:15 -- Call is answered by an agent 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec. In the given schematic what will be the Answered time and billed time. Thank you for the help in advance!! On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad asghar...@gmail.comwrote: If you have analog FXO ports then the call is considered answered as soon as dialing is completed not always true if FXO configured properly it should not send back answered as soon as dialed. On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.com wrote: If you have analog FXO ports then the call is considered answered as soon as dialing is completed. This does not apply to SIP, PRI, or other technologies which support far end answer detection. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Sunday, March 17, 2013 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Need help understanding CDR Hi, Attached is a sample CDR. I need some help to understand the billsec column. PS: the time value in billsec duration is same. With reference to the attached log, what does the 10 sec / 6 sec / 2 sec correspond to: (a) Time between call connection to asterisk and disconnection from asterisk? (b) Time after welcome greeting and before hangup -- the time the call rang on the extension? (c) Or any other scenario Thank you in advance. Best regards, Sans -- _ -- 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] Complex Call Distribution
Hello, I have Elastix ISO install (FreePBX 2.7.0.3) My current Setup is as follows: Inbound Route Queue (Dynamic Agents) The queue distributes calls based on rrMemory. I have been asked to redesign the call distribution as follows: Calls will be delievered to Level-1 Agents (say 4 dynamic agents) in rrMemory format. When Level-1 Agents are busy, distribute calls to Level-2 Agents (say 3 dynamic agents) in rrMemory format. When Level-2 Agents are busy, distribute calls to Level-3 Agents (say 2 dynamic agents) in rrMemory format. Is it possible to setup the call distribution in the above format using any kind of logic or algorithm ? I tried using Penalties function in Queues. Created 2 penalties : 0 (level-1) and 1000 (level-2) and assigned penalties to agents (static) I made a few test calls, but Level-2 agents were delivered calls inspite of Level-1 agents being available. Any help or pointers are appreciated. Thx, Vai -- _ -- 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] MySQL Query : Calls Answered for 5 sec
Hello, I am trying to construct MySQL query(s) to get a list of calls which lasted for less than 5 seconds between a given date range. Any help is appreciated. Thank you in advance. Regards, Sans -- _ -- 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] MySQL Query : Calls Answered for 5 sec
The following query gives me calls with disposition NO ANSWER On Fri, Sep 14, 2012 at 9:50 PM, Danny Nicholas da...@debsinc.com wrote: Select * from cdr where duration 5 and (calldate= date1 and calldate = date2) ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *RSCL Mumbai *Sent:* Friday, September 14, 2012 11:16 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] MySQL Query : Calls Answered for 5 sec ** ** Hello, I am trying to construct MySQL query(s) to get a list of calls which lasted for less than 5 seconds between a given date range. Any help is appreciated. Thank you in advance. Regards, Sans -- _ -- 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] MySQL Query : Calls Answered for 5 sec
@Raj I tried your query and variation by using replacing duration with billsec. In both cases, I get results including disposition NO ANSWER On Fri, Sep 14, 2012 at 9:58 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org wrote: On Friday 14 Sep 2012, RSCL Mumbai wrote: I am trying to construct MySQL query(s) to get a list of calls which lasted for less than 5 seconds between a given date range. Any help is appreciated. On the CDR database, to get all calls that lasted 5 seconds between 2012-09-01 and 2012-09-07 (inclusive), the MySQL query would be: select * from cdr where calldate = '2012-09-01' and calldate '2012-09-08' and duration 5; Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] MySQL Query : Calls Answered for 5 sec
I need a list of calls Answered and Disconnected in less than 5 sec. Thx On Fri, Sep 14, 2012 at 10:07 PM, Warren Selby wcse...@selbytech.comwrote: On Fri, Sep 14, 2012 at 11:33 AM, RSCL Mumbai rscl.mum...@gmail.comwrote: @Raj I tried your query and variation by using replacing duration with billsec. In both cases, I get results including disposition NO ANSWER If you don't want the NO ANSWER disposition, add an AND NOT DISPOSITION = 'NO ANSWER' to your query. This is all pretty basic SQL Query writing, not specific to asterisk... -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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
[asterisk-users] Need Hosted Predictive Dialer
Hi, Seeking recommendations for a good quality hosted predictive dialer service. Low volume, single agent US dialing. Thank you. Best regards, Sans -- _ -- 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] 9. any live queue monitor recommendation? (Jean Chassoul) chass...@gmail.com
I would recommended FOP2 Its awesome. On Fri, Nov 4, 2011 at 3:04 PM, Anthony Laudini alaudini.lo...@gmail.comwrote: Hi Jean, I suggest Queuemetrics. There are many out there but this one is good for monitoring and reporting. I know there's a free version you can try. All the best Anthony -- _ -- 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 (Trixbox) - VirtualBox - Linux Host
someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. Thanks in advance.! -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -- I installed and used Elastix 2.0.3 on VirtualBox 4.x (64bit) but I were constantly troubled by high CPU usage and performance issues. I am not a virtualization / Asterisk expert so I may have missed some aspect of settings or configurations. My general reading on various forums seemed to indicate that VirtualBox is still not the best platform real time application like asterisk. My 2 cents -- _ -- 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] Lag with Call Transfer (Patching)
Hi, Using Asterisk 1.6.2.13 We are now starting to use *call transfer (patching) function.* Call flow is as follows: --- John Calls me and requests him to be connected to Nancy. I place John's call on Hold I dial Nancy and speak with her about John I then patch the call between John and Nancy My line is free for the next call while John Nancy continue their conversation In this scenario, my conversation with John Nancy is perfect. No problems. But both John Nancy report that the conversation between them (after I patched them) has a lag of about 4-5 seconds. I am not able to understand why this is happening. I am using GrandStream GXP280 IP Phone. Pls suggest what may be the problem area and how should I resolve this. Thank you. Best regards, Sans -- _ -- 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] USA Did required
My favorite is didww.com , another one is ipcomms.net (not very prompt with their customer service) Hope this helps.. On Sat, Oct 1, 2011 at 12:51 AM, amit mehta amit.magn...@gmail.com wrote: Hello members, I am looking for USA incoming DID which can be registered on softphone/IP Phone/ Pap2 devices. The DID will only be required to receive inbound calls and no outbound calls. Let me know your best per month prices/cost for the above. Regards, Amit Mehta -- _ -- 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] Anyone using Asterisk on VirtualBox ?
On Sat, Sep 3, 2011 at 1:56 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 1 Sep 2011, RSCL Mumbai wrote: I tried and failed with VirtualBox too. Timing seemed impossible to maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and nothing else going on (single instance). I don't think VirtualBox is up to real-time stuff. We use LXC now, and it is fantastic. j Thx Jeff. Kindly share some more details on the kind of hardware you are using, LXC parameters and the kind of load the system can handle. I am sure this will help me and more like myself. Thx Sanjay My main interest of being on Virtual platform is portability / Backup. In case of any h/w issues, or crashes, simply copy the VM on to another box and you are up in minutes. Sanjay Hi Sanjay, LXC is more of a quasi-virtual platform - it doesn't give you hardware virtualization, but instead lets you share the kernel of the host between multiple instances. To me this allows for multiple efficiencies and advantages that you don't get with hardware virtualization: 1) the host's memory is shared between all instances 2) the host's disk is shared between all instances 3) a shell on the host has access to the files in all of the instances So an instance that is truly idle is taking up very little resource on the host. Versus a traditional hardware virt, which even when idle has an appreciable chunk of RAM and CPU in use all the time. For hosting lots of asterisk instances this is VERY efficient. We have it setup such that the host runs an asterisk image that is the PSTN gateway and has dahdi loaded for timing and access to interface cards. It accepts calls for subscribed DIDs and routes them to the appropriate instance. Each instance has an asterisk process that is dedicated to a customer, which includes their own instance of FreePBX. The dedicated asterisk instance uses a SIP peer connection to the asterisk running on the host which is its outbound access to the PSTN (or other instances). The one gotcha I ran into was configuring the instance to allow access to the dahdi kernel module of the host, which is needed for timing for meetme (we still run 1.4). The conf file needs to contain: # dahdi lxc.cgroup.devices.allow = c 196:0 rwm lxc.cgroup.devices.allow = c 196:253 rwm lxc.cgroup.devices.allow = c 196:254 rwm lxc.cgroup.devices.allow = c 196:255 rwm This is still in proof-of-concept mode for us, but we do have a half dozen customers representing about fifty seats running on it in beta. No complaints in over two months, and the load average may as well be zero. The machine is a quad core Xeon (X3450 @ 2.66Ghz) with 8G RAM, running Ubuntu 11.04. Each instance is a subtree of the host's filesystem, by default (at least in Ubuntu) under /var/lib/lxc. We created a template with a full asterisk and FreePBX installation. To create a new instance we simply untar the template and run a sed script over a set of files to give it an IP address, hostname, and minor edits to various asterisk config files. I haven't done it yet, but I intend to create a mirror of the host machine on another box with rsync, which will serve as the backup. At some point I would like to have the instances running on both mirrors with failover. LXC docs basically suck. If you do go down this road, you will have to be prepared to glean as much as possible from notes various people have posted. I settled on Ubuntu 11.04 as a base because a lot of LXC specific scripts have been created to help with management. Even so its kind of flaky shutting down and rebooting the instances. Once they are running as you like it is stable, but I had a lot of weird things happen along the way as I was tweaking. OpenVZ is the older and more mature equivalent, and may be a better choice to start, but it is not built into the kernel as LXC is. I don't have an real comparisons to provide operationally, but I can vouch for LXC being stable enough for production use so far. I haven't stress tested it yet to see how many instances we can provide on a single host, but am hoping it to be a function of the number of simultaneous calls rather than the number of instances... Would love to hear from anyone else that is using LXC, especially in production. Cheers, j -- @Jeff, @Tarek, I finally decided to move away from Virtualization. I have read a lot of posts on various forums which suggests VB is not fully ready for a real time application like Asterisk, and I have been facing issues all the way. LXC was a bit complicated for me and I was short on time. Did a bare metal install and its working good. My Quad Xeon 2.3 GHz CPU hardly hits 10% with 20 concurrent calls I have only 2GB RAM for now and its 50% used. Created a CloneZilla image last night, plan to install it on another similar hardware later today. I am wondering how to resolve ethernet conflict while
Re: [asterisk-users] Anyone using Asterisk on VirtualBox ?
Thx @James (1) We do not use any analog / digital phone lines. SIP based DIDs and Softphones. Do I still need timing source ? (2) What does timing source do, how does ithelp ? Any insights will help. Thx Rgds, Sanjay 2011/9/2 James zhu zhulizh...@live.com hi: please check the redfone solution. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: aster...@a-domani.nl To: asterisk-users@lists.digium.com Date: Thu, 1 Sep 2011 23:48:46 +0200 Subject: Re: [asterisk-users] Anyone using Asterisk on VirtualBox ? On Thu, 2011-09-01 at 21:32 +0530, RSCL Mumbai wrote: My main interest of being on Virtual platform is portability / Backup. In case of any h/w issues, or crashes, simply copy the VM on to another box and you are up in minutes. Sanjay -- Doing that right now, although in my case i use XEN. Besides being hw independant, it is easier to play with a different version for a while (1.4 / 1.6.0 / 1.6.1 / 1.6.2 / 1.8.0) and being able to switch back in minutes. -- _ -- 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] cli command show codecs
Bruce, that's exactly the command I was looking for. Thx a ton. Sans On Thu, Sep 1, 2011 at 12:17 AM, Bruce B bruceb...@gmail.com wrote: sip show channels is the command you are looking for. On Wed, Aug 31, 2011 at 2:45 PM, RSCL Mumbai rscl.mum...@gmail.comwrote: asterisk -rx core show channels verbose does not provide transcoding details. Unless I have missed something. Sans On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas da...@debsinc.comwrote: Core show channels verbose is probably your best bet. I think the answer also depends on your * version. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *RSCL Mumbai *Sent:* Wednesday, August 31, 2011 10:44 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] cli command show codecs ** ** Hi, Is there a CLI command which will tell me the codec used for active calls and if transcoding is happening ? Thx Sans -- _ -- 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 -- _ -- 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] cli command show codecs
Hi, Does audio files have codec formats? I simply convert all my audios (MOH, accouncements) to .wav format, 16bit, 11kHz (I believe this is the best format for asterisk). I am new to this and may be incorrect. Going forward, (a) How can I check the codec format of my announcements, MOH ? (b) How can I record/convert announcements, MoH etc to a particular format ? I believe its a good idea to prevent transcoding and save CPU overheads. Thx Sans On Thu, Sep 1, 2011 at 11:39 AM, Bruce B bruceb...@gmail.com wrote: if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If your IVR announcement is not recorded in g729 and you see g729 on the channel when you call into IVR then it's transcoding as well. On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling ewiel...@nyigc.com wrote: Assuming SIP sip show channels will show you which codec is used for each call leg. However it does not track transcoding. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Wednesday, August 31, 2011 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs asterisk -rx core show channels verbose does not provide transcoding details. Unless I have missed something. Sans On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas da...@debsinc.com wrote: Core show channels verbose is probably your best bet. I think the answer also depends on your * version. From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Wednesday, August 31, 2011 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cli command show codecs Hi, Is there a CLI command which will tell me the codec used for active calls and if transcoding is happening ? Thx Sans -- _ -- 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 -- _ -- 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] cli command show codecs
Thx @Danny I am feeling a bit lost here... We are using G711-aLaw for all our calls (endpoints) and I would like to align everything to this codec. I have an MOH file -- a custom wav file. How do I check its codec format ? And if its not G711-aLaw, how do I convert it to G711-aLaw. Thank you. Sans On Thu, Sep 1, 2011 at 6:35 PM, Danny Nicholas da...@debsinc.com wrote: Maybe this will be better than my first answer – Audio files do indeed have codec formats. If you are in a particular codec (say G729), Playback/Background and MOH will search for files that match the codec format first, then transcode WAV/GSM/whatever you have to that format if it isn’t found. Ideally, you want to have a copy of each codec you can play for all sounds and MOH. Each of the “canned sounds” comes in each codec format (you pick the ones you want when you do make menuselect). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Thursday, September 01, 2011 5:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs Hi, Does audio files have codec formats? I simply convert all my audios (MOH, accouncements) to .wav format, 16bit, 11kHz (I believe this is the best format for asterisk). I am new to this and may be incorrect. Going forward, (a) How can I check the codec format of my announcements, MOH ? (b) How can I record/convert announcements, MoH etc to a particular format ? I believe its a good idea to prevent transcoding and save CPU overheads. Thx Sans On Thu, Sep 1, 2011 at 11:39 AM, Bruce B bruceb...@gmail.com wrote: if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If your IVR announcement is not recorded in g729 and you see g729 on the channel when you call into IVR then it's transcoding as well. On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling ewiel...@nyigc.com wrote: Assuming SIP sip show channels will show you which codec is used for each call leg. However it does not track transcoding. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Wednesday, August 31, 2011 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs asterisk -rx core show channels verbose does not provide transcoding details. Unless I have missed something. Sans On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas da...@debsinc.com wrote: Core show channels verbose is probably your best bet. I think the answer also depends on your * version. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Wednesday, August 31, 2011 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cli command show codecs Hi, Is there a CLI command which will tell me the codec used for active calls and if transcoding is happening ? Thx Sans -- _ -- 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 -- _ -- 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
Re: [asterisk-users] cli command show codecs
Thanks again @Danny. File converter worked like a charm. asterisk -rx file convert /var/lib/asterisk/mohmp3/wav_Track11.wav wav_Track11.alaw I coped the new file from sounds/ folder to my desktop And I tried to upload the new .alaw file using FreePBX, I got the following error: Error Processing: sox failed to convert file and original could not be copied as a fall back for wav_Track111.alaw! This is not a fatal error, your Music on Hold may still work. Pls help with this last bit. Thx Sans On Thu, Sep 1, 2011 at 7:17 PM, Danny Nicholas da...@debsinc.com wrote: Asterisk has a built-in file convert help file convert Usage: file convert file_in file_out Convert from file_in to file_out. If an absolute path is not given, the default Asterisk sounds directory will be used. Example: file convert tt-weasels.gsm tt-weasels.ulaw -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Thursday, September 01, 2011 8:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs Thx @Danny I am feeling a bit lost here... We are using G711-aLaw for all our calls (endpoints) and I would like to align everything to this codec. I have an MOH file -- a custom wav file. How do I check its codec format ? And if its not G711-aLaw, how do I convert it to G711-aLaw. Thank you. Sans On Thu, Sep 1, 2011 at 6:35 PM, Danny Nicholas da...@debsinc.com wrote: Maybe this will be better than my first answer – Audio files do indeed have codec formats. If you are in a particular codec (say G729), Playback/Background and MOH will search for files that match the codec format first, then transcode WAV/GSM/whatever you have to that format if it isn’t found. Ideally, you want to have a copy of each codec you can play for all sounds and MOH. Each of the “canned sounds” comes in each codec format (you pick the ones you want when you do make menuselect). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Thursday, September 01, 2011 5:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs Hi, Does audio files have codec formats? I simply convert all my audios (MOH, accouncements) to .wav format, 16bit, 11kHz (I believe this is the best format for asterisk). I am new to this and may be incorrect. Going forward, (a) How can I check the codec format of my announcements, MOH ? (b) How can I record/convert announcements, MoH etc to a particular format ? I believe its a good idea to prevent transcoding and save CPU overheads. Thx Sans On Thu, Sep 1, 2011 at 11:39 AM, Bruce B bruceb...@gmail.com wrote: if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If your IVR announcement is not recorded in g729 and you see g729 on the channel when you call into IVR then it's transcoding as well. On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling ewiel...@nyigc.com wrote: Assuming SIP sip show channels will show you which codec is used for each call leg. However it does not track transcoding. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Wednesday, August 31, 2011 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs asterisk -rx core show channels verbose does not provide transcoding details. Unless I have missed something. Sans On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas da...@debsinc.com wrote: Core show channels verbose is probably your best bet. I think the answer also depends on your * version. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Wednesday, August 31, 2011 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cli command show codecs Hi, Is there a CLI command which will tell me the codec used for active calls and if transcoding is happening ? Thx Sans -- _ -- 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
Re: [asterisk-users] cli command show codecs
Surprisingly, despite the error message, the files is uploaded in /var/lib/asterisk/mohmp3 with correct permissions and ownership. Its not showing in FreePBX MOH Screen. I guess its a FreePBX issue. Sans On Thu, Sep 1, 2011 at 7:56 PM, RSCL Mumbai rscl.mum...@gmail.com wrote: Thanks again @Danny. File converter worked like a charm. asterisk -rx file convert /var/lib/asterisk/mohmp3/wav_Track11.wav wav_Track11.alaw I coped the new file from sounds/ folder to my desktop And I tried to upload the new .alaw file using FreePBX, I got the following error: Error Processing: sox failed to convert file and original could not be copied as a fall back for wav_Track111.alaw! This is not a fatal error, your Music on Hold may still work. Pls help with this last bit. Thx Sans On Thu, Sep 1, 2011 at 7:17 PM, Danny Nicholas da...@debsinc.com wrote: Asterisk has a built-in file convert help file convert Usage: file convert file_in file_out Convert from file_in to file_out. If an absolute path is not given, the default Asterisk sounds directory will be used. Example: file convert tt-weasels.gsm tt-weasels.ulaw -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Thursday, September 01, 2011 8:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs Thx @Danny I am feeling a bit lost here... We are using G711-aLaw for all our calls (endpoints) and I would like to align everything to this codec. I have an MOH file -- a custom wav file. How do I check its codec format ? And if its not G711-aLaw, how do I convert it to G711-aLaw. Thank you. Sans On Thu, Sep 1, 2011 at 6:35 PM, Danny Nicholas da...@debsinc.com wrote: Maybe this will be better than my first answer – Audio files do indeed have codec formats. If you are in a particular codec (say G729), Playback/Background and MOH will search for files that match the codec format first, then transcode WAV/GSM/whatever you have to that format if it isn’t found. Ideally, you want to have a copy of each codec you can play for all sounds and MOH. Each of the “canned sounds” comes in each codec format (you pick the ones you want when you do make menuselect). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Thursday, September 01, 2011 5:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs Hi, Does audio files have codec formats? I simply convert all my audios (MOH, accouncements) to .wav format, 16bit, 11kHz (I believe this is the best format for asterisk). I am new to this and may be incorrect. Going forward, (a) How can I check the codec format of my announcements, MOH ? (b) How can I record/convert announcements, MoH etc to a particular format ? I believe its a good idea to prevent transcoding and save CPU overheads. Thx Sans On Thu, Sep 1, 2011 at 11:39 AM, Bruce B bruceb...@gmail.com wrote: if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If your IVR announcement is not recorded in g729 and you see g729 on the channel when you call into IVR then it's transcoding as well. On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling ewiel...@nyigc.com wrote: Assuming SIP sip show channels will show you which codec is used for each call leg. However it does not track transcoding. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Wednesday, August 31, 2011 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs asterisk -rx core show channels verbose does not provide transcoding details. Unless I have missed something. Sans On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas da...@debsinc.com wrote: Core show channels verbose is probably your best bet. I think the answer also depends on your * version. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Wednesday, August 31, 2011 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cli command show codecs Hi, Is there a CLI command which will tell me the codec used for active calls and if transcoding is happening ? Thx Sans -- _ -- 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] Anyone using Asterisk on VirtualBox ?
Hi, Anyone using Asterisk on Virtualbox. I am using and facing CPU peaking issue. Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores and 2 GM RAM allocated to the asterisk VM -- thats the only VM as of now), 64bit CentOS 5.4. Only SIP and softphones. Max 10 simultaneous calls. Unable to ascertain if the problem is with Asterisk, Virtualbox, Configuration, or the whole system should not be the way it is. Anyone will to share their settings and help me. Thx Sanjay -- _ -- 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] Anyone using Asterisk on VirtualBox ?
On Thu, Sep 1, 2011 at 8:02 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 1 Sep 2011, RSCL Mumbai wrote: Hi, Anyone using Asterisk on Virtualbox. I am using and facing CPU peaking issue. Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores and 2 GM RAM allocated to the asterisk VM -- thats the only VM as of now), 64bit CentOS 5.4. Only SIP and softphones. Max 10 simultaneous calls. Unable to ascertain if the problem is with Asterisk, Virtualbox, Configuration, or the whole system should not be the way it is. Anyone will to share their settings and help me. Thx Sanjay I tried and failed with VirtualBox too. Timing seemed impossible to maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and nothing else going on (single instance). I don't think VirtualBox is up to real-time stuff. We use LXC now, and it is fantastic. j Thx Jeff. Kindly share some more details on the kind of hardware you are using, LXC parameters and the kind of load the system can handle. I am sure this will help me and more like myself. Thx Sanjay -- _ -- 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] Anyone using Asterisk on VirtualBox ?
On Thu, Sep 1, 2011 at 9:25 PM, RSCL Mumbai rscl.mum...@gmail.com wrote: On Thu, Sep 1, 2011 at 8:02 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 1 Sep 2011, RSCL Mumbai wrote: Hi, Anyone using Asterisk on Virtualbox. I am using and facing CPU peaking issue. Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores and 2 GM RAM allocated to the asterisk VM -- thats the only VM as of now), 64bit CentOS 5.4. Only SIP and softphones. Max 10 simultaneous calls. Unable to ascertain if the problem is with Asterisk, Virtualbox, Configuration, or the whole system should not be the way it is. Anyone will to share their settings and help me. Thx Sanjay I tried and failed with VirtualBox too. Timing seemed impossible to maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and nothing else going on (single instance). I don't think VirtualBox is up to real-time stuff. We use LXC now, and it is fantastic. j Thx Jeff. Kindly share some more details on the kind of hardware you are using, LXC parameters and the kind of load the system can handle. I am sure this will help me and more like myself. Thx Sanjay My main interest of being on Virtual platform is portability / Backup. In case of any h/w issues, or crashes, simply copy the VM on to another box and you are up in minutes. Sanjay -- _ -- 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] cli command show codecs
Hi, Is there a CLI command which will tell me the codec used for active calls and if transcoding is happening ? Thx Sans -- _ -- 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] cli command show codecs
asterisk -rx core show channels verbose does not provide transcoding details. Unless I have missed something. Sans On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas da...@debsinc.com wrote: Core show channels verbose is probably your best bet. I think the answer also depends on your * version. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *RSCL Mumbai *Sent:* Wednesday, August 31, 2011 10:44 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] cli command show codecs ** ** Hi, Is there a CLI command which will tell me the codec used for active calls and if transcoding is happening ? Thx Sans -- _ -- 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] Firewall Issue
Update: Yesterday I did not observe any unexpected traffic. So far so good. Thx Sans On Mon, Aug 8, 2011 at 9:24 PM, Антон Квашёнкин anton.juga...@gmail.comwrote: Ok, run your script and then do this: service iptables save And by the way, list chkconfig --list iptables output. 2011/8/8 RSCL Mumbai rscl.mum...@gmail.com On Mon, Aug 8, 2011 at 6:03 PM, Faisal Hanif fai...@vopium.com wrote: If you take a bit deep analyses on SIP packet you will be able to understand the issue, ** ** Iptables filter on layer-3 while SIP is on layer-7. It is easily possible to generate a SIP packet with different source-ip than physical interface. ** ** You can also simulate it if you set external-ip=some-else-ip in SIP.com in asterisk. All you SIP packets will contain new some-else-ip while layer-3 headers will still have actual physical interface IP. I am usingOS (Elastix distribution). I am not really a champ at system administration hence this went over the top. I will observe the system tonight and send my feedback tomorrow. Thx to everyone for being with me on this. Sans -- _ -- 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
Re: [asterisk-users] Firewall Issue
Hi, (1) Since a few days, I am seeing unexpected (unwanted) calls reaching my asterisk server. Please see attached log files. (2) I believe the source IP of these calls is the IP mentioned under the CHANNELS column. (3) But as per my firewall, these calls should not have reached Asterisk. The should have been dropped by the Firewall. Please suggest if my thinking is in the correct direction, and what should be my next step. Best regards, Sans +-+--+--+-++-++-+--+--+-+-+--+-+-+---+-+ | calldate| clid | src | dst | dcontext | channel | dstchannel | lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | uniqueid| userfield | dnid| +-+--+--+-++-++-+--+--+-+-+--+-+-+---+-+ | 2011-08-04 11:23:15 | 000441913561021 asterisk | asterisk | s | from-trunk | SIP/94.247.178.106-0285 || Hangup | | 19 | 19 | ANSWERED|3 | | 1312471395.2207 | | 000441913561021 | +-+--+--+-++-++-+--+--+-+-+--+-+-+---+-+ +-+--+--+-++++-+--+--+-+-+--+-+-+---+-+ | 2011-08-04 15:26:19 | 001441913561025 asterisk | asterisk | s | from-trunk | SIP/72.32.198.159-0401 || Hangup | | 18 | 18 | ANSWERED|3 | | 1312485979.6667 | | 001441913561025 | +-+--+--+-++++-+--+--+-+-+--+-+-+---+-+ +-+--+--+-++-++-+--+--+-+-+--+-+-+---+-+ | 2011-08-04 17:51:12 | 002441913561017 asterisk | asterisk | s | from-trunk | SIP/50.28.9.55-04b4 || Hangup | | 19 | 18 | ANSWERED|3 | | 1312494672.7195 | | 002441913561017 | +-+--+--+-++-++-+--+--+-+-+--+-+-+---+-+ +-++--+-++-++-+--+--+-+-+--+-+-+---+---+ | 2011-08-04 16:20:20 | 2441913561035 asterisk | asterisk | s | from-trunk | SIP/75.125.193.162-0446 || Hangup | | 16 | 16 | ANSWERED|3 | | 1312489220.6866 | | 2441913561035 | +-++--+-++-++-+--+--+-+-+--+-+-+---+---+ -- _ -- 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] Firewall Issue
On Mon, Aug 8, 2011 at 4:20 PM, Антон Квашёнкин anton.juga...@gmail.comwrote: Hi, Could you attach iptables-save output. iptables-save output is blank -- no output. Not sure why ? Thx Sans -- _ -- 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] Firewall Issue
On Mon, Aug 8, 2011 at 5:09 PM, Henrik sing...@common-hacking.org wrote: ** Also you can set allowguest=no in sip.conf, if you didn't do it already I will check sip.conf, but logically, the packets should not be reaching Asterisk. IP Tables should have blocked them. Sans -- _ -- 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] Firewall Issue
For some unknown reason, the firewall script was not executed. Now I get the output of iptables-save. May be this is the reason why unwanted packets hit the system... a big blunder. Sans On Mon, Aug 8, 2011 at 5:44 PM, RSCL Mumbai rscl.mum...@gmail.com wrote: On Mon, Aug 8, 2011 at 4:20 PM, Антон Квашёнкин anton.juga...@gmail.comwrote: Hi, Could you attach iptables-save output. iptables-save output is blank -- no output. Not sure why ? Thx Sans [root@e1 ~]# iptables-save # Generated by iptables-save v1.3.5 on Mon Aug 8 08:19:37 2011 *filter :INPUT DROP [1:78] :FORWARD DROP [0:0] :OUTPUT ACCEPT [2496:492015] -A INPUT -i lo -j ACCEPT -A INPUT -p icmp -m icmp --icmp-type 8 -m state --state NEW -j ACCEPT -A INPUT -i eth1 -m state --state RELATED,ESTABLISHED -j ACCEPT -A INPUT -i eth0 -m state --state RELATED,ESTABLISHED -j ACCEPT -A INPUT -p tcp -m tcp --dport 3100 -j ACCEPT -A INPUT -p tcp -m tcp --dport 4142 -j ACCEPT -A INPUT -i eth0 -p tcp -m tcp --dport 4445 -j ACCEPT -A INPUT -i eth1 -p tcp -m tcp --dport 4445 -j ACCEPT -A INPUT -s 67.18.110.210 -i eth1 -p tcp -m tcp --dport 3306 -j ACCEPT -A INPUT -s 192.168.1.0/255.255.255.0 -i eth0 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 192.168.1.0/255.255.255.0 -i eth0 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 192.168.1.0/255.255.255.0 -i eth0 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 61.16.181.9 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 61.16.181.9 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 61.16.181.9 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 203.109.120.65 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 203.109.120.65 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 203.109.120.65 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 81.201.82.128/255.255.255.192 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.82.128/255.255.255.192 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.82.128/255.255.255.192 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 81.201.83.0/255.255.255.192 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.83.0/255.255.255.192 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.83.0/255.255.255.192 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 81.201.84.0/255.255.255.0 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.84.0/255.255.255.0 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.84.0/255.255.255.0 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 81.201.86.0/255.255.255.192 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.86.0/255.255.255.192 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.86.0/255.255.255.192 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 74.55.98.122 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 74.55.98.122 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 74.55.98.122 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 74.55.98.120 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 74.55.98.120 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 74.55.98.120 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 64.154.41.150 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 64.154.41.150 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 64.154.41.150 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 64.154.41.100 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 64.154.41.100 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 64.154.41.100 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 46.19.209.8/255.255.255.248 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.209.8/255.255.255.248 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.209.72/255.255.255.248 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.209.72/255.255.255.248 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.210.8/255.255.255.248 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.210.8/255.255.255.248 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.210.72/255.255.255.248 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.210.72/255.255.255.248 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.85.224.40/255.255.255.254 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.85.224.40/255.255.255.254 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 212.150.88.20/255.255.255.252 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 212.150.88.20/255.255.255.252 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s
Re: [asterisk-users] Firewall Issue
2011/8/8 Антон Квашёнкин anton.juga...@gmail.com lsmod | grep ipt And what distribution do you use? [root@e1 ~]# lsmod | grep ipt ipt_REJECT 38977 1 iptable_filter 36161 1 iptable_nat40773 0 ip_nat 53101 1 iptable_nat ip_conntrack 91621 3 xt_state,iptable_nat,ip_nat ip_tables 55201 2 iptable_filter,iptable_nat x_tables 50505 5 ipt_REJECT,xt_tcpudp,xt_state,iptable_nat,ip_tables -- _ -- 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] Firewall Issue
On Mon, Aug 8, 2011 at 6:03 PM, Faisal Hanif fai...@vopium.com wrote: If you take a bit deep analyses on SIP packet you will be able to understand the issue, ** ** Iptables filter on layer-3 while SIP is on layer-7. It is easily possible to generate a SIP packet with different source-ip than physical interface. ** ** You can also simulate it if you set external-ip=some-else-ip in SIP.com in asterisk. All you SIP packets will contain new some-else-ip while layer-3 headers will still have actual physical interface IP. I am usingOS (Elastix distribution). I am not really a champ at system administration hence this went over the top. I will observe the system tonight and send my feedback tomorrow. Thx to everyone for being with me on this. Sans -- _ -- 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] Firewall Issue
Hi, I seem to be facing an intrusion issue, inspite of firewall (script attached). What am I missing ?? Any suggestions / recommendation are welcome pls. Best regards, Sans #!/bin/bash echo 0 /proc/sys/net/ipv4/ip_forward # Clear any existing firewall stuff before we start /sbin/iptables --flush # As the default policies, drop all incoming traffic but allow all # outgoing traffic. This will allow us to make outgoing connections # from any port, but will only allow incoming connections on the ports # specified below. /sbin/iptables --policy INPUT DROP /sbin/iptables --policy FORWARD DROP /sbin/iptables --policy OUTPUT ACCEPT # Allow all incoming traffic if it is coming from the local loopback device /sbin/iptables -A INPUT -i lo -j ACCEPT # Allow icmp input so that people can ping us /sbin/iptables -A INPUT -p icmp --icmp-type 8 -m state --state NEW -j ACCEPT # Allow returning packets /sbin/iptables -A INPUT -i eth1 -m state --state RELATED,ESTABLISHED -j ACCEPT /sbin/iptables -A INPUT -i eth0 -m state --state RELATED,ESTABLISHED -j ACCEPT # Allow incoming traffic on port 8000 for web server 2200 for SSh /sbin/iptables -A INPUT -p tcp --dport 8000 -j ACCEPT /sbin/iptables -A INPUT -p tcp --dport 2200 -j ACCEPT # ## RESTRICTED SIP ACCESS # # LAN /sbin/iptables -A INPUT -p tcp -i eth0 -s 192.168.1.0/24 --dport 5060:5062 -j ACCEPT /sbin/iptables -A INPUT -p udp -i eth0 -s 192.168.1.0/24 --dport 5060:5062 -j ACCEPT /sbin/iptables -A INPUT -p udp -i eth0 -s 192.168.1.0/24 --dport 1:2 -j ACCEPT # Allow traffic from VoIP Service Provider /sbin/iptables -A INPUT -p udp -i eth1 -s 11.11.11.11 --dport 5060:5062 -j ACCEPT /sbin/iptables -A INPUT -p tcp -i eth1 -s 11.11.11.11 --dport 5060:5062 -j ACCEPT /sbin/iptables -A INPUT -p udp -i eth1 -s 11.11.11.11 --dport 1:2 -j ACCEPT # Check new packets are SYN packets for syn-flood protection /sbin/iptables -A INPUT -p tcp ! --syn -m state --state NEW -j DROP # Drop fragmented packets /sbin/iptables -A INPUT -f -j DROP # Drop malformed XMAS packets /sbin/iptables -A INPUT -p tcp --tcp-flags ALL ALL -j DROP # Drop null packets /sbin/iptables -A INPUT -p tcp --tcp-flags ALL NONE -j DROP # Log and drop any packets that are not allowed. You will probably want to turn off the logging #/sbin/iptables -A INPUT -j LOG --log-level 4 /sbin/iptables -A INPUT -j REJECT -- _ -- 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] How to secure our Asterisk server from hacker's ?
I want to secure my server from the hacker's. What is the case by which I can protest it. I have done security of Dialplan, Sip,IAX base security. For linux we are working on Iptables. What else is left so that I will do it too... Can you share the steps / scripts / settings done to secure Dialplan, Sip, IAX It would be a good starting point for others too. Sans -- _ -- 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-cpu utilization 60 %
CPU utilization is constantly above 24% without any call activity.. *top - 05:53:09 up 1:28, 2 users, load average: 0.18, 0.27, 0.29 Tasks: 79 total, 1 running, 78 sleeping, 0 stopped, 0 zombie Cpu(s): 9.7%us, 2.3%sy, 0.0%ni, 87.8%id, 0.0%wa, 0.2%hi, 0.0%si, 0.0%st Mem: 1026824k total, 311300k used, 715524k free,19644k buffers Swap: 2064376k total,0k used, 2064376k free, 115668k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 2154 asterisk 15 0 765m 27m 10m S 24.0 2.7 6:11.13 asterisk 1 root 15 0 10348 692 580 S 0.0 0.1 0:03.59 init 2 root RT -5 000 S 0.0 0.0 0:02.13 migration/0 3 root 34 19 000 S 0.0 0.0 0:00.03 ksoftirqd/0 4 root RT -5 000 S 0.0 0.0 0:08.06 watchdog/0 5 root RT -5 000 S 0.0 0.0 0:00.09 migration/1 * Pls lister, help me, this is driving me crazy... Thx Sans -- _ -- 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-cpu utilization 60 %
logger.conf is only set for: full = notice,warning,error,debug I have now removed debug. On Fri, May 20, 2011 at 2:57 PM, Thorsten Göllner t...@ovm-group.com wrote: Maybe IO-Activity caused by intensive logging. Take a look at your Log-Files. Maybe one or more log files a growing rapidly? Am 20.05.2011 11:24, schrieb RSCL Mumbai: CPU utilization is constantly above 24% without any call activity.. *top - 05:53:09 up 1:28, 2 users, load average: 0.18, 0.27, 0.29 Tasks: 79 total, 1 running, 78 sleeping, 0 stopped, 0 zombie Cpu(s): 9.7%us, 2.3%sy, 0.0%ni, 87.8%id, 0.0%wa, 0.2%hi, 0.0%si, 0.0%st Mem: 1026824k total, 311300k used, 715524k free,19644k buffers Swap: 2064376k total,0k used, 2064376k free, 115668k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 2154 asterisk 15 0 765m 27m 10m S 24.0 2.7 6:11.13 asterisk 1 root 15 0 10348 692 580 S 0.0 0.1 0:03.59 init 2 root RT -5 000 S 0.0 0.0 0:02.13 migration/0 3 root 34 19 000 S 0.0 0.0 0:00.03 ksoftirqd/0 4 root RT -5 000 S 0.0 0.0 0:08.06 watchdog/0 5 root RT -5 000 S 0.0 0.0 0:00.09 migration/1* -- _ -- 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-cpu utilization 60 %
This seems to be an interesting post: http://forums.virtualbox.org/viewtopic.php?t=12903 As per OP's message, CONFIG_HG is indeed 1000 [root@e1 ~]# grep CONFIG_HZ /boot/config-2.6.18-194.3.1.el5 # CONFIG_HZ_100 is not set # CONFIG_HZ_250 is not set CONFIG_HZ_1000=y CONFIG_HZ=1000 [root@e1 ~]# Not sure if I need to recompile the kernel. I will not be able to do this myself anyways... not competent yet. Sans -- _ -- 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-cpu utilization 60 %
I think I managed to solve this issue The problem lay in the VirtualBox setting for the VM. I will post the exact setting tomorrow which should help others. Sorry for being a trouble to others :( Best regards have a great weekend. Sans On Fri, May 20, 2011 at 3:03 PM, RSCL Mumbai rscl.mum...@gmail.com wrote: This seems to be an interesting post: http://forums.virtualbox.org/viewtopic.php?t=12903 As per OP's message, CONFIG_HG is indeed 1000 [root@e1 ~]# grep CONFIG_HZ /boot/config-2.6.18-194.3.1.el5 # CONFIG_HZ_100 is not set # CONFIG_HZ_250 is not set CONFIG_HZ_1000=y CONFIG_HZ=1000 [root@e1 ~]# Not sure if I need to recompile the kernel. I will not be able to do this myself anyways... not competent yet. Sans -- _ -- 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] Dropping incompatible voice frame
Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 tail -f full shows the below: [May 19 12:00:53] NOTICE[6821] channel.c: Dropping incompatible voice frame on SIP/voxbone.com-0139 of format ulaw since our native format has changed to 0x8 (alaw) [May 19 12:01:05] NOTICE[6827] channel.c: Dropping incompatible voice frame on SIP/4420-013a of format alaw since our native format has changed to 0x4 (ulaw) I am confused... In the first line, it says native format has changed to alaw and next line it says native format has changed to ulaw... Thx Sanjay -- _ -- 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] Dropping incompatible voice frame
But why does *our *native format keep changing :) Going by layman terms, if native format is alaw and someone speaks to me in uLaw, I will say *format changed*. But if native format is alaw and someone is talking with me in alaw, I should be happy. On Thu, May 19, 2011 at 10:28 PM, Terry Brummell te...@brummell.net wrote: For 2 different hosts. SIP/voxbone.com and SIP/4420 -- *From:* RSCL Mumbai *Sent:* Thu 5/19/2011 12:23 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Dropping incompatible voice frame Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 tail -f full shows the below: [May 19 12:00:53] NOTICE[6821] channel.c: Dropping incompatible voice frame on SIP/voxbone.com-0139 of format ulaw since our native format has changed to 0x8 (alaw) [May 19 12:01:05] NOTICE[6827] channel.c: Dropping incompatible voice frame on SIP/4420-013a of format alaw since our native format has changed to 0x4 (ulaw) I am confused... In the first line, it says native format has changed to alaw and next line it says native format has changed to ulaw... Thx Sanjay -- _ -- 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-cpu utilization 60 %
Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 Anyone else facing high CPU usage problem with Asterisk 1.6.2.13 or any Elastix 2.0.3 users here ? With just 3 concurrent calls and none in queue, the CPU is constantly above 40%. The moment CPU goes above 50%, calls start to break. I am a newbie and at lack of options... Sans -- _ -- 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-cpu utilization 60 %
On Mon, May 16, 2011 at 6:19 PM, Pezhman Lali l...@lopl.net wrote: check your running process, if you have more than one asterisk in your top re install your asterisk. On Sun, May 15, 2011 at 7:07 PM, Satish Patel satish...@hotmail.comwrote: Check this out http://www.moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/ Moving forward with the suggestion provided on the above link, I have the activity dump of all asterisk processes when the load was 22%. Need help in understanding the output. What should I look for which would indicate undue CPU utilization. Thank you every one for your continued support. Thread 45 (Thread 0x4175d940 (LWP 4129)): #0 0x00307500aee9 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 #1 0x004df6a3 in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 44 (Thread 0x417d9940 (LWP 4130)): #0 0x0030744cb186 in poll () from /lib64/libc.so.6 #1 0x0042d181 in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 43 (Thread 0x41855940 (LWP 4131)): #0 0x00307500aee9 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 #1 0x00490ea3 in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 42 (Thread 0x41ef4940 (LWP 4132)): #0 0x00307449a3f1 in nanosleep () from /lib64/libc.so.6 #1 0x004de5df in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 41 (Thread 0x413ed940 (LWP 4133)): #0 0x0030744cb186 in poll () from /lib64/libc.so.6 #1 0x004ea175 in ast_wait_for_input () #2 0x004e0a61 in ast_tcptls_server_root () #3 0x004eb76e in ?? () #4 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #5 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 40 (Thread 0x4148e940 (LWP 4134)): #0 0x00307500aee9 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 #1 0x00460fc9 in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 39 (Thread 0x41ce8940 (LWP 4135)): #0 0x00307500aee9 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 #1 0x004df6a3 in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 38 (Thread 0x4150a940 (LWP 4136)): #0 0x0030744cd212 in select () from /lib64/libc.so.6 #1 0x00473315 in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 37 (Thread 0x418d1940 (LWP 4137)): #0 0x00307500aee9 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 #1 0x2aaab9d46e03 in ast_unregister_file_version () from /usr/lib64/asterisk/modules/res_timing_pthread.so #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 36 (Thread 0x4194d940 (LWP 4138)): #0 0x0030744cb186 in poll () from /lib64/libc.so.6 #1 0x0048a4d0 in ast_io_wait () #2 0x2aaac5360d3b in ?? () from /usr/lib64/asterisk/modules/pbx_dundi.so #3 0x004eb76e in ?? () #4 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #5 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 35 (Thread 0x419c9940 (LWP 4139)): #0 0x00307449a3f1 in nanosleep () from /lib64/libc.so.6 #1 0x00307449a214 in sleep () from /lib64/libc.so.6 #2 0x2aaac5360ba4 in ?? () from /usr/lib64/asterisk/modules/pbx_dundi.so #3 0x004eb76e in ?? () #4 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #5 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 34 (Thread 0x41d64940 (LWP 4140)): #0 0x00307449a3f1 in nanosleep () from /lib64/libc.so.6 #1 0x00307449a214 in sleep () from /lib64/libc.so.6 #2 0x2aaac5357272 in ast_unregister_file_version () from /usr/lib64/asterisk/modules/pbx_dundi.so #3 0x004eb76e in ?? () #4 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #5 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 33 (Thread 0x41256940 (LWP 4141)): #0 0x00307449a3f1 in nanosleep () from /lib64/libc.so.6 #1 0x2aaac1eff28e in ast_unregister_file_version () from /usr/lib64/asterisk/modules/pbx_spool.so #2 0x004eb76e in
Re: [asterisk-users] Asterisk-cpu utilization 60 %
http://www.moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/ Moving forward with the suggestion provided on the above link, I have the activity dump of all asterisk processes when the load was 22%. Need help in understanding the output. What should I look for which would indicate undue CPU utilization. Any finding in my *asterisk.stack.txt ? *Thank you.* * -- _ -- 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-cpu utilization 60 %
On Sat, May 14, 2011 at 11:43 AM, Leandro Dardini ldard...@gmail.comwrote: Check if someone is brute forcing your asterisk accounts. It used to happen to me before I install fail2ban. You can easily check the full log of asterisk or with just a tcpdump -i any -n port 5060 or port 4569. Thx for the tcpdump command. Checked, all looks good. Packets coming from trusted domains only. What should be the next step ? Thx Sans -- _ -- 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 1.6: Custom Name for Recordings file
Hi, I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13) I would like to customize the file name of call recordings: /var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav I would like to include the extension number in the file name. Did a lot of googling but not much help. Pls advice. Thx Sans -- _ -- 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 1.6: Custom Name for Recordings file
On Fri, May 13, 2011 at 11:07 PM, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6: Custom Name for
Recordings file
Hi,
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not much help.
Pls advice.
See the fname_base information below.
pbx*CLI core show application monitor
-= Info about application 'Monitor' =-
[Synopsis]
Monitor a channel.
[Description]
Used to start monitoring a channel. The channel's input and output voice
packets are logged to files until the channel hangs up or monitoring is
stopped
by the StopMonitor application.
By default, files are stored to /var/spool/asterisk/monitor/. Returns
'-1' if monitor files can't be opened or if the channel is already
monitored,
otherwise '0'.
[Syntax]
Monitor([file_format[:urlbase]][,fname_base[,options]])
[Arguments]
file_format
optional, if not set, defaults to 'wav'
fname_base
if set, changes the filename used to the one specified.
options
m: when the recording ends mix the two leg files into one and delete
the two leg files. If the variable ${MONITOR_EXEC} is set, the
application
referenced in it will be executed instead of soxmix/sox and the raw leg
files will NOT be deleted automatically. soxmix/sox or ${MONITOR_EXEC}
is handed 3 arguments, the two leg files and a target mixed file name
which is the same as the leg file names only without the in/out
designator.
If ${MONITOR_EXEC_ARGS} is set, the contents will be passed on as
additional arguments to ${MONITOR_EXEC}. Both ${MONITOR_EXEC} and the
Mix flag can be set from the administrator interface.
b: Don't begin recording unless a call is bridged to another channel.
i: Skip recording of input stream (disables 'm' option).
o: Skip recording of output stream (disables 'm' option).
[See Also]
StopMonitor()
Thx Eric.
I read the link e1*CLI core show application monitor but I could not
follow what I should do to customize the file name of the recording.
I guess some changes to the dialplan is required ?
Thx
S
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[asterisk-users] Asterisk-cpu utilization 60 %
Hi, On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest. Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly 1-2 concurrent calls. No other activity on server. Top shows asterisk on top. Its quad xeon server with 4 gb ram. Any suggestion where should I start and how should I go about with my investigation. Thank you and have a great weekend. Sans -- _ -- 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] Error: Unable to create channel of type 'SIP'
Hi, I am using Trixbox 2.6.2.3, ISO install I am getting the below error in `/var/log/asterisk/full` Unable to create channel of type 'SIP' (cause 3 - No route to destination) Is there anyway to figure out which extension is causing this error ? Thank you. Best regards, Sanjay -- _ -- 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] CDR: MySQL query
Thx Rudi. but this query results in *Empty set (0.32 sec) src AND dst like number *seems to be the problem area. * * Also, how can I get the hold time talk time as separate values OR may be total call connect time talk time (the difference of the 2 will be hold time). Thx Sans On Wed, Aug 4, 2010 at 6:18 PM, Rudi Oosthuizen rudi.oosthui...@nha.co.zawrote: Mysql Use asteriskcdrdb; Select calldate,src,dst,disposition,duration,billsec,uniqueid from cdr where src like 'NUMBER' and dst like 'NUMBER' order by calldate; Rudi Oosthuzen Hi, Can someone help me formulate MySQL Query(s) which will help me extract the following details for a given DID (date range can be excluded for simplicity). Date-Time DNID (I am recording this is `userfield`) CLID time-1 (when call was received) time-2 (when call was answered by agent) time-3 (when call was hung-up) My Call flow is as follows: - Caller dials a DNID - Call enters queue - Call rings in round-robin format to all logged in agents - Agent answers call - Both parties hand-up Any help with MySQL queries or pointers are deeply appreciated. Thx Sans -- _ -- 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] CDR: MySQL query
Hi, Can someone help me formulate MySQL Query(s) which will help me extract the following details for a given DID (date range can be excluded for simplicity). Date-Time DNID (I am recording this is `userfield`) CLID time-1 (when call was received) time-2 (when call was answered by agent) time-3 (when call was hung-up) My Call flow is as follows: - Caller dials a DNID - Call enters queue - Call rings in round-robin format to all logged in agents - Agent answers call - Both parties hand-up Any help with MySQL queries or pointers are deeply appreciated. Thx Sans -- _ -- 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] browser pop-up on call ring
Hi, I am looking for a Windows Desktop based application which will open a web browser with the below url upon CALL RING on the softphone. *http://192.168.1.4:3100/popup.php?did=DNID* (where DNID is the called DID number) Let me know for any help!! Thank you. Best regards, Sanjay -- _ -- 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] Need USA DIDs
Hi, Looking for some reliable and quality providers of USA DIDs. Any pointers ? Thx Sans -- _ -- 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] Need USA DIDs
On Wed, Jun 23, 2010 at 9:50 PM, Hall, Rick r...@readywire.com wrote: Agreed! Didforsale.com is THE way to go. -- Rick Hall Senior Vice President ReadyWire Multimedia Solutions Anyone having experience with didww.com ? Sorry, I forgot to mention I am looking for wholesale DID -- reseller option with API to that my customers can select country - city -- DID from my website. Thx -- _ -- 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] Create Conference and exit myself
Hi, I am using Trixbox trixbox CE 2.6.2.3 (Stable) using Asterisk 1.4.22-4 I am looking for the following functionality: `` I receive a call from Mr. A. I put Mr. A on hold. I dial Mr. B I connect Mr. A's call (which was on hold) to Mr. B and I get out of the call. Mr. A Mr. B are in conversation, while my line is free to accept a new calls. What is the simplest way to achieve this ?? Thx Sans -- _ -- 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] CDR: Add Dialed Number Identifierfield (DNID)field into MySQL
I have read 2 solutions
(a) Changing the Dial plan and capturing DNID and inserting it into
one of the existing column in CDR table.
(b) Copy new CDR related .c .h files which have added the
functionality of recording DNID into MySQL.
For this, CDR table structure needs to be changed and a new field has
be created in CDR table.
But I am still not very sure on how to go about doing this.
Since I only have a production server, I do not have the options of
experimenting.
Can someone help with a step-by-step?
Thx
Sanjay
On Mon, Mar 15, 2010 at 3:08 PM, Lee Archer lee.arc...@thebigword.com
wrote:
Isn't the use of DNID separate to the userfield? I'd like to have this
working also.
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
Balashov
Sent: 15 March 2010 08:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield
(DNID) field into MySQL
Use the userfield.
On 03/15/2010 04:25 AM, RSCL Mumbai wrote:
Hi,
I would like to see the DNID in my MySQL CDR logs.
I have read one big thread in the Asterisk Developer List, but I could
not figure out how to implement it ?
Is there a simple step-by-step.
If this is Asterisk 1.6.*, then you can use the adaptive ODBC, which is
configured using /etc/asterisk/cdr_adaptive_odbc.conf. If you compiled
Asterisk with samples, you will find a sample file that has pretty much
everything that you need. From there, simply set the fieldname that you
wish to write to the CDR, like this:
; Using Adaptive ODBC CDR's, sets the caller ID DNID to the CDR's custom
field named DNID
Set(CDR(DNID)=${CALLERID(DNID)})
Personally, I like to set the DNID to a variable, just in case, when the
inbound call first hits Asterisk from the trunk. This probably isn't
necessary, but I am always afraid that the CALLERID(DNID) value will change
with a transfer or a channel redirect, which we use. From there I write the
variable to the CDR.
For more information on the adaptive concept, please see
http://www.asterisk.org/node/48492. There is also more detail from Tilghman
Lesher here:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg210573.html
It's very elegant in it's design and it works like a champ- we use it in
production.
If you are using Asterisk 1.4.*, you can use the the CDR's userfield. This
is an optional, user defined field that can store just about whatever data
you wish depending on the data type defined in the database. You will have
to google around to find out more information on how to enable it, although
I believe that it's an option in the /etc/asterisk/cdr.conf configuration
file that you are using.
Again, if you are using Asterisk 1.6.* I would strongly recommend that you
take advantage of the Adaptive CDR system.
I am using Asterisk 1.4.*
My cdr_mysql.conf has only the following:
[global]
hostname = localhost
dbname=asteriskcdrdb
password = amp109
user = asteriskuser
userfield=1
;port=3306
;sock=/tmp/mysql.sock
---
I could not much info on the net on this subject.
Thx
Sanjay
--
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Do we need an update to cdr_addon_mysql for this to work?
Lee
Still no headway.
Any help is appreciated.
Thx
Sanjay
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Re: [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL
I have read 2 solutions
(a) Changing the Dial plan and capturing DNID and inserting it into
one of the existing column in CDR table.
(b) Copy new CDR related .c .h files which have added the
functionality of recording DNID into MySQL.
For this, CDR table structure needs to be changed and a new field has
be created in CDR table.
But I am still not very sure on how to go about doing this.
Since I only have a production server, I do not have the options of
experimenting.
Can someone help with a step-by-step?
Thx
Sanjay
On Mon, Mar 15, 2010 at 3:08 PM, Lee Archer lee.arc...@thebigword.com
wrote:
Isn't the use of DNID separate to the userfield? I'd like to have this
working also.
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
Balashov
Sent: 15 March 2010 08:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield
(DNID) field into MySQL
Use the userfield.
On 03/15/2010 04:25 AM, RSCL Mumbai wrote:
Hi,
I would like to see the DNID in my MySQL CDR logs.
I have read one big thread in the Asterisk Developer List, but I could
not figure out how to implement it ?
Is there a simple step-by-step.
If this is Asterisk 1.6.*, then you can use the adaptive ODBC, which is
configured using /etc/asterisk/cdr_adaptive_odbc.conf. If you compiled
Asterisk with samples, you will find a sample file that has pretty much
everything that you need. From there, simply set the fieldname that you wish
to write to the CDR, like this:
; Using Adaptive ODBC CDR's, sets the caller ID DNID to the CDR's custom
field named DNID
Set(CDR(DNID)=${CALLERID(DNID)})
Personally, I like to set the DNID to a variable, just in case, when the
inbound call first hits Asterisk from the trunk. This probably isn't
necessary, but I am always afraid that the CALLERID(DNID) value will change
with a transfer or a channel redirect, which we use. From there I write the
variable to the CDR.
For more information on the adaptive concept, please see
http://www.asterisk.org/node/48492. There is also more detail from Tilghman
Lesher here:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg210573.html
It's very elegant in it's design and it works like a champ- we use it in
production.
If you are using Asterisk 1.4.*, you can use the the CDR's userfield. This is
an optional, user defined field that can store just about whatever data you
wish depending on the data type defined in the database. You will have to
google around to find out more information on how to enable it, although I
believe that it's an option in the /etc/asterisk/cdr.conf configuration file
that you are using.
Again, if you are using Asterisk 1.6.* I would strongly recommend that you
take advantage of the Adaptive CDR system.
I am using Asterisk 1.4.*
My cdr_mysql.conf has only the following:
[global]
hostname = localhost
dbname=asteriskcdrdb
password = amp109
user = asteriskuser
userfield=1
;port=3306
;sock=/tmp/mysql.sock
---
I could not much info on the net on this subject.
Thx
Sanjay
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[asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL
Hi, I would like to see the DNID in my MySQL CDR logs. I have read one big thread in the Asterisk Developer List, but I could not figure out how to implement it ? Is there a simple step-by-step. Thx in advance. Vai -- _ -- 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] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL
I have read 2 solutions (a) Changing the Dial plan and capturing DNID and inserting it into one of the existing column in CDR table. (b) Copy new CDR related .c .h files which have added the functionality of recording DNID into MySQL. For this, CDR table structure needs to be changed and a new field has be created in CDR table. But I am still not very sure on how to go about doing this. Since I only have a production server, I do not have the options of experimenting. Can someone help with a step-by-step? Thx Sanjay On Mon, Mar 15, 2010 at 3:08 PM, Lee Archer lee.arc...@thebigword.com wrote: Isn't the use of DNID separate to the userfield? I'd like to have this working also. Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: 15 March 2010 08:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL Use the userfield. On 03/15/2010 04:25 AM, RSCL Mumbai wrote: Hi, I would like to see the DNID in my MySQL CDR logs. I have read one big thread in the Asterisk Developer List, but I could not figure out how to implement it ? Is there a simple step-by-step. Thx in advance. Vai -- 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 -- _ -- 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] Queues with unavailable members
What is the command to log off the agents ? Thx On Wed, Oct 14, 2009 at 6:45 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: You could configure them as agents and have them log off automatically after a while they're not responding. l. 2009/10/14 Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk We have the possibly rather unique setup where we have cell phones posing as SIP devices. The SIP registration for those unfortunately doesn't go away just because the phone is off, since the registration is done by our cell-phone=SIP gateway, and that gateway has no way of knowing whether the phone is on or off. This is usually ok, but it gets problematic if the cell phone is a member of a queue. In that case Queue() keeps sending the call to the phone, and the cell-phone=SIP gateway dutifully makes a call, which is then rejected by the cellular network. A few seconds later, Queue() tries again. This needlessly wastes resources both in Asterisk and in the cellular network. One idea is to run the call through chan_local (we do this anyway because we need to format the caller-ID differently for different phones) and then record if a call is rejected, and for the next 30 seconds just abort if we are asked to send a call to that particular phone. The downside is that we are still running a call through part of the dial plan, but at least it can be done in perhaps 3 lines of code. I would very much like to hear about smarter ways to do it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mp3 for IVR prompts
On Sat, Oct 10, 2009 at 7:59 PM, Steve Edwards asterisk@sedwards.comwrote: On Sat, 10 Oct 2009, gergis.rasmy wrote: can i use MP3 files as an IVR prompts directly without converting to .gsm format? You don't want to do this. Asterisk will attempt to use prompts encoded with the same codec being used for the channel. So, unless you have a channel that is using MP3, Asterisk would have to transcode the prompt every time it is used. Why would you want to burn CPU cycles for this useless activity? You should strive to have prompts available in all the channel encodings actually used by your system. I have systems that only use ULAW, so all of my prompts are encoded as ULAW. (Sometimes I cheat and use WAV files since they are easier to work with and transcoding from WAV to ULAW is cheap.) How should I convert my .wav prompts into aLaw, uLaw, G729 ? Thx Vai ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mp3 for IVR prompts
On Sat, Oct 10, 2009 at 11:47 PM, Steve Edwards asterisk@sedwards.comwrote: On Sat, 10 Oct 2009, RSCL Mumbai wrote: How should I convert my .wav prompts into aLaw, uLaw, G729 ? The standard Asterisk prompts are already available in a wide variety of encodings. Try googling for asterisk convert mp3 to wav Some will suggest to use Asterisk. Besides appearing to use a sledgehammer for a fly-swatter, I don't like using a mission critical resource when there are better alternatives like sox -- especially for scripting. g729 will be a bit of a bitch, however. I cobbled up a script to use mpg123 (to convert from MP3 to WAV), sox, and normalize. My IVR prompts are in .WAV format. Refering to your previous post : ` You should strive to have prompts available in all the channel encodings actually used by your system. I have systems that only use ULAW, so all of my prompts are encoded as ULAW. (Sometimes I cheat and use WAV files since they are easier to work with and transcoding from WAV to ULAW is cheap.) ` Can I convert my .WAV IVR greetings, MOH and other recordings into G729 format to prevent transcoding and hence CPU usage ? Thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best QoS for Linux
On Fri, Oct 9, 2009 at 2:18 AM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: On Thu, 2009-10-08 at 16:07 -0400, Michelle Dupuis wrote: More specificallyI'm looking for a Linux package to allow shaping, QoS, prioritization by port, etc. snip Spinning off from another topic...what are people using for QoS / Shaping? I'm using Wondershaper script with OK results...but I'd like better. Ideas? _snip I would imagine that tc, iproute2, and iptables are your friends. In our case, we try to keep things as simple as possible in a fairly complex environment. Thus, whenever we can, we try to set our DSCP/ToS bits in a way that will be handled properly by the default Linux queueing mechanism. I'm afraid I'm up to my eyeballs in a project right now but I have posted some of our work in earlier posts on this mailing list. In the case of Asterisk, we use b0 instead of b8 (expedited forwarding) for RTP traffic because it works better with the default pfifo_fast packet scheduler. We've also ensured the packet handling is consistent from end to end as much as possible. Even though we are using the Internet as a transport medium, we're very happy so far with the quality of the calls. See the previous posts for more details. Hope this helps - John -- John A. Sullivan III We were thinking on similar lines a while back and decide to implement Packet Prioritization. VoIP packets to have highest priority as compared to all other packets. I believe tc, iproute2, and iptables was to be used; thou I am not very sure. Due to lack of time, we could not do this, but its still on my ToDo list. hth, Sanjay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Peers Listed in sip show channels
Hi, I am using Trxibox 2.6 latest ISO install. Following is the output of : sip show channels [trixbox ~]# /usr/sbin/asterisk -rx sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 212.53.40.40 0218245 6cfb845d050 09011/0 0x0 (nothing)No 192.168.1.116(None) YTc4ZmM3NjV 00101/6 0x0 (nothing) No Rx: REGISTER 195.189.173.10 301241893b37329b407 18996/0 0x0 (nothing)No 192.168.1.13 10072da66c6d6a1 00102/0 0x280100 (g729| No Tx: ACK 192.168.1.13 100567384261131 00102/0 0x280100 (g729| No Tx: ACK 192.168.1.13 1010041c9a77455 00102/0 0x280100 (g729| No Tx: ACK 81.201.84.45 3473290576 PUM273-UMU5 00101/00102 0x100 (g729) No Rx: ACK 81.201.84.45 2706513184 ISB67X-ZJQN 00101/00102 0x100 (g729) No Rx: ACK 81.201.84.45 4023308836 G7JP5O-AA4J 00101/00102 0x100 (g729) No Rx: ACK 192.168.1.13 10160758ea9a349 00102/0 0x0 (nothing)No (d) Tx: ACK 122.169.113.145 1006379b29497d0 00102/0 0x0 (nothing) No Init: NOTIFY 122.169.113.145 10063a4fc558695 00102/0 0x0 (nothing) No Init: NOTIFY 122.169.113.145 10067678828b011 00102/0 0x0 (nothing) No Init: NOTIFY 13 active SIP channels The last 3 rows have been there since past 6 days. There is no user 1006, logged into the system... I have 2 questions: (1) Where does Trixbox store this information (2) How can I periodically remove these records Thx in advance. Sanjay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peers Listed in sip show channels
do you have that user 1006 defined by IP ? *I have a user 1006. Its not defined by IP. * does it have mailbox= also defined ? *Yes. 1006 has a Mail box*. my wild guess is that there's unchecked voicemail and asterisk tries to initialize sending NOTIFY MWI messages *I will delete all messages from the Mailbox and see if 1006 is removed from the listing. * you can't remove these messages they remove themselves after some timeout *Any idea where there are 3 rows with 1006*? Thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peers Listed in sip show channels
my wild guess is that there's unchecked voicemail and asterisk tries to initialize sending NOTIFY MWI messages *I will delete all messages from the Mailbox and see if 1006 is removed from the listing.* Just checked, no messages in 1006. Any other reasons! Thx Sanjay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to remove peers from channels
Pls see below output. I would like to remove the last 3 peers. How can I do this ? Thx Vai [trixbox ~]# /usr/sbin/asterisk -rx sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 192.168.1.126(None) MjkzYjNiMmY 00101/4 0x0 (nothing) No Rx: REGISTER 64.154.41.1067552235573 147111b67e3 11342/0 0x0 (nothing)No 81.201.84.45 3866719789 3TUNX3-CPYZ 00101/00102 0x100 (g729) No Rx: ACK 195.189.173.10 301241893b37329b407 19037/0 0x0 (nothing)No 81.201.84.45 4172802551 NC75LK-XAU5 00101/00102 0x100 (g729) No Rx: ACK 192.168.1.13 10100cb8ef0d570 00102/0 0x280100 (g729| No Tx: ACK 192.168.1.13 10053cc07973759 00102/0 0x280100 (g729| No Tx: ACK 81.201.84.45 7709498956 ECSTS5-MU5R 00101/00102 0x100 (g729) No Rx: ACK 81.201.84.45 9147616530 VTTE3C-CN2Z 00101/00102 0x100 (g729) No Rx: ACK 81.201.84.45 9414471279 GC2W4P-ZPWN 00101/00102 0x100 (g729) No Rx: ACK 192.168.1.13 1007080f5e47519 00102/0 0x280100 (g729| No Tx: ACK 81.201.84.45 9858786358 CQQ5M7-ZM4F 00101/00102 0x100 (g729) No Rx: ACK 81.201.84.45 3189496064 FDK6CY-2LSF 00101/00102 0x100 (g729) No Rx: ACK 192.168.1.13 10160758ea9a349 00102/0 0x0 (nothing)No (d) Tx: ACK 122.169.113.145 1006379b29497d0 00102/0 0x0 (nothing) No Init: NOTIFY 122.169.113.145 10063a4fc558695 00102/0 0x0 (nothing) No Init: NOTIFY 122.169.113.145 10067678828b011 00102/0 0x0 (nothing) No Init: NOTIFY 17 active SIP channels ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to remove peers from channels
On Fri, Sep 25, 2009 at 10:27 PM, Philipp Kempgen philipp.kemp...@amooma.de wrote: RSCL Mumbai schrieb: Pls see below output. I would like to remove the last 3 peers. How can I do this ? [trixbox ~]# /usr/sbin/asterisk -rx sip show channels Use grep. (See `man grep`.) I may not have explained my requirement well. I do not wish to remove the peers from the listing. I want the peers to not be there at all. These peers (EyeBeam extensions) had connected to the Trixbox about 24+ hours ago. At this moment, I do not have anyone connected to my Trixbox server from IP: 122.169. using extension 1006. But I see 3 peers showing as connected from 122.169. using extension 1006. Thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
