[asterisk-users] Asterisk call forward for T1 incoming calls
Is there a way to divert incoming calls on DAHDI T1 channels so telco gets the diversion and send the call to new number and releasing the channel? -- _ -- 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] handset forwarding Diversion header cannot be set on Local channels
is there anyway to change Sip headers in local channels? if a user sets forward on their handset, calls coming in to the handset get diversion header added: Diversion: 202 sip:202@192.168.1.46;reason=deflection Then asterisk sends the call to local channel: - Now forwarding SIP/201-0483 to 'Local/33@test' (thanks to SIP/202-0484) and not all Telco providers handle diversion header gracefully, some dont like to see 202 in header. i tried to set the sip header in target 33@test but asterisk see's this as local channel and wont do sip add header: WARNING[13584]: chan_sip.c:20562 func_header_read: This function can only be used on SIP channels. is there anyway around this? -- _ -- 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] AppVoicemail overwrites voicemail.conf
yes, thanks you! On Sat, Mar 22, 2014 at 9:13 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Mar 21, 2014 at 11:58 PM, Al lists asteris...@gmail.com wrote: looking more into this, looks like this is not a issue, its related to users changing voicemail password from handset, asterisk rewrites the file. Right, use passwordlocation = spool, create a secret.conf for each mailbox, now when a user changes their password, secret.conf gets updated not voicemail.conf. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] AppVoicemail overwrites voicemail.conf
We noticed issues with voicemail and somehow looks like voicemail.conf has been overwritten: ;! ;! Automatically generated configuration file ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf) ;! Generator: AppVoicemail ;! Creation Date: Thu Mar 20 06:48:16 2014 ;! i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are not using realtime. anyway to prevent AppVoicemail ro auto generate files? -- _ -- 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] AppVoicemail overwrites voicemail.conf
passwordlocatio seems to be related to vmsecret from voicemail.conf sample : ; passwordlocation=spooldir ; Usually the voicemail password (vmsecret) is stored in ; this configuration file. By setting this option you can ; specify where Asterisk should read/write the vmsecret. ; Supported options: ; voicemail.conf: ; This is the default option. The secret is read from ; and written to voicemail.conf (or users.conf). ; spooldir: ; The secret is stored in a separate file in the user's ; voicemail spool directory in a file named secret.conf. ; Please ensure that normal Linux users are not ; permitted to access Asterisk's spool directory as the ; secret is stored in plain text. If a secret is not ; found in this directory, the password in ; voicemail.conf (or users.conf) will be used. ; Note that this option does not affect password storage for ; realtime users, which are still stored in the realtime ; backend. but the issue i was explaining was voicemail.conf getting overwritten apparently by appvoicemail On Fri, Mar 21, 2014 at 5:36 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Mar 21, 2014 at 3:22 PM, Al lists asteris...@gmail.com wrote: We noticed issues with voicemail and somehow looks like voicemail.conf has been overwritten: ;! ;! Automatically generated configuration file ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf) ;! Generator: AppVoicemail ;! Creation Date: Thu Mar 20 06:48:16 2014 ;! i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are not using realtime. anyway to prevent AppVoicemail ro auto generate files? passwordlocation = spooldir Read voicemail.conf about how to use it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] AppVoicemail overwrites voicemail.conf
looking more into this, looks like this is not a issue, its related to users changing voicemail password from handset, asterisk rewrites the file. On Fri, Mar 21, 2014 at 9:31 PM, Al lists asteris...@gmail.com wrote: passwordlocatio seems to be related to vmsecret from voicemail.conf sample : ; passwordlocation=spooldir ; Usually the voicemail password (vmsecret) is stored in ; this configuration file. By setting this option you can ; specify where Asterisk should read/write the vmsecret. ; Supported options: ; voicemail.conf: ; This is the default option. The secret is read from ; and written to voicemail.conf (or users.conf). ; spooldir: ; The secret is stored in a separate file in the user's ; voicemail spool directory in a file named secret.conf. ; Please ensure that normal Linux users are not ; permitted to access Asterisk's spool directory as the ; secret is stored in plain text. If a secret is not ; found in this directory, the password in ; voicemail.conf (or users.conf) will be used. ; Note that this option does not affect password storage for ; realtime users, which are still stored in the realtime ; backend. but the issue i was explaining was voicemail.conf getting overwritten apparently by appvoicemail On Fri, Mar 21, 2014 at 5:36 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Mar 21, 2014 at 3:22 PM, Al lists asteris...@gmail.com wrote: We noticed issues with voicemail and somehow looks like voicemail.conf has been overwritten: ;! ;! Automatically generated configuration file ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf) ;! Generator: AppVoicemail ;! Creation Date: Thu Mar 20 06:48:16 2014 ;! i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are not using realtime. anyway to prevent AppVoicemail ro auto generate files? passwordlocation = spooldir Read voicemail.conf about how to use it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] is this expected behaviour?
i noticed in asterisk 10.12.3, i get messages like this: [2014-01-08 19:03:59] NOTICE[2987]: chan_sip.c:23900 handle_request_invite: Failed to authenticate device 305sip:3...@my.server.ip;tag=0d516e63 but not mentioning attacker ip (to be used for fail2ban) is this expected? -- _ -- 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] Function DB_KEYS()
Hi All, Anyone know how to use the function DB_KEYS()? Info on this is non-existant on the net incl. the wiki and there are absolutely NO examples of it anywhere. I was hoping that unlike the other DB functions, this is able to get the Key for a given Value OR at least list ALL keys of a given Family Tree through which we can maybe iterate and get the values of each key etc. Speaking of which, it WOULD be quite cool if there was a function that could do as above, i.e. find the key(s) if instead of a value lookup for a given key, a key was returned for a given value or pattern of a known value Thx \a -- _ -- 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] Function DB_KEYS()
Ok, nevermind. Got it! Does at least one of the things I needed. Now would be great to have a function that does the opposite ...and yes, I do know about func_odbc, my current need just isn't big enough to setup a local MySQL/PostGreSQL etcwas hoping to get this out of the built in DB. I guess the next step is to maybe use AGI On Mon, Jan 21, 2013 at 5:10 PM, Al Efron [gmail] all.efor...@gmail.comwrote: Hi All, Anyone know how to use the function DB_KEYS()? Info on this is non-existant on the net incl. the wiki and there are absolutely NO examples of it anywhere. I was hoping that unlike the other DB functions, this is able to get the Key for a given Value OR at least list ALL keys of a given Family Tree through which we can maybe iterate and get the values of each key etc. Speaking of which, it WOULD be quite cool if there was a function that could do as above, i.e. find the key(s) if instead of a value lookup for a given key, a key was returned for a given value or pattern of a known value Thx \a -- _ -- 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] Web based Click to Call Application
Hi, Here is a starting point (WebRTC): https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support Regards. // Binan. Från: akhilesh chand omakhileshch...@gmail.com Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Skickat: fredag, 9 november 2012 11:32 Ämne: [asterisk-users] Web based Click to Call Application Dear All, I want to develop click to call(C2C) web based application.Is there any study material. I will really appreciate your help, thank you. Regards Akhilesh -- _ -- 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] monitor application, file name change on attended transfer
Hi,
You are using b flag in monitor command. This means don't begin recording
untill call is bridged.
So what you get if you delete this flag ?
// Binan
Från: Grzegorz Pycia grzegorz.py...@thulium.pl
Till: asterisk-users@lists.digium.com
Skickat: lördag, 20 oktober 2012 23:49
Ämne: [asterisk-users] monitor application, file name change on attended
transfer
Hi
I have some problem with monitor application when call i transferred in
attended mode and the transfer occurs before call is answered.
Here is how it looks:
A calls B(let's assume ${UNIQUEUEID}=1)
exten = _,1,NoOp
seme = n,Set(MONITOR_FILENAME=call-${UNIQUEID})
same = n,monitor(alaw,/var/spool/asterisk/monitor/${MONITOR_FILENAME},bm)
When B answers the call, files call-1-in* and call1-out* are created. During
The call, B tries to make attended transfer A is put on hold and B calls C
using the same dialplan logic:
B calls C(let's assume ${UNIQUEUEID}=2)
At the time off invoking monitor application none off the call-2 channels are
monitored so the monitor application starts without errors, if B waits till C
answers, everything is OK monitor starts recording and files call-2-in* and
call-2-out* are created, When B transfers the call call-2 monitor is stopped.
And call-2 files contain only the call between B and C.
But there is problem when B does not wait until C answers the call, if transfer
is done before C answers the call, the call-2* are not created and the call is
still recorded to the call-1* files, but when the transferred call between A
and C ends, the call-1* files get renamed to call-2* and the MONITOR_EXEC
application is called with call-2* file names as parameters.
This makes it impossible to locate the call record since the file names get
changed, can someone tell if I should file a BUG report or is it intended to
act like this?
Regards
--
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Re: [asterisk-users] Sound problem with format files but not codecs
Hello, It means that one of clients, is using 'silence suppression' mechanism which sends audio frames that do not contain any samples. Asterisk complains about silence supression and appears these warnings on CLI. If the client turn off the silence suppression the message will disappear. // Binan. Från: Administrator TOOTAI ad...@tootai.net Till: Asterisk-Users asterisk-users@lists.digium.com Skickat: söndag, 21 oktober 2012 10:34 Ämne: [asterisk-users] Sound problem with format files but not codecs Hi all, on asterisk 1.8.16 [2012-10-20 19:36:17] VERBOSE[743] pbx.c: -- Executing [801@OFFICE-Numbers:2] MusicOnHold(Local/801@OFFICE-Numbers-e54a;2, ) in new stack [2012-10-20 19:36:17] VERBOSE[743] res_musiconhold.c: -- Started music on hold, class 'TOOTAi', on Local/801@OFFICE-Numbers-e54a;2 [2012-10-20 19:36:17] WARNING[742] translate.c: no samples for ulawtolin [2012-10-20 19:36:21] VERBOSE[742] pbx.c: == Spawn extension (from_to-OFFICE, 801, 23) exited non-zero on 'SIP/8081773619-2f28' [2012-10-20 19:36:21] VERBOSE[743] res_musiconhold.c: -- Stopped music on hold on Local/801@OFFICE-Numbers-e54a;2 or asterisk 10.8.0 -- Executing [801@macro-GeneralNumbers:1] Set(SIP/105-0081, CHANNEL(musicclass)=TOOTAi) in new stack -- Executing [801@macro-GeneralNumbers:2] MusicOnHold(SIP/105-0081, ) in new stack -- Started music on hold, class 'TOOTAi', on SIP/105-0081 [2012-10-20 22:48:48] WARNING[22435]: translate.c:343 framein: no samples for g722tolin -- Stopped music on hold on SIP/105-0081 This is when calling extension: exten=801,1,Set(CHANNEL(musicclass)=TOOTAi) exten=801,n,MusicOnHold() exten=801,n,Hangup What does mean those WARNINGS and how to solve this problem? MeetMe, Voicemail or holding a call are working fine. From what I understand, codecs are used in channels and format for handling files. In both cases, two different servers, asterisk is compiled from tar.gz and in menuselect all codecs and formats are activated. Is this a bug? Did I forget something? On a third server I run latest Elastix with an asterisk 1.8.16 version. On this server I have no MusicOnHold at all even during calls. Logs show VERBOSE[19717] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/104-00b3 VERBOSE[19717] res_musiconhold.c: -- Stopped music on hold on SIP/104-00b3 which is MusicOnHold stop immediately. On all servers wav files are installed, even try with original ones delivered with Asterisk. Thanks for any hint Regards -- Daniel -- _ -- 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] SIP Register DOS attack
I'll check this option and see if it helps next time, just to clarify, there were no actual calls in place, just DOS register attack. On Wed, Jun 1, 2011 at 12:22 PM, Ira i...@extrasensory.com wrote: At 10:56 AM 6/1/2011, you wrote: Do you have: sip.conf [general] allowguest=no So because of this I decided to type sip show channels into my Asterisk and got this: Peer User/ANRCall ID Format Hold Last Message Expiry Peer 216.xxx.69.xxx (None) f2d8db55-0a7edd (nothing) NoRx: OPTIONS guest 216.xxx.69.xxx (None) 2ce0b9a5-6de7f4 (nothing) NoRx: OPTIONS guest 64.xxx.41.xxx6314098389 2a482e4b684a59a (nothing) No guest 192.168.233.xxx (None) ioh3fna2aw.n4mz (nothing) NoRx: REGISTER guest 4 active SIP dialogs I have allowguest=no and all of those IPs are either my providers or a SIP phone on my network so why would it show guest as the peer? I'm running Asterisk SVN-trunk-r319759M if that matters. Ira -- _ -- 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] SIP Register DOS attack
Hi List Recently i have noticed this attack on couple of servers, usually a foreign IP starts sending tons of register request without any answer to authentication, if you type sip show channels in cli you will see tons of these: 1.2.3.4 (None) 2389603298 00101/1 0x0 (nothing)No Rx: REGISTER since there is no authentication in place, asterisk does not see any failed register attempt, so there wont be anything added to log file as failed attempt. thus fail2ban wont see any activity and wont block the IP. it simply brings down the internet link and the box due to too many sip channels. -- _ -- 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] Echo on Sangoma A400 and background noise
I'm a long time user of Digium carts and stupid me i wanted to give Sangoma a try. We got Sangoma A400 with 6 FXO ports. Asterisk version: 1.4.35 Zaptel version: 1.4.11 Wanpipe version: 3.5.11 we tried to use fxtune but looks like it wont work with Sangoma card, ( please correct me if i'm wrong) Echo is really bad and also we have background noise on all lines. We tried both mg2 and oslec echo canceler. was wondering if you have any experiense with that because Sangoma tech support is not helpfull, just look at their response: As you mentioned you have tried Oslec algorithms for echo cancellation.Which is a good way to solve echo cancellation issues. If that is not woking for you you may want to upgrade to hardware echo cancellation..with cards which have echo cancellers. Hope this helps. -Sri -- _ -- 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] Dahdi PRI T1 Setup for TE210P
Hello, I have been struggling with the configuration of this card on my box. I have a T1 line and I am trying to setup asterisk with it. I followed all the instructions and I still see a blinking red light on the card. I use fedora 12. If everything is fine should I see a green line when I plug in the T1 line ? I want to isolate the issue so I di not start asterisk. When I run asterisk I get red alarms on all the channels. Thanks for any help. I have sent 2 days on this and this is what I did: 1. I installed the card. 2. libpri-1.4.11 - dahdi-linux-complete-2.3.0+2.3.0 - dahdi-tools-2.3.0 3. dahdi_hardware: pci::01:06.0 wct4xxp+ d161:0210 Wildcard TE210P 4. cat /proc/dahdi/1 :cat /proc/dahdi/1 Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 B8ZS/ESF BLUE RED 1 TE2/0/1/1 Clear RED 2 TE2/0/1/2 Clear RED 3 TE2/0/1/3 Clear RED 4 TE2/0/1/4 Clear RED 5 TE2/0/1/5 Clear RED 6 TE2/0/1/6 Clear RED 7 TE2/0/1/7 Clear RED 8 TE2/0/1/8 Clear RED 9 TE2/0/1/9 Clear RED 10 TE2/0/1/10 Clear RED 11 TE2/0/1/11 Clear RED 12 TE2/0/1/12 Clear RED 13 TE2/0/1/13 Clear RED 14 TE2/0/1/14 Clear RED 15 TE2/0/1/15 Clear RED 16 TE2/0/1/16 Clear RED 17 TE2/0/1/17 Clear RED 18 TE2/0/1/18 Clear RED 19 TE2/0/1/19 Clear RED 20 TE2/0/1/20 Clear RED 21 TE2/0/1/21 Clear RED 22 TE2/0/1/22 Clear RED 23 TE2/0/1/23 Clear RED 24 TE2/0/1/24 HDLCFCS RED 5. dahdi_scan [1] active=yes alarms=BLU/RED description=T2XXP (PCI) Card 0 Span 1 name=TE2/0/1 manufacturer=Digium devicetype=Wildcard TE210P location=Board ID Switch 0 basechan=1 totchans=24 irq=22 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [2] active=yes alarms=RED description=T2XXP (PCI) Card 0 Span 2 name=TE2/0/2 manufacturer=Digium devicetype=Wildcard TE210P location=Board ID Switch 0 basechan=25 totchans=24 irq=22 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF asterisk-users@lists.digium.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
On 7/30/09, Steve Totaro stot...@asteriskhelpdesk.com wrote: The first time is always free :) On Thu, Jul 30, 2009 at 1:50 PM, John Todd jt...@digium.com wrote: I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium’s web store and purchase (for zero dollars) the Skype For Asterisk product. The web store does require a Digium.com account, which can be set up during the purchase process if you don’t already have one. Once the web store process is complete, you will be e-mailed your license key and directions on where to download Skype For Asterisk beta software. This is a time-expiring beta - the software will stop working on August 31. The download is also currently time-limited - it will be available until August 7 on our website. After the 31st, you would need to have purchased a license for the SfA software (sorry, no pricing that I can give you right now - that will be a separate announcement. I'm just the community guy - I have no idea about pricing or commercial contracts or the like, so please wait until that's been announced as I will find out about the same time as you do. :-) Trial purchase page: http://store.digium.com/productview.php?product_code=804-00019 JT --- John Todd email:jt...@digium.comemail%3ajt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Load balancing Asterisk.
Foundry serverIron does support SIP and its ASIC not a linux box Load balancer like F5, Refer to Chapter 10 (page 677) of ServerIron manual. It explains everything in detail. Also you may need to play with source nat a little bit to make your specific configuration work, but it should work, at least in theory. On Thu, Nov 20, 2008 at 10:25 AM, Alex Balashov abalas...@evaristesys.comwrote: SIP wrote: As for the current F5 SIP load balancer, we tried it a few years back and it was a dismal failure. It wanted to do cookie-based SIP load balancing and only worked with certain SIP proxies. I assume that is because there is no way RFC-supported way to insert a cookie into a SIP session that persists throughout the entire exchange with a client, including all in-dialog requests, subsequent sessions, etc? The only way I know of to make a cookie stick on the UAC side is to put an LR parameter into the route set, but that will only last within a dialog. So, I'm assuming certain SIP proxies had proprietary ways of getting around that in order to work with F5? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP security
yes, make sure context line in general area has a dummy context, something with one line to hangup. On Fri, Nov 28, 2008 at 12:56 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Fri, Nov 28, 2008 at 11:00 AM, Mike l...@virtutel.ca wrote: I was looking at my CLI the other day, and found a lot of those types of messages: NOTICE[2242]: chan_sip.c:14383 handle_request_invite: Call from '' to extension '000452555169' rejected because extension not found. Looking at the IP, it originated from Asia and was clearly an attempt to screw with my Asterisk server. My quick fix was simply to block the IP adress at the firewall level. So that was the end of that. What I don`t get is how the person got that far. How could he attempt to dial extensions (even though he probably was in the default context which has nothing in it) when all my SIP peers are either password protected or linked to a fixed IP. And, more to the point, Call from `` means a call from what exactly? It's not one of my phones, it's not one of my peers…..Shouldn't the lack of a peer be enough to block the would-be hacker from tyring extensions? Any help is appreciate, I clearly don't understand SIP peers. Mike I think if you remove context from the [general] section, you would not see these messages. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] database queries from extensions.conf
Klaus Darilion wrote: Wolfgang Pichler schrieb: Hi, you yould also use DBQuery (does only support mysql) - take a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+DBQuery (it does also contain a cdr backend to write customzied cdr entries to the database) hi wolfgang! Have you programmed this yourself? Do you know how it compares to MYSQL function and func_odbc? regards klaus regards, Wolfgang Klaus Darilion schrieb: Hi! What is the preferred way to make database lookups from within the dialplan? I only know the MYSQL function from asterisk-addons. Are the other methods too? (e.g. for postgresql, unixodbc) thanks klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users func_odb only allows a SINGLE database statement Ergo you cannot do Transactions or Multi-statement SQL It is a MAJOR Backstep in DB Access. The MYSQL add-on is the BEST way to access DB from Dial Plan Digium should support and ADD to this rather than non putting a SINGLE mention of it in the last book and making no mention of it at Astricon. With this Add-on, and if DIGIUM would fix the brain dead implement ion of REAL-TIME for Exstensions.conf, things would/could be Soo Sweet. [ Can I get a Amen for having LABELS for steps in exstensions.conf when it is in Real Time ? Why the Heck do I have to use Different Format for Applications in exstensions.conf ] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] database queries from extensions.conf
Klaus Darilion wrote: Hi Jared! Thanks for the info - looks very flexible - you only have to edit 4 configuration files for a simple query :-) just a few questions: The ODBC library is unixodbc? How does it compare to the other solutions in terms of performance? e.g. (I have to make several queries for each call (caller preferences, LNP, LCR...) regards klaus Jared Smith schrieb: On Thu, 2008-11-13 at 15:16 +0100, Klaus Darilion wrote: What is the preferred way to make database lookups from within the dialplan? The preferred method is to use func_odbc, which takes SQL queries and builds custom dialplan functions from them. I've used it quite a bit, and am very happy with it. I also presented on func_odbc at AstriCon, and you can download my presentation: http://www.astricon.net/2008/glendale/web/presentations/DatabaseDriven_JSmith.pdf Quote The preferred method is to use func_odbc, which takes SQL queries and builds custom dialplan functions from them. I've used it quite a bit, and am very happy with it. How can you be VERY HappY with something that allows ONLY single statemts of SQL, ipso-facto you CANNOT do Begin Transaction SQL Statement SQL Statement End Transaction This isSOOO much more limited that the MYSQl add-on ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Dan Austin wrote: Yehavi wrote: Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Sam Houston University migrated from a Cisco CallManager and Nortel setup to Asterisk a couple years back. I do not know any of the specific details, but maybe you can track down someone involved in the project. Dan Remember - You are going from a CARRIER GRADE purpose built piece of hardware with Software built under a rigid CMM with extensive soak-testing to software that has been developed under , shall we say, a somewhat less rigid and stringent methodology. You will be moving from an environment supported by hundreds of highly trained people, some with decades of TELCO experience to one where you support comes from a somewhat less seasoned group of individuals. 10,000 extensions ??? On Asterisk ??? You pays your money, you takes you chances. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 ???
Brendan Martens wrote: On Oct 6, 2008, at 3:52 PM, Gordon Henderson wrote: The answer you are looking for is that you should be using a supported, stable version, and right now, 1.4 is the only one that fits. If I were starting today, I'd go with 1.4. 1.6.0 has just been released. Personally I'd start with that because then you don't stuck with generation old features, and as you are just starting you aren't locked into any feature sets or syntax issues, etc. Of course as it has just been released there are undoubtedly some bugs yet to be discovered, 1.4 has been around a while and will probably be easier to find support/documentation for. Quote are undoubtedly some bugs yet to be discovered Good laugh, look at the BUG reporting site. 1.4 had how many HUNDREDS of bugs reported ? How many more continue to flow in ? and you think someone should go to 1.6 ? May want to reconsider that. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Improving the voice Quality,
USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Asterisk REFER
is this a feature in asterisk? On Mon, Sep 15, 2008 at 3:20 AM, Patrick Maartense [EMAIL PROTECTED]wrote: Ice is the feature you're looking for I think If two clients support ice, a direct link between them will be made -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Al lists *Sent:* Dienstag, 09. September 2008 23:40 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Asterisk REFER Hi All, from what i'm understanding, Asterisk is back to back user agent. Base on this my initial thought was even if we enable reinvite in sip.conf, asterisk still will be in sip path after transfer. But i read some information in asterisk using refer to transfer a call completely to another sip or per say, a call comes in from voip provider and get transferred by asterisk to a cell phone number by using same provider and then asterisk will not be in SIP path anymore. is it doable ? No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.6.19/1661 - Release Date: 09.09.2008 04:58 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle
Indirectly it comes from DIGIUMS Very Strong Advice not to put more than 1 QUAD T1 card in a * box regardless of the size/power/configuration of the box. As TDM is generally one of the more straight forward and widely used protocols for VOICE, it is not totally unreasonable the logical conclusions one could draw from that limitation. Mind you, those conclusions are not necessarily Valid, but, in absence of Standardized Work Load Metrics for provisioning it is sort of a amorphous gray area Eric ManxPower Wieling wrote: Where did you hear this? Shaun Wingrin wrote: I have heard it said that, Asterisk falls over at 100 simultaneous calls. Is this true? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle
There is a LOT of dimensioning info out there. Mostly really old, based on anecdotal info as opposed to solid, standardized metric testing such as TP-C, TP-B, TP-C type work. Dimensioning is one area that appearers really lacking on Asterisk that you do have on brand Name VOIP. Gordon Henderson wrote: On Tue, 16 Sep 2008, Shaun Wingrin wrote: I have heard it said that, Asterisk falls over at 100 simultaneous calls. Is this true? No. google asterisk dimensioning Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Asterisk REFER
Hi All,from what i'm understanding, Asterisk is back to back user agent. Base on this my initial thought was even if we enable reinvite in sip.conf, asterisk still will be in sip path after transfer. But i read some information in asterisk using refer to transfer a call completely to another sip or per say, a call comes in from voip provider and get transferred by asterisk to a cell phone number by using same provider and then asterisk will not be in SIP path anymore. is it doable ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Asterisk IVR Scalability
DO NOT put all your eggs in one basket i.e. all you calls on ONE BOX Sriram wrote: Hi My Scenario is to implement Asterisk in a Call center.. I;ve TE420 Digium card and plan to terminate 4 PRIs (E1) on it. I;ve 30 Agents inside..Since its a PRI i m not using any hardware echo cancellation module.The calls would first land on Asterisk and depending on the options would be transferred to the Agent. I've read lot of opinions on voip-info.org giving asterisk hardware dimensions. I would like to take a final call depending on your expert answers : Scenario : There would be 120 calls for sure during a 2 hour period of a day , rest of times it would be serving max. 50 calls. No matter how many calls come only 30 would be able to talk to agents rest would be listening to some files on the IVR or be involved in some polling..This is what the client wants as of now but he needs a scalable solution depending on traffic.. Queries : 1. For this initial setup Will a Dual processor (Xeon) with 2 Gigs of RAM with TE420 be able to handle the load ? All my agents would be using the softphones 2. What should be the ideal CPU load that i need to watch - may be if the load average crosses 6 or 7 - should i worry ? 3. Even though i m adviced against AGI scripts (as they eat precious CPU cycles)- they seem very powerful and i m desperate to use them...Will the above setup get hampered in any way if i use them ? 4. Now scalability - If i want to increase the agents to 50 from 30 and add another PRI - what are the areas i should focus on - another machine ? or some additional RAM and processor ? I;ve been working all along on Dialogic but want to shift to Asterisk as it has lot of features and just fits in my needs (PBX + IVR in 1 box! ). Please advice Thanks in advance Sriram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Problems with DTMF on IVRs
last time i had this issue with teliax, they recommended to upgrade to 1.4 On Fri, Aug 29, 2008 at 3:44 AM, Chris Mason [EMAIL PROTECTED] wrote: I tried DTMFmode=auto and it did not help. Any further ideas? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Asterisk Realtime pounds MySQL
J.M. wrote: On Fri, Aug 22, 2008 at 7:41 PM, Tilghman Lesher [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Thursday 21 August 2008 10:08:53 J.M. wrote: I am running Asterisk 1.4.21.2 http://1.4.21.2 with Realtime. I have a phone setup in the database and when I connect that phone to Asterisk there are suddenly an endless number of SELECT * FROM sip WHERE name = '1001' AND host = 'dynamic' queries being run. The only way to stop the flood of queries coming from Asterisk to restart the Asterisk process. Even disconnecting the phone doesn't stop Asterisk from running the queries. Has anyone seen this before? Why would Asterisk do that and does anyone know the fix? Asterisk does that because realtime data is not cached by default, so for each access of the peer in question, Asterisk needs to reload the data on the peer from the database. If you'd like, turn on rtcachefriends in sip.conf, which will cache the peer for the duration of the registration interval (or whatever you have rtexpire set to). Also, to get correct behavior on reload, you'll need to have rtupdate turned on. Some of the behavior isn't quite right in 1.4.21.2 http://1.4.21.2, even, but it should be fixed once 1.4.22 is released. BTW, I would otherwise have responded sooner, but I am on vacation this week, and I am not responding to email as quickly as I would usually. Another way, which has worked so far for me, is to set the qualify field in the sip table (or whatever you called the table that corresponds to the sip.conf file) to no. I found this out from reading the following URL: http://www.asteriskguru.com/tutorials/peer_is_now_unreachable.html If this continues to work it has the advantages of putting as little in the .conf files as is possible and keeping the real-time feel of using a database without having to worry about whether the cache is updated or not. jm Ok, but WHY is he getting an ENDLESS # of selects. Sure * needs to get the data, but unless he had an ENDLESS series of CALLS to/from that phone should * be making all those queries ?? and HOW is this going to scale up ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 query a remote MySQL DB from dialplan
Yes, the add on will allow you to do this. No problem at all, as this is one of the Most Valuable parts of * I don't know Why they don't cover it in * the future of ... Rich wrote: I want to query an existing MySQL DB from my Asterisk Dialplan. This to check one field in a table in a database on a remote DB server. Is this possible using 'app_addon_sql_mysql' from asterisk-addons pkg? I would like to use an ODBC connector eg. unixODBC. I would like it to be 'stock', ie. part of the standard release and supported into the future. I have researched this for a couple days now (Asterisk Wiki and maillist) and am confused by the many and varied 'solutions' I find online. I so far haven't been able to navigate thru all the online docs to a real solution. If I am finally successful... I will contribute a step by step HOWTO to the Wiki. Any hints and pointers will be greatly appreciated. Thanks, Rich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Asterisk stress call test
Saul Bejarano wrote: Remember the rule of 30Mhz per call when you kill the machine and also the bandwidth usage on connected calls. Kind regards, Saul Bejarano aby azid wrote: Hi everyone, I'm required to make a stress call on Asterisk server ( 2000 calls per seconds). Are there tools for me to do this sort of test. I was thinking of sending loads of Asterisk call files simultaneously (starting with 100 call files). Really appreciate if anyone can come up with ideas or tools for me to achieve this. Cheers, Aby Azid Vyke Asia Where did you get the Rule of 30Mhz per call ??? Wouldn't this be highly dependent on whether it to TDM over a T1 or whether it was in SIP , and which CODEC it was using. And why would a properly configured machine Die, have a HIGH Load Average - YesDIED - Sounds like WinBlows to me Yoa - Aby - You need to define your test scenario more fully. Are you making a call OUT of the box , into the box, a MIX How Long are the calls ? Net, net , how many simultaneous call are you going to have ? How man Call Originations are there ? How may Call Answers ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Asterisk stress call test
OK - but again - more specifics are needed. If you are going TDM over T1 that is a Totally Different Animal than cranking up all these using IAX or . Also, you still have to identify how many simultaneous calls you will have. Again 1000 calls done essentially all at once is a different animal than if if stagger them , even a little. So your traffic Profile will make a huge difference aby azid wrote: Hi, Thanks for the reply mates, to Al Baker, It's a stress test for Asterisk outgoing calls, this is to see how Asterisk cope when thousands(1000 - 2000) of calls made simultaneously from the server. To Mik, where do I find the pbx_spool.c ?, really appreciate if u can explain more details on the method you used. Cheers, Aby Azid Vyke Asia On Fri, Aug 15, 2008 at 1:45 PM, Saul Bejarano [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Remember the rule of 30Mhz per call when you kill the machine and also the bandwidth usage on connected calls. Kind regards, Saul Bejarano aby azid wrote: Hi everyone, I'm required to make a stress call on Asterisk server ( 2000 calls per seconds). Are there tools for me to do this sort of test. I was thinking of sending loads of Asterisk call files simultaneously (starting with 100 call files). Really appreciate if anyone can come up with ideas or tools for me to achieve this. Cheers, Aby Azid Vyke Asia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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] HP server and Meetme applications
aymen warfalli wrote: Hi list I got one *HP* ProLiant *DL380 G5* - *Quad*-*Core* Xeon E5440 2.83 with 4 gig *RAM* I install Centos 5.2 64 bit and it is rumming pretty well and I need to use it as voice conferencing application (Meetme) server for high number of users fit to 8 E1 links (240 users ) with echo cancellation using same coding use g711 my qustion is this server is this server suitable for 240 users on meetme application on the same asterisk at the same time ?and what is the dimensions of one conference room should I biuld ? and finally if i can go for more users at same server ? AyMaN ALMONTAHA .ICT 11 AUG 2008 Whatever answer you get, I would approach this project in a SLOW, METHODICAL manner. i.e put 1 E1 card, get system performance metrics and user experience ADD the second card, TEST for voice quality and gather metrics. And then TEST some more. It will Likely work, BUT, I think you are venturing into an area with some large potential alligators. But, don't plug everything in Friday afternoon and expect zero problems Monday morning :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] shared mysql connection in dialplan
HOW ? Rizwan Hisham wrote: have done it, and its working fine. but still expecting to receive some new ideas. On Wed, Aug 6, 2008 at 2:12 PM, Rizwan Hisham [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi all, i just finished developing some incoming call features in a macro. that macro gets executed everytime an incoming call is received and a new mysql connection is made using the MYSQL cmd in dialplan. i want to use a single mysql connection for every incoming call. my idea of doing it is like this, i want to get a mysql connection in a global variable, just to share the connection with different incoming calls. Im not sure if this can be done. I am going to try doing it somehow, meanwhile i want your suggestions about how i can share a mysql connection with different calls in a dialplan. I am using asterisk1.4.2 and asterisk addon1.4.0 package for mysql connectivity. Thanx in advance -- Best Regards Rizwan Hisham -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] vars in Macros called by DIAL with option M
I would suggest putting a NOOPIn the MACRO to ensure the variable IS
actually getting SET.
As I understand it VARIABLES are GLOBAL and what you are doing is
correct, BUT, This could be a learning opportunity for me too.
Be advised, there seems to be push by DIGIUM for folks to use the
subroutine rather than MACROS now
H, Can the Dial command CALL a SUBROTINE as it does a MACRO ???
Ruddy Gbaguidi wrote:
And if you use DIALSTATUS and ANSWERTIME to check the last dial status,
you need to take care of the following bug
http://bugs.digium.com/view.php?id=13216
Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and
would
like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in macros
called by DIAL so that I can use these vars in the Dialplan or in the h
extention.
best regards
Thomas
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Re: [asterisk-users] vars in Macros called by DIAL with option M
Err - Ok - let me ask this in MUCH simpler way
1 - In dialplan , you set a Variable called MYVAR, to Apple
2 - You go into MACRO and NOOP the VALUE of MYVAR -What SHOULD it BE ?
3 - While IN MACRO you set VALUE of MYVAR = Pear
4 - You leave MACRO and get back to DIALPLAN and NOOP the value of MYVAR
- What should it be
=== 2nd Question ===
CAN the DIAL command call a SUBROUTINE instead of a MACRO ?
If so WOULD that help him out ?
Any clarification much apprecatted
Tilghman Lesher wrote:
On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and
would like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in macros
called by DIAL so that I can use these vars in the Dialplan or in the h
extention.
There isn't any good way to do that, period. When it comes to inheritance,
variables are only inherited from a master channel to a slave channel. In the
case of the Macro operating within the Dial, that Macro is occurring
exclusively on the slave channel. You cannot directly set variables on other
channels (for obvious race-condition reasons).
However, you could do this in a roundabout way, either by using a database or
by using shared variables in trunk. You'd need to first set (in the master
channel, before the Dial) an inherited variable containing the name of the
master channel, i.e. Set(_masterchan=${CHANNEL(name)}), then use that
inherited variable to set the shared variable in the master channel from the
slave channel, i.e. Set(SHARED(foo,${masterchan})=...). Finally, you would
be able to access the shared variable in the master channel with
${SHARED(foo)}. Again, the SHARED function is only available in trunk at this
time, although you could probably backport it to 1.4 with minimal trouble.
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Re: [asterisk-users] vars in Macros called by DIAL with option M
Thomas Winter wrote: On Tuesday 05 August 2008 18:02, Ruddy Gbaguidi wrote: I don't think you can do that because, asterisk, in the caller thread will only read MACRO_RESULT to know if he has to connect the call or not. A workaround will be to : 1. before the dial, add a row in a database table and retrieve an id 2. pass the id to test_connect and test_connect will then write his variable value into the database 3. after the dial,. use the id to retrieve the needed variable. Yes - this is possible, but the enormous ovehead to accomplish somthing that otherwise would seem to be very straightforward, i.e , get the value BACK from a very basic programming constract, call it a MACRO or Subrotine, just seems starteling and excessive. It seems to necessitate writing spagetti code. Or am I missing soimething ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 many quad T1 cards
You mean running , 400 Calls on 1 BOX ? Even if you COULD do it, the gods of TELCO would have you burn in hell for stacking that much critical traffic on ONE Intel, non - high availability box Jerry Geis wrote: Assuming you have a Quad core machine, at least 4 GIG ram, will a machine like this handle 4 Quad T1 cards? is that advisable? What about running AGI's on such a machine. Will the machine handle starting/stopping all those AGI's? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Custom Filename for Incoming Agent Calls
Could you clarify what you mean as inherited
In The dial plan for a given call I thought All variable were GLOBAL
to that call ??
Thanks
lenz wrote:
Hi Ricardo,
Try this:
exten = s,11,Set(MONITOR_FILENAME=/var/spool/queues/PA-${UNIQUEID})
exten = s,12,Set(TRANSFER_CONTEXT=queuetransfer)
exten = s,13,queue(q-pa|t|||)
The TRANSFER_CONTEXT is used for transfers. If you need the filename
inherited, add a double underscore before it.
Thanks
l.
In data Wed, 30 Jul 2008 23:09:11 +0200, Ricardo Melendez
[EMAIL PROTECTED] ha scritto:
Hi, to all, I have configured 3 Inbound/outbound agents queues, I
record
Outgoing calls with custom filename like
outgoing-${callerid(num)}-${EXTEN}-${TIMESTAMP}.gsm
but I need to record Incoming calls and asterisk by default add 13 digits
number to inbound recordings like Agent-001-1298375678-890.gsm, how I
can
customize this filename recordings?
Thanks in advance.
Ricardo Melendez
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Re: [asterisk-users] HI ~ good friend,
I must disagree. Dimensioning of Asterisk is a very sorely lacking area and is one of the main area CISCO and such eats its lunch. There simply no a base of solid metric that allow for true provisioning . Yes, there are INVALUABLE anecdotal reports from people who have been kind, and sharing of their experiences and for which are all very very grateful. BUT That that just is not the same as as solid, vendor based Metrics. Can you imagine calling and asking DISCO, What do I need for 400 calls an their answer is Here please go read these mostly outdated anecdotal reports and call back with your order Sorry. I love *, but this area of it is not where it needs to be. Dean Collins wrote: Hi welcome to the asterisk community. The answer you want are here; http://www.voip-info.org/wiki/view/Asterisk+dimensioning The short answer is; Pretty much yes, depending on hardware and horizontal scaling with multiple servers sharing the load. Cheers, Dean *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *??? *Sent:* Friday, 1 August 2008 9:43 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] HI ~ good friend, hi ~ nice to meet you, i just join here, today, i am a student, and i am very interesting in asterisk. and i have a IP-PBX server, made by me with my friend, while when i studying, i have a question, is there any limit users for asterisk? ex) registed users number is 1000 or 1 or 10 like that, is that possible? and how about the concurrent calls? 1000 concurrent calls is possible? or 2000 concurrent calls? my PBX server's user is just less then 15, almost my friends, so, i can't test, over 10 users and 1000 concurrent calls, please tell me, it is possible or not? thanks your permission to join there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Custom Filename for Incoming Agent Calls
Wow - Thanks a bunch. Likely save me about 12 hours of struggle
Tilghman Lesher wrote:
On Friday 01 August 2008 14:16:42 Al Baker wrote:
lenz wrote:
Hi Ricardo,
Try this:
exten = s,11,Set(MONITOR_FILENAME=/var/spool/queues/PA-${UNIQUEID})
exten = s,12,Set(TRANSFER_CONTEXT=queuetransfer)
exten = s,13,queue(q-pa|t|||)
The TRANSFER_CONTEXT is used for transfers. If you need the filename
inherited, add a double underscore before it.
Could you clarify what you mean as inherited
In The dial plan for a given call I thought All variable were GLOBAL
to that call ??
Thanks
No, variables are global to the CHANNEL. Calls are bridges between two
channels. Variables are not transferred to a dialled (slave) channel, unless
you set up inheritance, as noted above.
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Re: [asterisk-users] announcement server using asterisk
Quote Recently I discovered a cool new site called Google. They have lots of information about ISDN cards. :-P Grüße, Philipp Kempgen Yes - There is also a lot of bogus, incorrect, crap. His question was fair, on-topic, politely asked and as such hardly deserves to be made fun off Dean Collins wrote: Lol crackup. Having said that here is some help. Don't even think of using a laptop that's just dumb. Next - check out www.voip-info.org you'll find what you need there. Regards, Dean Collins +1-212-203-4357 (Direct) +61-2-9016-5642 (Sydney in-dial) http://www.Cognation.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Saturday, 26 July 2008 9:57 AM To: Asterisk Users Subject: Re: [asterisk-users] announcement server using asterisk ballamudi madhulika schrieb: Can I use Asterisk as an announcement server. We want to build announcement server with ISDN PRI card terminating on our server and announcement being fed on the incoming calls. Yes. Also is there any ISDN card available for Laptop. Recently I discovered a cool new site called Google. They have lots of information about ISDN cards. :-P Grüße, Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
Quote Yet amazingly (if this is, indeed, a source of amazement for you), CCM and other Cisco software can be just as buggy as anything OSS, if not worse. This is simply NOT TRUE and shows a complete lack of understanding of modern software development. CISCO software is developed in a CMM environment. It has a formal test methodology and uses Automated Testing on EACH new release to ensure that 100% of the software that functioned in the Last Release, actually works in this release. Further, there is mandatory soak-testing for all new software. Sorry, anyone who wants to compare Professional TELCO GRADE software development with Open Source is just Completely and Totally freakin clueless. Alex Balashov wrote: T G wrote: I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and Telepresence systems I have two IP patents for the VoiP Lite protocols and have been designing and building OSS IPBXs for companies including Google going back to 2001. I'm not mentioning any of that to be jerk I mentioned it to say I'm as qualified as anyone to to compare the CCM and OSS servers. The only fair way to compare the two is a list of weights features, for example if cost is your biggest feature then OSS is better, if support is your biggest feature than Cisco wins. When a customer is comparing the costly (TCO) and best supported systems in the world with hundreds of thousands installed systems for the large global companies on the planted backed by 54,000 employees and over $25b in the bank vs, a FREE system with one layer of support maybe two layers of support, the features don't even come in the evaluation in my opinion. I once asked a manager why did you buy the CCM and he said no one ever got fired for buying Cisco if anything wrong, If push the OSS and it goes I could loose my job. I would get a list of the important features, because there is no answer to your question of which is better. Yet amazingly (if this is, indeed, a source of amazement for you), CCM and other Cisco software can be just as buggy as anything OSS, if not worse. Depending on how critical the bugs or other support exigencies, the TCO can be driven way up. Except with the OSS community, you report the bug, and usually get a quick fix - even if it's a significant issue for you, not necessarily most of the installed base. If by chance that proves not to be the case, the source code is available, and you can fix it yourself. With Cisco, you pay for expensive support and get to file some complaint with the TAC. Yay. There are many, many angles from which onec an look at this in one's TCO / OPEX formula. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco Call Manager to Asterisk conversion
Quote I need to replace a cisco call manager with an asterisk box. WHY ? You want your TELCO to be LESS Reliable with LESS SUPPORT Grygoriy Dobrovolskyy wrote: Search someone in local area, remote configuration of server is possible but configuring the phones is more difficult, you need someone to load firmwares, ect 2008/7/24 Chad Whitten [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: I need to replace a cisco call manager with an asterisk box. Phones are cisco 7940 and 7910. I know the 40's can use SIP but the 7910's have to use the skinny/sccp driver. Its been quite awhile since I did anything with asterisk, so I am looking for some assistance with the configuration and am willing to pay. Its a basic setup, 30+ phones, incoming lines via PRI, 1 dial plan for incoming and outgoing - nothing fancy there, voicemail for each phone and DID number for each phone. -- Chad Whitten [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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] finding out on hold channels
While this is in place, how about sip show channels and show channels ? On Fri, Jul 25, 2008 at 4:56 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Jul 25, 2008 at 2:59 AM, Al lists [EMAIL PROTECTED] wrote: I noticed that i' m not getting any manager event for hold and unhold of a channel. is this normal? Also is there any easy way through either CLI or manager to find out which one of the channels are on hold? I checked show channels that did not show a channel being on hold or not, also sip show channels does show that but it has call id instead of channel id. Hi, There was recently a thread regarding this on asterisk-dev (http://lists.digium.com/pipermail/asterisk-dev/2008-June/033466.html). There was message explaining how to do this by adding custom code to Asterisk sources, and I guess it could be already done in trunk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Acceptance testing of a new PRI
Quote
seems like a dial-by-span syntax.
What is Dial-by-span ?
I have looked and cannot seem to fund that term.
More likely a comment on my ability to find it than on it obscurity
Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
So I guess I need finally to end up with
exten = _88XX1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _88XX1NXXNXX,2,Dial(Zap/${EXTEN:2:2}-1/${EXTEN:4},30,o)
Err.. that's not mine. It seems like a dial-by-span syntax.
Just remove the '-1' .
exten = _88XX1NXXNXX,3,NoOP(${DIALSTATUS})
exten = _88XX1NXXNXX,4,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _88XX1NXXNXX,5,Hangup
exten = _880X1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _880X1NXXNXX,2,Dial(Zap/${EXTEN:3:1}-1/${EXTEN:4},30,o)
exten = _880X1NXXNXX,3,NoOP(${DIALSTATUS})
exten = _880X1NXXNXX,4,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _880X1NXXNXX,5,Hangup
Which I just retested and it works.
Now to figure out how to do it across IAX channels from one server to
another.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL
PROTECTED]
Designer The Things I Think RFC
2100
Ashworth Associates http://baylink.pitas.com '87
e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647
1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
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[asterisk-users] finding out on hold channels
I noticed that i' m not getting any manager event for hold and unhold of a channel. is this normal? Also is there any easy way through either CLI or manager to find out which one of the channels are on hold? I checked show channels that did not show a channel being on hold or not, also sip show channels does show that but it has call id instead of channel id. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
I agree, No manager gets fired even if a Cisco Call Manager goes south. that's not the case with Asterisk. With limited experience that i have with both, i hit more bugs using Asterisk than a CCM, but this is not relevant to your final answer. If you can afford CCM, and you can live with less flexibility and features, i would choose Cisco. If you prefer to have cheaper solution and more features and flexibility, Asterisk is good. With Cisco, everything is cisco, handsets are designed for Cisco, it connects to Exchange much more in depth than even microsoft response point. unlike Asterisk, unfortunately exchange integration is not something you may get in close future and that can be a deal breaker for some companies, but you dont pay per seat license. and so on. On Thu, Jul 24, 2008 at 2:56 PM, Senad Jordanovic [EMAIL PROTECTED] wrote: T G wrote: I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and Telepresence systems I have two IP patents for the VoiP Lite protocols and have been designing and building OSS IPBXs for companies including Google going back to 2001. I'm not mentioning any of that to be jerk I mentioned it to say I'm as qualified as anyone to to compare the CCM and OSS servers. The only fair way to compare the two is a list of weights features, for example if cost is your biggest feature then OSS is better, if support is your biggest feature than Cisco wins. When a customer is comparing the costly (TCO) and best supported systems in the world with hundreds of thousands installed systems for the large global companies on the planted backed by 54,000 employees and over $25b in the bank vs, a FREE system with one layer of support maybe two layers of support, the features don't even come in the evaluation in my opinion. I once asked a manager why did you buy the CCM and he said no one ever got fired for buying Cisco if anything wrong, If push the OSS and it goes I could loose my job. I would get a list of the important features, because there is no answer to your question of which is better. What you mentioned above is mostly correct presuming you are referencing OSS being provided by an organisation with limited resources and perhaps limited experience in OS. Spin that into a perspective of a well organised company harvesting full potential of OS, adding its own proprietary software level allowing it to offer value products and EXCELLENT support, then I will strongly disagree with you. In particular where customer solution isn't just a solution, but rather its products and people becomes your business's communications partner. Senad www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] sip extension compromised, need help blocking brute force attempts
If you are trying to reject an IP address to connect to asterisk, there is no need to run iptables. Each SIP definition in sip.conf can have: deny=0.0.0.0/0.0.0.0 permit=192.168.135.1/255.255.255.0 just set these values and it wont accept anything from that IP. On Mon, Jul 7, 2008 at 7:37 PM, Dovid B [EMAIL PROTECTED] wrote: - Original Message - From: spectro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 01, 2008 8:02 PM Subject: Re: [asterisk-users] sip extension compromised,need help blocking brute force attempts On Tue, Jul 1, 2008 at 11:19 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Fix your logger.conf, then. -- Tzafrir Cohen What am I missing? [EMAIL PROTECTED] ~]# cat /etc/asterisk/logger.conf ; ; Logging Configuration ; ; In this file, you configure logging to files or to ; the syslog system. ; ; For each file, specify what to log. ; ; For console logging, you set options at start of ; Asterisk with -v for verbose and -d for debug ; See 'asterisk -h' for more information. ; ; Directory for log files is configures in asterisk.conf ; option astlogdir ; [logfiles] ; ; Format is filename and then levels of debugging to be included: ;debug ;notice ;warning ;error ;verbose ; ; Special filename console represents the system console ; ;debug = debug ;console = notice,warning,error ;console = notice,warning,error,debug ;messages = notice,warning,error full = notice,warning,error,debug,verbose ;syslog keyword : This special keyword logs to syslog facility ; ;syslog.local0 = notice,warning,error ; [EMAIL PROTECTED] ~]# The script seems to run off the messages log. Uncomment the messages line and the reload the logger in asterisk (logger reload from the CLI). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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 monitor Asterisk logs ?
I think this is another area DIGIUM has failed to address in any meaningful way. If they TRULY see themselves as a TELCO replacements for large shop they REALLY need to step up to proving INFO, WARN, ERROR messaging in a unified reliable manner. Such as a SNMP messaging ability for all INFO, ERROR, and WARN level messages. The very ideal of having to parse a log file for error messages whose form and meaning may be added to , deleted, or changed by the vendor at any time is just reasonable Anthony Francis wrote: perl script. Olivier wrote: Hi, How can I be notified anytime a given warning message appears in Asterisk logs ? I've got a running system that produces cause 34 warnings (Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)) once or twice a week. I would like to like to be notified (by email, phone, ...) anytime such warning message occurs in log file. I was thinking of using logwatch but wondered if anything better exists. Any advice ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Asterisk as an IVR solution
SO does that mean that if he used BACKGROUND is a SubRoutine he would get the correct or desired action , from his point of view? It would jump to the 1 Extension in the SUBROUTINE ? Tilghman Lesher wrote: On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote: It's a known problem. If you call Background() in a macro, then Asterisk will look for the extensions to jump to in the CALLING Macro/context and NOT the Macro that the Background() app was called in. I wouldn't call it a known problem. It works precisely as it was designed to work. It may not work the way that you want it to, but it works like a Macro: an independent set of instructions, with substitution, that acts as if it were invoked inline with the calling location. That is why Background will match in the context of the calling location: it acts like it never left that original context (and, in a way, it really didn't). Subroutines are a different beast, and they are available with the Gosub/ Return set of routines in app_stack.so. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Diagnosing dropped calls...
Quote Seriously though, if your business lives and dies by the phone system, get T1 with SIP from your provider directly If your business lives and dies, get that regular, boring, RELIABLE, TDM-T1. SIP/VOIP/Whatever - Cool fun, great when it works TDM-T1 - Unsurpassed reliabilty Steve Totaro wrote: Unfounded rumors say that ABE doesn't come with app_rnddropcall ;-] On Fri, Jul 11, 2008 at 12:40 PM, Carlos Chavez [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The other thing that baffles me about this setup is that it only seems to happen to people who are connected to the internal network in the office. They have about 30 remote users that have not reported this same problem, their issue is usually bandwidth related from their home connection. We have checked the internal network several times and there is not any obvious problem (apart from the dropped calls). They use high end Cisco switches and they were just audited to make sure there were no configuration errors. All the internal phones are Aastra (most are 9133i and some others 53i). On Thu, 2008-07-10 at 20:55 -0400, Steve Totaro wrote: Try dropping the IAX2 and only use SIP. Don't ask why? Just give it a try and see if things improve for you. Also when you assume, you make and ass out of you and me (just a little joke, get it? ass-u-me.) You could be hitting an overloaded router or whatever along the way, 10mbs fiber does not mean low latency or lost packets. Seriously though, if your business lives and dies by the phone system, get T1 with SIP from your provider directly (point to point) with G729 or just get a real ISDN or POTS lines. And then you will still have dropped calls depending on your volume and how vocal your users are. Usually, once they perceive a problem, then even if the other side of the call is on a cell and the cell drops the call, you will get a complaint. The only way to track those down are on a case by case basis with ANI II codes 61-63 http://www.nanpa.com/number_resource_info/ani_ii_assignments.html Thanks, Steve Totaro On Thu, Jul 10, 2008 at 7:15 PM, Carlos Chavez [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: My customer has a 10mpbs fiber connection to the Internet so we have always assumed that the connection is not really a problem. We will look into it. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 -- ?Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Asterisk as an IVR solution
Could you clarify how you end up with 1.4 Backport ? If you go to DIGIUM and download 1.4 do you have a backport 1.4 or is there a super-secret-non-more-secret-archive one would get it from ? I have never really understood this. Thank You Tilghman Lesher wrote: On Friday 11 July 2008 12:07:37 Douglas Garstang wrote: A subroutine with arguments? In 1.6, yes, or in the 1.4 backport, yes. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Asterisk as an IVR solution
Douglas Garstang wrote: Well, a macro is the closest thing the dial plan has to a subroutine, and without that, we might as well be programming in Assembler (no subroutines, local variables, lots of goto's... sound familiar?). Doug. - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 7:20:40 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Friday 11 July 2008 01:28:34 Douglas Garstang wrote: Well I can tell you that it makes a difficult programming environment, just a little more difficult. It means I can't implement a menu as a single reusable piece of code inside a macro. That's the point. A Macro is NOT a subroutine. It's like saying that you can't effectively hammer a nail with a screwdriver, and therefore you think the screwdriver has a known problem. There's nothing wrong with the screwdriver; it simply is the wrong tool for the job. I must somewhat disagree with you on this. 1) A MACRO could reasonably viewed as the Current Context, so if the jumping/branching from extension to extension that takes place in other contexts, it would if fact be quite reasonable and expected that this would happen in a MACRO. 2) As SUBROUTINES did not come standard until 1.6, it might be reasonably stated that no appropriate tool existed until 1.6, and since good programming practice uses subroutines, and a MACRO did not work like subroutine, even though it LOOKS like one, people are not fully happy that the closest tool they had, did not do the job Just a thought , no flame intended or implied. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Asterisk as an IVR solution
Thank You - clears up a LOT I did not fully grasp Tilghman Lesher wrote: On Friday 11 July 2008 01:05:22 Al Baker wrote: Tilghman Lesher wrote: On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote: It's a known problem. If you call Background() in a macro, then Asterisk will look for the extensions to jump to in the CALLING Macro/context and NOT the Macro that the Background() app was called in. I wouldn't call it a known problem. It works precisely as it was designed to work. It may not work the way that you want it to, but it works like a Macro: an independent set of instructions, with substitution, that acts as if it were invoked inline with the calling location. That is why Background will match in the context of the calling location: it acts like it never left that original context (and, in a way, it really didn't). Subroutines are a different beast, and they are available with the Gosub/ Return set of routines in app_stack.so. SO does that mean that if he used BACKGROUND is a SubRoutine he would get the correct or desired action , from his point of view? It would jump to the 1 Extension in the SUBROUTINE ? Yes, if he used Background within a Gosub, it would behave the way that he expects. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Asterisk as an IVR solution
Why can't you call Background() from a MACRO ? Isn't is just an Application like any other ? Curious minds want to know ! Quote There's also the fact that you can't call Backgound() in a macro, Douglas Garstang wrote: Don't know about Asterisk 1.4, but in Asterisk 1.2 it expects the input in seconds. If you try and use 0, it seems to drop back to a default of 5s. - Original Message From: MFH [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 10, 2008 12:37:31 PM Subject: Re: [asterisk-users] Asterisk as an IVR solution From what I can tell Read allows for a floating point input which uses ast_waitfordigit that accepts milliseconds as input. Douglas Garstang wrote: Admittedly I have not used the ExternalIVR app. Is it any good? I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, it can do it, but boy it is UGLY. There's also the fact that you can't call Backgound() in a macro, which forces you to use Read() which won't accept a timeout of 1s. There's no DTMF background detection while playing SayDigits so you have to roll your own by calling an external AGI and concatenating sound files. Yuck. By the time you code in logic for handling timeouts and incorrect responses to menu's with all the gotos and what-not, it turns into a god aweful mess. Sure, you can do it. Doug. - Original Message From: Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Sent: Thursday, July 10, 2008 10:37:55 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi. We are building an application that will provide users with the ability to call in and report an absence. The caller will have to validate themselves and the call tree will be dynamic, based on data in a MySQL database. We will have many customers, each calling a separate phone number, each having a different call tree. New customers will be added regularly and we do not want a solution that requires extensive programming each time (the call trees are different in subtle ways from each other). Is Asterisk a great solution for this? If not do you know what would? If so, we need someone to help us set it up, can you suggest someone? Thanks in advance. Best. Mark Asterisk certainly is a great solution for this. If you find you need or want extra flexibility, the external IVR app. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Asterisk as an IVR solution
Yes , you could easily do this with asterisk. If you have formal specs for this project, I would be interested in exactly what you are trying to do. Email me off-line. Steve Totaro wrote: On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi. We are building an application that will provide users with the ability to call in and report an absence. The caller will have to validate themselves and the call tree will be dynamic, based on data in a MySQL database. We will have many customers, each calling a separate phone number, each having a different call tree. New customers will be added regularly and we do not want a solution that requires extensive programming each time (the call trees are different in subtle ways from each other). Is Asterisk a great solution for this? If not do you know what would? If so, we need someone to help us set it up, can you suggest someone? Thanks in advance. Best. Mark Asterisk certainly is a great solution for this. If you find you need or want extra flexibility, the external IVR app. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Loose connection with MySql.
errr -you mean Asterisk doesn't ALWAYS check this and reconnect with the database ?!? WTF Since the CDRs are the literal Cash and Life Blood of many application why the heck would it NOT do this as part of its minimal basic operation ??? If it Doesn't do this for CDRs does it NOT do it for RealTime ?? If not, one could it up,screwed,blued and tatoed Is this functionality or lack there of documented anyplace ??? Michiel van Baak wrote: On 09:54, Tue 24 Jun 08, Catalin S. wrote: Hello, I configured asterisk to use mysql for CDR. Well when i check from time to time I realize that asterisk loose connection with mysql (i use phpmyadmin and i watch the processes). Can anybody tell me how can i solve that problem? I want to have all cdr statistics logged in mysql, is very important for billing. Thank you for support. Use cdr_adaptive_odbc backport for 1.4. That one does a check if the connection is still working, and if not it will reconnect. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Dial Command Option D Early Bridged
How do other applications, such a the automated dialers from telemarketers, reliably detect when the call has been answered ? I thought this sort of basic functionality that had been around for quite awhile. Jared Smith wrote: On Thu, 2008-06-12 at 16:43 +0800, tcchan wrote: However, in my experience, the timing the call get bridged is not consistance, Do you happen to be calling out over an analog phone line? In the case of dialing out an analog line, we have no easy way of knowing when the far-end has answered the call, so the call is considered answered at the time the call is dialed. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Asterisk Openfire Asterisk-IM Plugin Performance Observation
i used it on one server a little while ago. my primary use was ability to show each user's status on spark. i did not get consistence results, phone status was not accurate. and did not try it after that, maybe its fixed in newer versions. On Fri, Jun 20, 2008 at 2:44 PM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: See below: Erik Anderson wrote: On Fri, Jun 20, 2008 at 12:47 PM, JR Richardson [EMAIL PROTECTED] wrote: So now the PBX is over 1.2 Gig for the installation. Typical PBX installs are under 600 Meg. This makes me wonder about server stability, reliability and performance as uptime creeps on and user count increases over 50 to 100+. Increased data on the hard drive won't really have an affect on reliability or performance. Can anyone give me feedback on real world experience with this type of setup and any performance issues that my arise? I can't speak directly to the asterisk + openfire situation. I can, however, say that I've been running openfire for nearly a year now on a very highly-loaded server (other than openfire, it's running nagios and cacti, monitoring about 300 devices around our network) - the load average on this 5-year single processor old dell server is pegged near 1.00 24x7. I haven't had a single problem with openfire, and I have between 50 and 100 open sessions at any one time. In the year that I've been running openfire, I've only had to restart it once, and that was to upgrade the software. It takes very little CPU, and a modest amount of RAM. Is it better for production to run Openfire on a separate server than the PBX? What's your definition of better. Is it better to not have all your eggs in one basket? Is it better to only need to purchase one server? Is it better to only have one server to manage/update/etc versus two? My biggest concern is deploying a 100+ user environment with high call volume and high chat volume. Java seems to be a bit resource hungry with the user notifications and call pop ups. I would hate to have the IM server walking over Asterisk and affecting call quality or PBX stability. Speaking personally, I'd have no problems putting openfire and asterisk on the same box. If needed, you could even just nice the We run with the openfire process on the same box as the * server - we have not had a single problem with openfire in over 2 years now. openfire process down to a lower priority than asterisk - it's not as latency-sensitive as asterisk is. I'd doubt you'll need to do that, though. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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] suggestions for IAX ATA device or phone in US
anyone has used or bough one? would appreciate comments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up, the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference with about 12 other people from around the USA. Bandwidth issues aside, will this work or will all the different latencies cause issues? Yea I know, I could just try it and find out but it is going to take alot of time to get everyones schedule to line up, I don't want to go through the trouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mysql and extensions.conf
yes - but what would REALLY BE GOOD is if func_odbc
allowed Muli-stepped SQL. Since that is the ONLY way to execute a
TRANSACTION
How they thought it was a Good Idea to hamstring func_odbc like they
did is beyond me.
Tilghman Lesher wrote:
On Monday 02 June 2008 05:48, Atis Lezdins wrote:
You can use func_realtime in dialplan, that will be much faster as it
doesn't create separate process (as AGI does), and uses internal
asterisk connection pool, so no extra code in dialplan.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+realtime
That assumes that he's using a realtime table. From the OP's description,
it sounded like he wanted to query a column of an arbitrary table. Another
solution, in addition to the MYSQL app, would be func_odbc:
func_odbc.conf:
[FOO]
dsn=mysql-asterisk
read=SELECT status FROM foo WHERE id='${ARG1}'
extensions.conf:
GotoIf($[0${ODBC_FOO(123)} 0]?open:closed)
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Re: [asterisk-users] Asterisk just stops working...
No - I just would like to suggest that if you provide a solution in a more clear English manner, more people can benefit from you knowledge Which I assume is why you posted it in the first place. Jay R. Ashworth wrote: On Thu, May 29, 2008 at 04:24:57AM -0400, Al Baker wrote: Quote THen, fire up under the debugger. When you're all locked up, use ^C to halt and leave the debugger in command, and do the thread apply all bt thing. That should be revealing. If I may suggest , what would REALLY be 'Revealing' is if you could be just a bit more clear in your explanation and about 900% LESS in the techno babble. While the thought is in the Right Place do you REALLY expect anybody to know what the hell you mean by : When you're all locked up, use ^C to halt and leave the debugger in command, and do the thread apply all bt thing. That should be revealing *Just a thought* If you want paid-quality tech support... pay someone. You might want to read this: http://www.catb.org/~esr/faqs/smart-questions.html if you have just any questions at all about the tone of the conversations you see on a technical mailing list on the Internet. HTH. HAND. Cheers, -- jra ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What does reason 8 for failure means in Manager
you mean the CO gave an All-circuts-are-busy tone ??? If not, what does AST_CONGESTION mean Philipp Kempgen wrote: Sanjay Rajdev schrieb: I tried to call a number on the ZAP channel through manager, I got an Unknown reason for failure, with the following Originate Response. Event: OriginateResponse Privilege: call,all Response: Failure Channel: Zap/G0/ Context: callback Exten: 6563 Reason: 8 Uniqueid : NULL CallerID : 1234 CallerIDNum: 1234 CallerIDName: ABCD Can anyone Please let me know what does Reason 8 means here. Congestion (AST_CONTROL_CONGESTION). Grüße, Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] Asterisk 1.6 Realtime Database must use ', ' not '|'
quote And hackers ignoring pleasantries to get right down to the technical issues isn't abusive at all ABUSIVE - No not at all. Unnecessarily rude, insensitive, tacky - Yep Jay R. Ashworth wrote: On Fri, May 23, 2008 at 01:25:43PM -0400, Donny Kavanagh wrote: This is getting downright abusive, and is totally uncalled for, this is not a list for personal attacks. You thought that Steve suggesting JT step in was abusive? If that's not what you meant, then you need to either a) be clearer, or b) reply to the proper message. And hackers ignoring pleasantries to get right down to the technical issues isn't abusive at all. See Jargon File; see also Asperger's Syndrome, How To Ask Good Questions. Cheers, -- jra ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc.
Quote Oh and also, in my implementation there are no queues. It seems to be not related, I've had it in EVERY version of Asterisk I've used. Hmmm- maybe this should be mentioned in the next is * Really Good Thread ? Mark Hamilton wrote: Same here. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: May 22, 2008 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc. Steve Totaro wrote: On Thu, May 22, 2008 at 3:56 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Steve Totaro wrote: On Thu, May 22, 2008 at 2:02 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Mark Hamilton wrote: Hi, Yesterday I made a change in queues.conf and so tried doing a reload app_queue.so in the CLI. (Using 1.4.18). It didn't seem to do anything, infact all action on CLI stopped. Then, I did a reload. Same thing. After that there was no other way.. because even stop now wouldn't work, so I did a service asterisk restart And then asterisk kept giving the same thing on prompt Died successfully and all that it usually says when you issue a stop now, except it kept showing that on root prompt after doing a service asterisk restart. Did a killall asterisk, and finally it stopped. Then started asterisk service. It was fine. Did a full restart at night, and it was fine. NOW, I wanted to do a reload again today mid-day when in full use, and it still didn't work, and ALL of the above happened again. -- How do I diagnose what's causing this? Thanks, Mark. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I've had this problem before, haven't debugged it. I definitely look forward to hearing what is said about this. Example from my recent experience, I wanted to restart the server and so did pbx0*CLI restart now But nothing happened...system continued to allow calls to take place. I've found that sometimes exiting and reconnecting to the CLI helps, but there have been a couple occasions where NOTHING would allow the server to restart save for a reboot. Even killall asterisk didn't kill the process Sherwood McGowan You are using Asterisk 1.2.x? I have seen this many, many times. Sometimes the CLI becomes unresponsive, sometimes queues crap out or stops delivering calls to agents, sometimes it just takes a bit and then becomes responsive again. The rule of thumb is don't reload queues when there are people in queue, at least that seems to eliminate the problems I have seen. Makes sense too. Not sure if it is fixed in 1.4. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Oh and also, in my implementation there are no queues. It seems to be not related, I've had it in EVERY version of Asterisk I've used. I have observed it on repeated general reloads on all versions. That's why I don't reload very much, only planned. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My problem exists even when issuing a restart now or stop now command at the CLI. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk concurrent calls count
Glad I was able to foster some good open discussion. Hopefully DIGIUM will take to heart some of the thoughts expressed here and end up with a BETTER SOLUTION for ALL. Steve Totaro wrote: Inline On Fri, May 16, 2008 at 11:44 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Friday 16 May 2008 09:11:11 Steve Totaro wrote: On Fri, May 16, 2008 at 9:56 AM, Tilghman Lesher wrote: On Friday 16 May 2008 06:59:15 Al Baker wrote: this is one very weak area for *. There is NO ANSWER. Now in fairness to *, the answer DOES depend on a # of critical variables. How much CODEC to CODEC transcription is going on. How many MEET Me conferences are going on. On the other hand, DIGIUM COULD, since they have a lab take 4-5 'standard' workloads on two of the most common hardware boxes, say Dell HP, and run x # of transcriptions and show the #'s. Then x # of meet-me conferences. Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks Rockwell and NORTEL can tell you this for every piece of hardware they sell. It is a an area DIGIUM need to man-up in. I'm not sure what your problem is with Digium. They sell several machines for which they publish very specific numbers as to how many users those machines will support (the Switchvox appliances). Note that these machines are configurable only from the web interface, and they do not allow you to install additional software. In other words, when they give you a specific machine, with a ton of those variables controlled, they can give you a number. Digium is under no obligation to give you numbers for your own hardware. That's up to you (and you get to control your own set of variables). It seems any constructive criticism offered, you take as an attack against Digium. That is not a good attitude. I don't see how you figured out what I was thinking. Al said Digium doesn't publish any numbers, and I responded, saying that he was incorrect; Digium does indeed publish numbers (they're just not for his hardware). I'm not sure what your problem is with Digium. Proof, period. While under no obligation, it certainly would help sales. Whose sales? If you're talking about the appliances, then yes, I'm sure the publication of those numbers help with sales. If you mean your own sales, well, you're right, Digium's numbers probably don't help your sales. You could certainly put together a lab and do your own testing. Why don't you do that? Sales in general. You don't need to benchmark everything, just a few basic benchmarks, maybe gear it to your hardware and SIP as a gateway, then build from there. Most companies do this. I have my own lab and bechmarks but they are for Sangoma hardware and very specific servers and all geared to callcenter apps. I take Appliance Numbers with a grain of salt. The sales model of SwitchVox (and most others) is based on number of ports (SoHO, SMB, Enterprise) not maximum number of ports that the appliance could actually handle if not artificially constrained. Consider the maximum number of ports that Switchvox will enable on a single machine and consider that the maximum number that they're willing to support comfortably without running into some hard limit. You never want to run into a hard limit in the field anyway. High powered ervers are cheap and so are appliances once you settle on an enclosure and guts and start cranking out boxes. Hard limit common. This is in the style of legacy proprietary systems and anther reason why the sale cycle goes a little tougher than a custom job. Asterisk with FreePBX (and maybe Druid) eliminate these artificial constraints on usage. Yes, but the point of those constraints is to permit support a manageable job. Yes, you could probably add 2 or 3 or 10 or 15 to the number of calls that a particular machine could handle, but from a support perspective, it doesn't matter how many the machine could theoretically handle, it matters how many it could handle in the particular installation in a supportable configuration (those are all those pesky variables we've been talking about). Maybe that is what the official corporate answer is or, you were brainwashed to believe, but I tend to think it is to sell SMB and Enterprise software and support. It is all about money. I didn't fall off the turnip truck yesterday. I have load averages and CPU usage stats in my mind for all the various usages and hardware through experience in my mind. Of course they are only valuable to the exact setup I was doing. Precisely. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX deployment big problems: Voip traffic analysis
look - you MUST have a minimum of the following - a clear 24x7 graph of all you network segments show packet loss, packet delay for several weeks prior to 1st turn up. -Unless you have a 100% totally dedicated IP network for you voice, you must have Qos on every piece of hardware in the network, and you must test it to makes sure it works. As you turn up the service, in controlled stages you MUST measure network and systems. and fix any errors or bad trends in the data. Or you can say the hell with, just turn it up, and see what happens :) Bhrugu Mehta wrote: hi, Yes, there are many problem to implement and setup asterisk in a callcenter. but , all these problem can be remove if you set up your hardware and your LAN network verywell. Generaly, your server Configuration should be greater and your LAN also. You have to use Proper Codecs for voice. Generaly , g729 is greater. regards, Bhrugu Mehta On 5/16/08, gincantalupo [EMAIL PROTECTED] wrote: Hi, hope not to be OT :) after more than 3 years of PBX installations we can adfirm Asterisk is stable enough to be considered a good product but still we encounter a lot of problems when deploying a new PBX. It seems that the biggest problems are all networking related: one way voice (also inside a LAN), calls drops, etc... How do you face this kind of problems? Which diagnose tools/methods do you use? Thank you. Giorgio Incantalupo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk concurrent calls count
this is one very weak area for *. There is NO ANSWER. Now in fairness to *, the answer DOES depend on a # of critical variables. How much CODEC to CODEC transcription is going on. How many MEET Me conferences are going on. On the other hand, DIGIUM COULD, since they have a lab take 4-5 'standard' workloads on two of the most common hardware boxes, say Dell HP, and run x # of transcriptions and show the #'s. Then x # of meet-me conferences. Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks Rockwell and NORTEL can tell you this for every piece of hardware they sell. It is a an area DIGIUM need to man-up in. Alexey Shimeshov wrote: Hello, Alexander. AO Hi Asterisk Users, AO I'm interested in how many concurrent calls Asterisk can process without AO troubles. I mean 1 Asterisk server (software) like either proxy or media AO server (any numbers will be appropriate). AO 1. Is there any limitations by the software? What is this number? AO 2. What is the maximum count of concurrent calls you've ever seen/tested? Look at this example http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a PSTN gateway to Asterisk using PRI
This is 'basically' a tie-line between the boxes. Yes - it is done all the time between PBX's. You are basically nailing up a circut between the boxes. It could be a simple as a simple POTS leased line or a multi-t1 bundle between them. How it is physically done with DIGIUM's boards under * ? Someone else will have to answer that Pascal Maugeri wrote: Hi I have a system (S) that has a PSTN gateway to accept incoming calls and setup outgoing calls from/to Telco network. In the other hand I have a distant Asterisk box (A) that I would like to connect to (S) using the PRI interface. I understand that the proper way is to order to my Telco two PRI lines one for (S) and another for (A), and configure (S) and (A) to call each other numbers when they have to interconnect. Now, might it be possible to connect directly (A) and (S) using their PSTN interfaces without having to go through to my Telco ?! Does it make sense ? Is it technically feasible ? I guess that the Telco network is providing routing, number assignation, etc. and it sounds pointless to do this. Nevertheless could you confirm it is possible/impossible and why ? Is there a better way to do that ? Thanks in advance, Pascal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk for Larg
Whoa - you need some highly reliable, TELCO quality iron with some 1st class support for that. Do you realize what your downtime in that environment would would cost you ? Look, * is cool , fun an customizeable etc. But it IS NOT carrier grade hardware and it is NOT software produced in Certified Software Enviroment with a Certified CMM rating. Yes -its cool and neat and cheap. But for that big of a set up you need a whole different kind of solution Is Asterisk practically stable and reliable for a larg Enterprise has say a 1 phones , is there any case study like this? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk Deployment - Need some tips
The items most people do not address are: - QA - How do You tell if you you having Jitter,Packet Loss etc BEFORE the user scream - Disaster Recovery - from the small - DNS smokes - To Larger - * box with 96 ports smokes - Insuring EACH and EVERY piece ox network SUPPORT and USES QoS -Vendor SLA - How do YOU measure the service, WHAT happens outside 9-5 -HW Support - Your Quad port DIGIUM card smokes. Can you live w/out it ? Should you have a spare on hand ? If so how many -What TOOLS are you going to use to MONITOR this whole thing - all servers, switches -800 Phones - Minimum . Could be painful if folks are used to traditional TELCO reliability and Quality Andrew Latham wrote: Ditto. If you need to quantify the consultant to the powers that be just ask for an Infrastructure Audit. I have done several in the past that have saved tons of money that encouraged further phone projects. Finding dead phone lines to discovering unused but rented telcom gear is always fun. Also when setting up you test group make sure they actually use the phone and often... On Wed, May 14, 2008 at 9:32 AM, John Signorello [EMAIL PROTECTED] wrote: I would have to agree with Grey Man, a pilot project is one way to start up. I would also seriously recommend buying some consulting time from an experienced Asterisk PBX vendor/dealer/consultant. The cost is negligible in light of the scope of your project. A pilot project will only give you a glimpse of what is required. You have to have a design that incorporates your eventual build out. A pilot by itself is not going to give you that. You will need help from a source that can bring their experience to help you tip toe around the potential land mines you can encounter. regards, John Signorello Managing Partner ispbx.com 866 GO ISPBX Grey Man wrote: On Tue, May 13, 2008 at 12:17 PM, Matthew Ratliff [EMAIL PROTECTED] wrote: I'll be doing a new Asterisk deployment soon, and would like to gather your thoughts. Here are some items that need to be kept in mind: Support 800 phones (400 of which are analog) Concurrent calls ... ? but need to guess high so that the server can handle this. Voicemail will be required along with sending voice mail attachments to email server. Flash panel for switchboard operator. Needs to be a distributed server design for redundancy and fail-over. Will need to be integrated into an existing PBX until each building is switched over to use the Asterisk servers. If calling 911 from a building among multiple buildings, how can EMS find that person based upon the call? What type of data line should be used in this setup? T1? The physical network will support QOS and the like, so that is not an issue. What type of design/setup do you recommend for this? How about server resources...ie...CPU, RAM, Disk space. How about backups? Does imaging work best if a server were to fail? Any thing else you can think of? If this is a project for your work and it's your first Asterisk deployment then definitely don't go the big bang approach in the way you've outlined. If you do you could well be out of that job in 6 months! The first thing I'd recommend you do is find 10 or 20 people who are suitable as early adopters. The set up a single Asterisk server and give the early adopters a SIP phone each thats in addition to their normal desk phone and ask them to see how they go using the SIP phones for calls to each other, external calls and whatever else would make sense. Then 6 months and a lot of learning/experience/frustration later you'll know whether to get answers to your original questions or not. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS and Asterisk
You SHOULD be concerned with QOS. All the way to an including the vendor or your service cold really sucku Michael Graves wrote: On Thu, 15 May 2008 15:39:26 -0600, Joseph L. Casale wrote: I will have a small shop with ~4 phones using an HP server with Asterisk on it, it has two NICS and so I planned on plugging one into the cable modem, and the other into the switch. I was going to let this box perform NAT for the company but I am concerned about QOS for the VOIP portion. Anyone got a similar setup and care to share what they successfully implemented? Thanks! jlc You should take a serious look at Astlinux. It's en embedded Asterisk distro that handles routing, including QoS, when necessary. See www.astlinux.org. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installation Question
I thought that the point that you had to have a timing source for *. That source could be the clock off the T-1. But if you didn't have something like your T1 to provide master clocking ztdummy was something to provide the required a source for timing. Joseph L. Casale wrote: Sure if you don't need ztdummy, or is there a newfangled way around that? Thanks, Steve Totaro Hi Steve, I read the wiki and see this provides timing for Asterisk. Can you point me toward a description of what exactly this does? I was checking out the tutorial at http://www.hotbutteredit.com/video/voip/asterisk/asterisk_install.htm and noticed they never compiled either this or the Libpri which is what prompted me to assume I may not need it in my scenario. Appreciate the help! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3U server chassis Digium TE405P?
Getting the RIGHT card for the RIGHT bus type and the RIGHT Chassis is NOT as simple as everyone will lead you to believe. My suggestion, worth exactly what you paid for it :) Get Exact Spec for the card your are considering and FAX / Email to PC vendor and have him send you In Writing that the card WILL fit in the box and in the bus. Then I would get the Exact Spec for the BOX and BUS in the box and send to DIGIUM or their OEM and get THEM to tell you it should all work. Overkill - some will say yes. But THEY won't be sitting there with you if your expensive Server comes in and your expensive card come in an they no-workie together Sherwood McGowan wrote: Matt Watson wrote: I'm not sure if a full-height card would fit (vertically) in a 3U chassis... but I would probably also assume that if it would not, that the chassis/mobo would have a PCI/PCI-Express riser card that would mount the cards horizontally. Might want to check that out with the manufacturer of the chassis. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Monday, May 12, 2008 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 3U server chassis Digium TE405P? Gentlemen, First let me say it's great to be back on the Asterisk mailing lists. Those of you who have been around for a while will remember me as Rushowr. I look forward to answering questions and whatnot in the future, but for the moment I have a minor question that I cannot find a definitive answer for online. I am in possession of a Digium TE405P card which I _know_ will fit in a 4U chassis, but we are building a new server and cannot get a 4U from the supplier that my current client wants to use. However, we can get a 3U chassis. My question is, will this card fit? Does anyone out there have a 405 out there that they have installed in a 3U? Thanks in advance for any help that can be offered, Sherwood McGowan VoIP / Telecom Solutions Consultant ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks for the heads up, I've found full height capable 3U chassis. The worst thing about this whole ordeal was that I assumed (very bad idea, of course, stupid stupid stupid) that the 2u server had a riser card, which it did not :( Ah well, live and learn... Sherwood McGowan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which sound file formats?
Asterisk will automatically chose the best format - per ATFOT Roderick A. Anderson wrote: I've got the text files created -- thanks to Russell Bryant -- for re-building the core and extra sounds using another voice but I'm not sure which formats to actually build. This will be a small/personal system using Vitelity.net so will only have SIP connections. The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729, .gsm, .ulaw, and .wav. What are the minimal formats I need or can get by with. Possibly even an ordered preference list. Thanks, Rod ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_odbc creating records or best practice
Quote func_odbc can do whatever queries you give it. SELECT/UPDATE are simply the simplest cases that make it easy to understand the functionality *OK - but are the Limited to SINGLE STATEMETS or can you have a Muli-Statemnt Transaction ?*? Tilghman Lesher wrote: On Monday 28 April 2008 17:30, Robert McNaught wrote: I am trying to write a custom application which will integrate with an existing MSSQL crm system. We need to get ahold of the CDR(uniqueid) field in during call-time - I see from doing a DumpChan(), the CDR unique ID is available as soon as the call is created. CDRs usind odbc are only written once the call is completed. Does anyone know if it is possible to use func_odbc to create a temporary record then delete it so that this information is available to MSSQL. I was not sure if func_odbc was limited to just using UPDATE/SELECT queries. func_odbc can do whatever queries you give it. SELECT/UPDATE are simply the simplest cases that make it easy to understand the functionality. Would there be a better way to do this using the AMI or AGI? It just seems a little strange to use a database for storing temporary data such as this? I'd agree with you on that. I would tend to set variables directly in the channel, then query them out using AMI. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_odbc creating records or best practice
I would love to be able to issues the necessary Mysql commands to have true TRANSACTIONS Such as - Begin Transaction Select @var=agent.id, agent.exstension where agent.status='free' Update agent.status='BUSY' where [EMAIL PROTECTED] End Transaction Of Course the syntax I used above is just psuedo-code and NOT correct MySQL but I think you can see what I am trying to do. Which I think would be darn handy !!! Tilghman Lesher wrote: On Friday 09 May 2008 01:39:53 Al Baker wrote: Quote func_odbc can do whatever queries you give it. SELECT/UPDATE are simply the simplest cases that make it easy to understand the functionality *OK - but are the Limited to SINGLE STATEMETS or can you have a Muli-Statemnt Transaction ?*? As we don't isolate connections to a single channel, we do not support multi-statement transactions, no. It's an interesting idea, though. Could you expound on what you would like to see? It may wind its way into a future version of func_odbc. Perhaps only three extra statements, one to start a transaction, which also reserves the connection handle to the channel (note that this would require turning off connection sharing, which is the default, except for TDS databases), one to commit, and one to rollback (the last two here would also release the connection handle back to the pool). Would that be sufficient? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Thank you for your very kind offer. After repeatedly re-opening the ticket I finally got a clear specific answer. Strangely, in the 30 mins it took for me to take the answer, try it, and report back the results they had closed the ticket again so I couldn't report whether their solution fixed the problem or not. In fact it did, but I would have liked to have been able to document that so that others running into the same problem and scanning the bug report would know definitively if their answer was indeed correct. But - THANK YOU - and I will Certainly take you up on your most kind offer in the future! Tilghman Lesher wrote: On Thursday 08 May 2008 23:38:14 Al Baker wrote: Take a big shot of Valium before dealing with the bug tracker folks. There idea of help is to post You have an extra space in your line then CLOSE the ticket. That kind of clear, specific help is just what my doctor ordered to keep my BP nice and low If you have a problem with one of the explanations, please post the bug number here, and I'll be happy to explain it in more detail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Linux distribution to use in Asterisk server
this often becomes a religious discussion. my free advice worth all you paid for it - Redhat or one of the other distros that has been Certified on your choice of hardware and which has a Support Contract on it. Despite what others will tell you... Its a lonely place when your box no-workie and you have no support contract. if you buy Redhat on an HP box that is certified for it. It WILL work or they WILL get it working. sure maybe someone somewhere on some mailing list has the answer. But you got 96 lines down with a box and customers screaming... You want to hope, that maybe, someone will respond an respond correctly to your problem on a mailing list, or call a Customer Support Center staffed 24x7 with engineers trained on your specific hardware and you specific O/S ? Philipp Kempgen wrote: equis software schrieb: Hi, I allways use Gentoo y my Asterisk servers and work well, but what do you think about to use Ubuntu or another distibution?? I prefer Debian, but if everything works well and if you're familiar with Gentoo why change? Grüße, Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Channels Collide (Incoming Outgoing)
I know that everyone has gaps in their knowledge, but I am just staggered that systems are being sold/deployed with such fundamental TELCO workings not being understood. Frightening. C. Chad Wallace wrote: At 5:22 PM on 08 May 2008, Forrest Beck wrote: I have a client that is using the Sangoma A200DE with two phone lines attached. The problem is: They use their phone (Grandstream GXP2020) to dial out of the system. Instead of getting ringing, there is someone on the other end of the line that happened to dial in at the exact same moment. So now they are stuck talking with this person, instead of the one the originally called. The ZAP channels are in a dial plan context that instructs it to just dial the office phones. [zap1] exten = s,1,Dial(SIP/1001SIP/1002SIP/1003) exten = s,n,Voicemail([EMAIL PROTECTED]) Anyone know how to get around this? This is known in the telephony world as glare, and there's not much you can do about it, especially if you only have one line. If you have multiple lines on an over-ring (or hunt group or whatever you call it), the best thing to do is find out which way the telco assigns calls to those lines wrt how they are assigned to the Asterisk box. And then allocate outgoing calls in the other direction. On our installation, the calls are allocated from the first FXO port (Zap/25) up. So we set Asterisk to dial out starting from the last FXO port in the group by calling Dial(Zap/G2) (capital G means dial down from last, lowercase g means dial up from first). That minimizes glare. But, as I said before, if you only have one line, you can't do that... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI asterisk high balance
I think his connect/disconnect is going to take far longer than his 3 queries. The fact that Asterisk doesn't support sustained MySQL connection from the DialPlan is in fact quite a big deal that Digium seems to have its head in the sand about. And one of those things that SHOULD come up in those Is * Ready For Prime Time Threads Rizwan Hisham wrote: Well database really is a bottleneck for me. I am currently trying to do rating stuff in agi using perl. What im doing is i lookup the rate of every dialed code for every call from the mysql database using the longest match technique. The longest match technique costs atleast 2-3 mysql queries for every call untill the dialed code is matched out of 14000 dialcodes. I dont know how to calculate the exact delay due to execution of agi, but on the asterisk cli whenever that agi executes, there is a visual delay of about half a sec to move from the agi extension to the next extension (can anybody tell me how to calculate the delay). Now im planning to use the manager api for constant connectivity to mysql and to enhance the longest match technique. Can anybody help me with this? Is it a good idea to ue manager api for lookingup the rate of the live call? On Sun, May 4, 2008 at 1:34 PM, Grey Man [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If you've got anything but trivial AGI loads you should switch to FastAGI and put your business logic on a separate server to your Asterisk server. I use a deployment where a call could make up to 3 AGI requests per call before being put through (for things such as looking up accountcode, checking account credit, setting PSTN callerid). We monitor the time thw whole process takes and on average it's less than 100ms on an Asteisk server that peaks at 200 simultaneous calls (400 bridged) and 3 to 5 call set ups per second. The business logic processing the FastAGI calls is C# and .net which means Java would be able to handle it easily as well. The most likely bottleneck under high load will be your database. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Perhaps this should be tagged under Is * Ready For Prime Time ? Thread Isn't an 'appliance' supposed to be a 'plug-it-in-and-runs' sort of thing ? Julian Yap wrote: On Tue, May 6, 2008 at 1:38 AM, Benoit Plessis [EMAIL PROTECTED] wrote: We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy deadlock and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). There's an IAX issue with the security patch for 1.4.18.1... and 1.4.19.1. There's another thread on this. - Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Take a big shot of Valium before dealing with the bug tracker folks. There idea of help is to post You have an extra space in your line then CLOSE the ticket. That kind of clear, specific help is just what my doctor ordered to keep my BP nice and low Benoit Plessis wrote: Tilghman Lesher a écrit : On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote: lordfuknowsyou a écrit : Vinícius Fontes wrote: I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlocks Have you reported these issues on the bugtracker? Well, the problem is finding usefull data to report. I've 4 core dumps thats show differents things: two seems to be related to ControlPlayback: #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 #1 0x0809c579 in ast_readframe () #2 0x0809defc in ast_streamfile () #3 0x0805e786 in ast_control_streamfile () #4 0xb698be5c in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #5 0x08298700 in ?? () #6 0xb470aec0 in ?? () #7 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #8 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #9 0x in ?? () One is pretty generic: #0 0x0809c9bc in ast_closestream () #1 0x08085d91 in ast_hangup () #2 0x080cd3d8 in pbx_builtin_setvar_helper () #3 0x080cf08e in ast_pbx_outgoing_exten () #4 0x080fde65 in ast_inet_ntoa () #5 0xb7eec560 in start_thread () from /lib/tls/libpthread.so.0 #6 0xb703667e in clone () from /lib/tls/libc.so.6 and the latest is thread/iax2 related: #0 0xb7ee71c7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0 #1 0xb562a969 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #2 0x0079 in ?? () #3 0x in ?? () #4 0xb547a148 in ?? () #5 0x080f0508 in ast_sched_add_variable () #6 0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #7 0x0012 in ?? () But my main problem is when the system just froze, it start mostly by the Queue not working anymore, with member stuck in 'in use' stack (should not happen with IAX2 agent IIRC, given that we had to build macros using GROUP() to detect in use IAX2 agent) Then the console (asterisk -rcTvvv) start to freeze (completion doesn't work, message stop from being displayed and even command output is lost). And i'm reading http://www.asterisk.org/developers/bug-guidelines which speak of using SVN trunk version of asterisk, thing i'm not really eager to try on a live system... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
Are you saying the * server does NOT TRY to re-establish the BD connection ? Does your whole * SERVER freeze ? If NOT, what happens to you CDR records ? Anthony Francis wrote: Tilghman Lesher wrote: On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote: 5 maj 2008 kl. 19.58 skrev Tilghman Lesher: On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. So, we're back to where we started. A developer that can help us with a connection pool or a separate connection for each query would be a Nice Thing (TM). What issues are you specifically seeing that merit using multiple connections? I can specify an issue that would merit multiple connections, if the link to your db goes away Asterisk likes to freeze writing CDRs. I have a few remote * servers that this happens to. My solution so far has been to record CDR's to a local DB and then have a perl script that attempts to move them over to my transaction DB. I would suggest this solution to anyone who depends on their CDR records. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
I looked all over VOIP-INFO and ATFOT and could not find anything that
said or even suggested not using the mysql driver.(except NOT to have
BOTH drivers loaded at the same time). I could easily be missing
something. But the apparent BUG I am seeing is at such a Basic and
Simple Level of functionality that either DIGIUM ought to fix it ASAP or
update VOIP-INFO pages and their own documentation to say Broke - No
Workie and We Are No Gonna Fixie :)
Steve Totaro wrote:
On Mon, May 5, 2008 at 4:21 AM, Al Baker [EMAIL PROTECTED] wrote:
I would appreciate any and all advice on what appears to be a BUG (or a
brainfart on my part) with the MySQL add-on for Asterisk this is of FEDORA 8
fully patched with Asterisk Addons 1-4-6 with the Asterisk 1.4.18.1
It appears that the interface eats the first field requested from a
table. If only One Field is Requested from the Table , that field is eaten
ENTIRELY by Asterisk. If several fields are requested, the First Field Is
Eaten and the remaining filed are returned, but place in the WRONG Variable
since the 1tst fileld data was eaten. In the DIALPLAN below I have tried 3
Different ways to approach this.
Extension – Get only ONE (1) field from Table
Extension – Get THREE(3) fields from the Table and Quote Them.
Extension - Get THREE(3) fields from the Table
I have show the Output from the Asterisk CL for each, which clearly show
that SOMETHING is not
right. Maybe the Software, maybe the person using the software :)
Here is the Table in the Database.
mysql select * from agent;
+--+-+++-+
| id | cust_id | status |phone |tlce |
+--+-+++-+
| 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 |
| 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 |
| 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 |
| 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 |
+--+-+++-+
4 rows in set (0.00 sec)
Here is the DIALPLAN
exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
exten = ,n,MYSQL(Query resultid ${connid} SELECT\ cust_id\, \
status\,\ tlce\ from\ agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE is
${mytlce})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
exten = ,1,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\, \
'status'\,\ 'tlce'\ from\ agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE is
${mytlce})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
exten = ,n,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\ from\
agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid)
exten = ,n,NoOp(CUSTID is ${custid})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
Here is the Asterisk CLI Output
dial
== Console is full duplex
*CLI -- Executing [EMAIL PROTECTED]:1] MYSQL(OSS/dsp, Connect connid
localhost ivr ivrxxx dtc) in new stack
-- Executing [EMAIL PROTECTED]:2] MYSQL(OSS/dsp, Query resultid 5 SELECT
cust_id from agent where phone='1234567890') in new stack
-- Executing [EMAIL PROTECTED]:3] MYSQL(OSS/dsp, Fetch fetchid 6
custid) in
new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(OSS/dsp, CUSTID is ) in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(OSS/dsp, FETCHID is 1 RESULUT ID
is
.. 6 CONNID is 5) in new stack
-- Executing [EMAIL PROTECTED]:6] MYSQL(OSS/dsp, Clear 6) in new stack
-- Executing [EMAIL PROTECTED]:7] MYSQL(OSS/dsp, Disconnect 5) in new
stack
-- Executing [EMAIL PROTECTED]:8] Hangup(OSS/dsp, ) in new stack
== Spawn extension (default, , 8) exited non-zero on 'OSS/dsp'
Hangup on console
*CLI dial
== Console is full duplex
*CLI -- Executing [EMAIL PROTECTED]:1] MYSQL(OSS/dsp, Connect connid
localhost ivr ivrxxx dtc) in new stack
-- Executing [EMAIL PROTECTED]:2] MYSQL(OSS/dsp, Query resultid 5 SELECT
cust_id, status, tlce from agent where phone='1234567890') in new stack
-- Executing [EMAIL
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
I must be overlooking it, I pulled up the electronic version and searched for and read every instance where ODBC was mentioned and I could not find a single place where it said ODBC was to be the only or even the best method. If so I would never ever have gone down this road :( Quote And according to the O'Reilly book ODBC is the way to go. Roderick A. Anderson wrote: Steve Totaro wrote: A quote from Tilghman Lesher from a previous post. That's fine, but I have had the most horrid results using any distribution- supplied ODBC drivers. The best results are obtained by source-compiling the latest ODBC drivers, whether they be the MySQL ODBC Connector 3.51 or PsqlODBC. UnixODBC is fairly safe to use from distribution channels, however. And according to the O'Reilly book ODBC is the way to go. Though they use PostgreSQL for their examples and Asterisk is installed on a CentOS system the instructions are really good. Getting it to work with MySQL should be pretty simple and I'm sure on-line resources for doing this are be out there. soapbox Personally I never use MySQL except in cases where I am under extreme duress. Therefore I tried and tossed trixbox, AsteriskNOW, and freeePBX. Yes I know I can get around the database engine issue but that is what a distribution should be for: no hacking (or at least not-too-much) required. It is now CentOS 5, Asterisk from source, PostgreSQL (on another system) and hand edited (for now anyway) *.conf files. Maybe AsteriskGUI later. /soapbox Rod ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimum upload speed for Asterisk?
You also need to check for Packet Loss on the Link Erik Anderson wrote: On Thu, May 1, 2008 at 3:19 PM, Frank Tarczynski [EMAIL PROTECTED] wrote: Is 384kB up too slow? Probably not. Is there any guidance for the minimum upload speed for an Asterisk box? I'm guessing this is for just a few calls at a time, correct? I'd guess that rather than these quality issues being caused by cramped bandwidth, they're actually being caused by latency issues. Have you ever checked the latency of the connection between your asterisk server and your SIP/IAX endpoint? If it's really high (say 300ms+) or if the latency is really erratic, you'll have quality issues. You didn't mention whether you are doing traffic shaping on your upstream connection, so I'll assume you're not. That would be something good to look into - with traffic shaping, you can prioritize your VoIP traffic over all other types of network traffic. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
any of you guys have used FOP for drag and drop transfer on 30 40 phones environment? how stable is that? I'm playing with it but so far drag and dropping phone icon to another phone disconnectes the call. On Wed, Apr 16, 2008 at 2:02 PM, Lee Jenkins [EMAIL PROTECTED] wrote: Al lists wrote: Hi list, Any good drag and drop transfer call application for windows based systems you can advise ? Something like HUD perhaps? Yes. Maestro Control Panel (I authored this one) http://www.datatrakpos.com/pos/datatalk/maestro.aspx. There is also the nice flash based Flash Operator Panel http://www.datatrakpos.com/pos/datatalk/maestro.aspx There a couple of other ones out there too that I thought were nice, but can't remember the names. You should be able to find them by gooling for Asterisk Control Panel or such query. -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need examples of asterisk and mysql integration
Why would you go to the trouble of writing a PERL AGI and take the Performance Hit of using AGI as opposed to using the built-in MYSQL from the dial plan ? Mike Trest - On Travel wrote: Hi, I suggest you look at writing a PERL agi program to handle all of the MYSQL / DB access and just pass variables between your CONTEXT/dialplan. I have done a lot of these things. You can get PERL examples for DBI and use one of provided agi scripts as a prototype. ..mike.. At 04:13 PM 4/22/2008, you wrote: I'm presently working on a project to build a scheduling system accessible by both web and phone. on the web side one can query what items are available when by using the time or the item as a key then reserve for an available time slot. reservations may also be modified by the user that made them or an admin. Where may I find examples of doing similar things with asterisk? all I've been able to find thus far is examples of how to store call detail records and voicemail using a database. Thanks in advance, Eric P.S. Has anyone already built an asterisk/web based scheduling system like this? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Next step in extensions.conf after answer the phone in Queue
Why would you want a channel to continue after the caller has hung up.
I clearly am missing something here because I can't see what good that
would be. What do people do with this Continued Channel ?
What is is used for ? How Does having it help you ? ???
Atis Lezdins wrote:
Queue will continue if called person hangs up (and there's no option).
If caller hangs up, call goes to h extension in same context. Just the
same way as Dial with 'g'. There's a change in 1.6 that allows called
channel to continue if caller hangs up, so probably something like
this could be applied also to Queue (or was that actually working with
using Local channels?).
Regards,
Atis
On Wed, Apr 23, 2008 at 7:13 PM, AnDY [EMAIL PROTECTED] wrote:
Thank you for your answer.
But the Dial command has a option 'g' which means that after succes will
proceed next priorities in the dialplan. Is there something also for
Queue() because according to manual there is no option for it. So I am
looking for some other solution.
Andy
Tony Mountifield napsal(a):
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] wrote:
Hello everybody.
I was looking for the solution but nothing found. I have this in my
extensions.conf:
exten = 233,1,SetAccount(queue1)
exten = 233,2,Queue(queue1|rn)
exten = 233,3,NoOp(${QUEUESTATUS})
exten = 233,4,NoOp(${DIALSTATUS})
But when the call is placed in the queue and somebody answer it, it will
throw an error:
== Spawn extension (default, 211, 4) exited non-zero on
'Local/[EMAIL PROTECTED],2'
And no other command in extensions is executed.
Any suggestions?
Queue() is like Dial(), in that if it succeeds in connecting to someone,
it will not return to the next priority in the dialplan. However, if you
define an 'h' extension, that will get executed when the call is complete:
exten = 233,1,SetAccount(queue1)
exten = 233,2,Queue(queue1|rn)
exten = 233,3,NoOp(${QUEUESTATUS})
exten = 233,4,NoOp(${DIALSTATUS})
exten = h,1,NoOp(${QUEUESTATUS})
exten = h,2,NoOp(${DIALSTATUS})
Cheers
Tony
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Re: [asterisk-users] need examples of asterisk and mysql integration
I did not mean to stir up a hornet's nest or religious war :) The BASIC QUESTION I was trying to ask is this... Since the MYSQL add-on provides a way to interface with MySQL what is it that one gains or is trying to gain by writing their OWN AGI script to do the interface ? The only reason I mention this being a performance hit was that the book ATFOT mentioned that AGI scripts in general were less efficient and therefore one should do all possible work in the dial plan. Maybe this is true, maybe this is not true. Regardless. I would just like to borrow from the expertise and experience of other and ask the BASIC QUESTION above and hope that folks will share their experiences. I'm pretty sure they did what they did for very very good reasons that I am just not experienced enough yet to know. So The answer to the questions is ??? Steve Edwards wrote: On Wed, Apr 23, 2008 at 11:07:01AM -0700, Steve Edwards wrote: AGIs do not have a substantial performance hit and I think people need to get this misconception out of their heads. Writing AGIs in a scripting, non-compiled language may be great for prototyping and proving concepts where performance is not expected to be an issue. Personally, I don't write AGIs in anything but C. It's the sharpest tool in my kit and I know it best. On Wed, 23 Apr 2008, Tzafrir Cohen wrote: The dialplan isn't compiled either. I guess I didn't make that point forcefully enough. Now, if it is simple enough to be implemented with the dialplan, then you don't need to execute half the logic in a separate environment and don't need the performance hit. I'd like to put this performance hit thing to rest. I respect your knowledge and expertise -- I always seem to learn something new (to me) from your posts. How would you quantify the hit? On my recently retired dev box (1.6GHz Celeron), I could execute 100 AGIs per second but that doesn't really answer the question. Anyway, what about dialplan logic in lua? Anybody actually uses that thing? Oh great -- another scripting language to learn :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need examples of asterisk and mysql integration
Thx so much for taking the time to share. Damn Insightful Damn Helpful THANKS! Steve Edwards wrote: On Wed, 23 Apr 2008, Al Baker wrote: The BASIC QUESTION I was trying to ask is this... Since the MYSQL add-on provides a way to interface with MySQL what is it that one gains or is trying to gain by writing their OWN AGI script to do the interface ? I like doing serious work in an AGI instead of the dialplan because: 1) It allows me to use a compiled language. Not just for performance reasons, but because a compiler (or a strict interpreter) helps me protect me from me. My production dialplan is just over 600 lines. The sources to my AGIs are a bit more than 15,000 lines. I like that if I fat-finger a variable, the compiler will help me. If I fat-finger something in 16,000 lines of dialplan will I ever find it? 2) It allows me to hide complexity. I like having a single statement in my dialplan that says agi(block-ani). I know that in this single statement I am invoking code that acts as the gatekeeper to my system, allowing me to block callers by area code, area code and prefix, and the complete subscriber number. I know it works well and I don't have to look at it any more. 3) I can share better. It is easier to integrate a single statement into an existing dialplan than 2,000 statements with potential conflicts in context, template, and variable names. 4) I don't have to learn (what appears to me to be) a really obtuse syntax. Funky quoting and whitespace rules lead to accelerated hair loss. 5) I have a full toolbox. For instance, one of my AGIs (play-path) lets me pass a path and it returns the name of a WAV file in that path at random. Calling ftw() was a simple solution. Another AGI (auth-card) lets me submit an authorization request to my credit card processor. While waiting for the card response, I play Please wait while we validate your card in another thread in the same AGI. By the time the STREAM FILE is finished, I have the response so to the customer it appears instantaneous. How would I do either of these in dialplan? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Drag and Drop transfer application
Hi list, Any good drag and drop transfer call application for windows based systems you can advise ? Something like HUD perhaps? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Please keep us updated on your progress. I am considering putting several of these boxes in and I would love to hear how this comes out. Wish I had something to suggest. Ex Vito wrote: Hi list, After a lot of testing + troubleshooting, I guess I'm observing what I am now calling a regression with zaptel 1.4.10 (is it?) As such I call for peer feedback, before either asking Digium install support or filing a bug. Thanks in advance! System: HP Proliant DL380 G5 with 2x PCI-X + 1x PCIe riser card OS: Centos 5 Kernel: 2.6.18-53.1.14.el5 (also tested under 2.6.18-53.el5) HW: Digium TE220B, the one with HW echo cancellation (configured as 2x E1 via jumpers) Context: Pre-site installation of system, no E1 conectivity (loopbacks tested) /etc/zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=25-39,41-55 dchan=40 span=2,2,0,ccs,hdb3,crc4 bchan=56-70,72-86 dchan=71 Under zaptel 1.4.10, when ztcfg runs this gets logged in the kernel buffer: About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 1: Primary Sync Source VPM400: Not Present VPM450: echo cancellation for 64 channels BUG: soft lockup detected on CPU#0! [c044d448] softlockup_tick+0x96/0xa4 [c042ddc8] update_process_times+0x39/0x5c [c04196f7] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [f89bc1e7] init_vpm450m+0x32d/0x34a [wct4xxp] [f89a3b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f89a7ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042621c] release_console_sem+0x17e/0x1b8 [c0407406] do_IRQ+0xa5/0xae [f8994311] t4_dacs+0x211/0x24b [wct4xxp] [f8a01f6a] zt_ioctl+0x273/0x144f [zaptel] [c0457600] mempool_alloc+0x28/0xc9 [c04ddd33] cfq_resort_rr_list+0x23/0x8b [c04deb6c] cfq_add_crq_rb+0xba/0xc3 [c04dec72] cfq_insert_request+0x42/0x498 [c04d5175] elv_insert+0x10a/0x1ad [c04d908b] __make_request+0x31d/0x366 [c04de8b1] cfq_dispatch_requests+0x26a/0x46b [c04dde27] __cfq_slice_expired+0x8c/0xa5 [c04de8b1] cfq_dispatch_requests+0x26a/0x46b [c04d505d] elv_next_request+0x15c/0x16a [f88bc101] start_io+0x77/0xdc [cciss] [f88bf63e] do_cciss_request+0x32c/0x337 [cciss] [f88ccff0] __split_bio+0x408/0x418 [dm_mod] [f88cd6a6] dm_request+0xce/0xd4 [dm_mod] [c04d6a81] generic_make_request+0x248/0x258 [c04d8734] submit_bio+0xbf/0xc5 [c04548e2] find_get_page+0x18/0x38 [c04719ad] __find_get_block_slow+0xfb/0x105 [c0471cea] __find_get_block+0x15c/0x166 [c0471cea] __find_get_block+0x15c/0x166 [c0471d24] __getblk+0x30/0x270 [f885a485] journal_cancel_revoke+0x8a/0x96 [jbd] [f885a472] journal_cancel_revoke+0x77/0x96 [jbd] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [c041f871] __wake_up+0x2a/0x3d [f8856679] journal_stop+0x1b0/0x1ba [jbd] [c042a209] current_fs_time+0x4a/0x55 [c048626d] touch_atime+0x60/0x8f [c04552ee] do_generic_mapping_read+0x421/0x468 [c045478b] file_read_actor+0x0/0xd1 [c04548e2] find_get_page+0x18/0x38 [c0457319] filemap_nopage+0x192/0x315 [c046048f] __handle_mm_fault+0x85e/0x87b [c047f46b] do_ioctl+0x47/0x5d [c047f6cb] vfs_ioctl+0x24a/0x25c [c047f725] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 2: Secondary Sync Source Completed startup! Soft lockup ?! Hmmm... I'm ignorant on this, but it smells fishy ! For completeness sake, driver was previously loaded ok: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.10 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :18:08.0[A] - GSI 19 (level, low) - IRQ 98 Found TE2XXP at base address fdff, remapped to f8854000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x375a2400 Reg 1: 0x375a2000 Reg 2: 0x Reg 3: 0x Reg 4: 0x3101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff2031 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) After trying lot's of things (disable ILO, disable USBs, try different kernel, different TE220B, etc), I figured that this soft hangup does not show under zaptel 1.4.9.2... In all due honesty, I haven't got the faintest idea what kind of impact this could have. Side testing zaptel 1.4.10 on a simpler system, an HP Proliant ML110 (nearly a PC), the error does not show up as well. I checked the zaptel 1.4.10 ChangeLog
Re: [asterisk-users] Is Asterisk really good??
Quote We had a master source location.with a master image We cloned the hard drive with linux dd copy of master image Did the dd to clone it actually work on RAID devices Mike Trest - On Travel wrote: -Original Message- I'd be interested in sections like Rolling out a new server or How we maintain all the little configuration files without losing our sanity. Hi, I will contribute my 2-cents on how I maintained consistency on a large application with 64 + Asterisks that all had to have the same config and links back to a central DB. Whenever we needed a new machine, we just We had a master source location.with a master image We cloned the hard drive with linux dd copy of master image boot the new machine with this disk assign appropriate IP address perform some sanity checks prior to shipping Send either disk or full machine to remote COLO for physical install. After the machine came on line, it would have enough configuration to join the other members of the farm of asterisks. For intermediate updates, we used SSL-DSA keys between the master master image machine and each of the 64+ remotes. We would wrote our own script and gave it a list of each machine on which to perform the particular steps. When it was launched, we just went out to lunch or home at night while the remotes were updated. This application had as many as 6,000 simultaneous call running and we wrote the scripts such that each remote were placed in a take no calls status by the script so we did not kill any active traffic. We found that no canned package was useful to do this because each maintenance cycle was addressing a different part of the overall configuration and had slightly different commands that were needed. Any good script writer can do the same for what you described. Regards, ..mike.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
exvito - I know it is a pain in the cahoonkus - but would you consider sharing the OTHER Digium board issues you are having , the recommended steps you were given by Digium to troubleshoot them, and the results ? I think this real-wold experience wold be invaluable to the list. THX in Advance for sharing ! Ex Vito wrote: Your stack trace appears to possibly be stack corruption. Could you try either this branch: http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Or with a kernel that does not have 4K stacks enabled? You can check if your installed kernel does with the following command. $ cat /boot/config-`uname -r` | grep 4K # CONFIG_4KSTACKS is not set ...thanks for your feedback Shaun. I am currently nearing other troubleshooting issues regarding a TC400B (which will probably lead me to get in touch with Digium install support). So I have no schedule today to test your suggestions; maybe tomorrow / thursday. They are noted, however. :) Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
