Menu
▾
▴
oreka-user — Users mailing list
You can subscribe to this list here.
| 2005 |
Jan
|
Feb
|
Mar
|
Apr
|
May
|
Jun
|
Jul
|
Aug
|
Sep
|
Oct
(1) |
Nov
|
Dec
(4) |
|---|---|---|---|---|---|---|---|---|---|---|---|---|
| 2006 |
Jan
(7) |
Feb
(22) |
Mar
(11) |
Apr
(23) |
May
(13) |
Jun
(22) |
Jul
(8) |
Aug
|
Sep
(2) |
Oct
(8) |
Nov
(4) |
Dec
(2) |
| 2007 |
Jan
(16) |
Feb
(4) |
Mar
(6) |
Apr
(15) |
May
(10) |
Jun
(10) |
Jul
(43) |
Aug
(29) |
Sep
(15) |
Oct
(9) |
Nov
(6) |
Dec
(13) |
| 2008 |
Jan
(7) |
Feb
(7) |
Mar
(8) |
Apr
(26) |
May
(14) |
Jun
(12) |
Jul
|
Aug
(3) |
Sep
(6) |
Oct
(19) |
Nov
(12) |
Dec
(14) |
| 2009 |
Jan
(31) |
Feb
(25) |
Mar
(9) |
Apr
(7) |
May
(7) |
Jun
(15) |
Jul
(23) |
Aug
(20) |
Sep
(17) |
Oct
(14) |
Nov
(31) |
Dec
(15) |
| 2010 |
Jan
(38) |
Feb
(58) |
Mar
(42) |
Apr
(30) |
May
(22) |
Jun
(19) |
Jul
(12) |
Aug
(16) |
Sep
(25) |
Oct
(15) |
Nov
(28) |
Dec
(8) |
| 2011 |
Jan
(8) |
Feb
(18) |
Mar
(15) |
Apr
(13) |
May
(9) |
Jun
(18) |
Jul
(10) |
Aug
(23) |
Sep
(8) |
Oct
(8) |
Nov
(19) |
Dec
(1) |
| 2012 |
Jan
(1) |
Feb
(2) |
Mar
(4) |
Apr
(10) |
May
(1) |
Jun
(1) |
Jul
(4) |
Aug
(4) |
Sep
(7) |
Oct
(4) |
Nov
(4) |
Dec
(4) |
| 2013 |
Jan
(3) |
Feb
(1) |
Mar
(6) |
Apr
(1) |
May
(8) |
Jun
(13) |
Jul
(10) |
Aug
(7) |
Sep
(28) |
Oct
(22) |
Nov
(40) |
Dec
(19) |
| 2014 |
Jan
(19) |
Feb
(22) |
Mar
(12) |
Apr
(11) |
May
(6) |
Jun
(11) |
Jul
(8) |
Aug
(2) |
Sep
(10) |
Oct
(25) |
Nov
(9) |
Dec
(13) |
| 2015 |
Jan
(28) |
Feb
(8) |
Mar
(6) |
Apr
(9) |
May
|
Jun
(1) |
Jul
(3) |
Aug
(2) |
Sep
(4) |
Oct
(2) |
Nov
(4) |
Dec
(3) |
| 2016 |
Jan
(10) |
Feb
(5) |
Mar
(10) |
Apr
(6) |
May
(7) |
Jun
(7) |
Jul
(6) |
Aug
(14) |
Sep
(2) |
Oct
(6) |
Nov
(7) |
Dec
(3) |
| 2017 |
Jan
(4) |
Feb
(2) |
Mar
(6) |
Apr
(4) |
May
(1) |
Jun
(4) |
Jul
|
Aug
(1) |
Sep
(5) |
Oct
(1) |
Nov
(1) |
Dec
(1) |
| 2018 |
Jan
(4) |
Feb
(1) |
Mar
(10) |
Apr
|
May
(1) |
Jun
|
Jul
|
Aug
(1) |
Sep
|
Oct
(1) |
Nov
(1) |
Dec
|
| 2019 |
Jan
(1) |
Feb
(2) |
Mar
|
Apr
(3) |
May
(1) |
Jun
(4) |
Jul
(3) |
Aug
(1) |
Sep
(1) |
Oct
(1) |
Nov
|
Dec
|
| 2020 |
Jan
|
Feb
|
Mar
|
Apr
|
May
(1) |
Jun
|
Jul
(3) |
Aug
(2) |
Sep
(1) |
Oct
(1) |
Nov
|
Dec
|
| 2021 |
Jan
|
Feb
|
Mar
|
Apr
(1) |
May
(3) |
Jun
(2) |
Jul
(1) |
Aug
(2) |
Sep
(2) |
Oct
|
Nov
|
Dec
|
| 2022 |
Jan
(1) |
Feb
(2) |
Mar
|
Apr
(1) |
May
(5) |
Jun
(3) |
Jul
(2) |
Aug
(1) |
Sep
(1) |
Oct
|
Nov
|
Dec
|
| 2023 |
Jan
|
Feb
(2) |
Mar
|
Apr
(1) |
May
|
Jun
(1) |
Jul
|
Aug
|
Sep
|
Oct
|
Nov
|
Dec
(1) |
| 2024 |
Jan
(1) |
Feb
(2) |
Mar
|
Apr
(1) |
May
|
Jun
(2) |
Jul
(1) |
Aug
(3) |
Sep
(1) |
Oct
|
Nov
|
Dec
|
| 2025 |
Jan
|
Feb
|
Mar
|
Apr
|
May
|
Jun
(1) |
Jul
(1) |
Aug
(1) |
Sep
|
Oct
|
Nov
|
Dec
|
|
From: johnny_xing <joh...@ba...> - 2008-12-12 08:40:12
|
Hi, I have downloaded the free oreka audio and installed in my Linux asterisk server (around 30 concurrent calls). And installed orekaweb & orekatrack in my another windows server, these two servers are located in the same LAN. What I want to achieve are: 1. Record all the incoming / outgoing calls made through this asterisk server (both internal and TDM calls) 2. The audio files need to store in the windows server instead of the linux server. 3. Possible to search and replay through the windows server. I am not sure whether this could be done using oreka software, I don't mind paying for the commercial one if it's working. Any suggestion will be appreciated. Best Regards, Johnny Xing |
|
From: Henri H. <he...@or...> - 2008-12-10 23:07:30
|
Edward, Yes, this is a common problem if you look around in the mailing lists. Just update orkservice so that the hostname is the IP address instead of the DNS name. However, you will need to use the latest subversion version of OrkWeb/OrkTrack and compile it because the last posted binary has a bug where any orkservice manual setting is reset when a new recording comes in. Cheers, Henri -----Original Message----- From: Edward Ronquillo [mailto:ero...@gm...] Sent: 05 December 2008 14:32 To: ore...@li... Subject: [Oreka-user] Cannot playback recordings Hi everyone! I've finally installed orkaudio with orkweb and orktrack, and I can see the recordings coming in on the web gui (192.168.0.220) The only trouble I have is that I cannot playback the recordings. I checked the log directory and it does have the .wav and .mcf files - they are playable if I ftp the.wav files to my windows system and play them back with media player. I'm beginning to suspect it's the way I'm accessing the orkweb gui via it's IP address (192.168.0.220), because I noticed that when I move the mouse pointer over the play buttons, javascript would show http://voiprecorder:8080//2008/12/05/13/20081205_134915_192.168.1.200,18330. wav instead of http://192.168.0.220:8080//2008/12/05/13/20081205_134915_192.168.1.200,18330 .wav Has anyone faced this problem before? Edward ---------------------------------------------------------------------------- -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ _______________________________________________ Oreka-user mailing list Ore...@li... https://lists.sourceforge.net/lists/listinfo/oreka-user |
|
From: Edward R. <ero...@gm...> - 2008-12-05 19:31:33
|
Hi everyone! I've finally installed orkaudio with orkweb and orktrack, and I can see the recordings coming in on the web gui (192.168.0.220) The only trouble I have is that I cannot playback the recordings. I checked the log directory and it does have the .wav and .mcf files - they are playable if I ftp the.wav files to my windows system and play them back with media player. I'm beginning to suspect it's the way I'm accessing the orkweb gui via it's IP address (192.168.0.220), because I noticed that when I move the mouse pointer over the play buttons, javascript would show http://voiprecorder:8080//2008/12/05/13/20081205_134915_192.168.1.200,18330.wav instead of http://192.168.0.220:8080//2008/12/05/13/20081205_134915_192.168.1.200,18330.wav Has anyone faced this problem before? Edward |
|
From: Henri H. <he...@or...> - 2008-12-04 15:13:42
|
Edward, Is this the 32 bit or 64 bit version of fedora 8? Do you use the latest oreka svn revision or do you use the last official release as an rpm package? Henri -----Original Message----- From: Edward Ronquillo [mailto:ero...@gm...] Sent: 04 December 2008 10:09 To: Henri Herscher Cc: ore...@li... Subject: Re: [Oreka-user] high cpu in fedora core 8 Hi Henri! I just installed it, not connected to anything. There's no traffic going through, which concerned me quite a bit. I'm wondering if it might have to do with problems I was having installing ACE, which i had to ignore the dependencies so it could install. Edward On Thu, Dec 4, 2008 at 9:56 AM, Henri Herscher <he...@or...> wrote: > Edward, > > It depends on the workload. 99% CPU can be perfectly normal in high traffic > volume. How many concurrent calls do you think you have on your platform? > > Henri > > -----Original Message----- > From: Edward Ronquillo [mailto:ero...@gm...] > Sent: 04 December 2008 09:53 > To: ore...@li... > Subject: [Oreka-user] high cpu in fedora core 8 > > Hi! I'm noticing that oreka is taking up to 99% cpu on my fedora core > 8 based setup. Is this normal? > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > Oreka-user mailing list > Ore...@li... > https://lists.sourceforge.net/lists/listinfo/oreka-user > > |
|
From: Edward R. <ero...@gm...> - 2008-12-04 15:09:32
|
Hi Henri! I just installed it, not connected to anything. There's no traffic going through, which concerned me quite a bit. I'm wondering if it might have to do with problems I was having installing ACE, which i had to ignore the dependencies so it could install. Edward On Thu, Dec 4, 2008 at 9:56 AM, Henri Herscher <he...@or...> wrote: > Edward, > > It depends on the workload. 99% CPU can be perfectly normal in high traffic > volume. How many concurrent calls do you think you have on your platform? > > Henri > > -----Original Message----- > From: Edward Ronquillo [mailto:ero...@gm...] > Sent: 04 December 2008 09:53 > To: ore...@li... > Subject: [Oreka-user] high cpu in fedora core 8 > > Hi! I'm noticing that oreka is taking up to 99% cpu on my fedora core > 8 based setup. Is this normal? > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > Oreka-user mailing list > Ore...@li... > https://lists.sourceforge.net/lists/listinfo/oreka-user > > |
|
From: Henri H. <he...@or...> - 2008-12-04 14:56:46
|
Edward, It depends on the workload. 99% CPU can be perfectly normal in high traffic volume. How many concurrent calls do you think you have on your platform? Henri -----Original Message----- From: Edward Ronquillo [mailto:ero...@gm...] Sent: 04 December 2008 09:53 To: ore...@li... Subject: [Oreka-user] high cpu in fedora core 8 Hi! I'm noticing that oreka is taking up to 99% cpu on my fedora core 8 based setup. Is this normal? ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ Oreka-user mailing list Ore...@li... https://lists.sourceforge.net/lists/listinfo/oreka-user |
|
From: Edward R. <ero...@gm...> - 2008-12-04 14:52:36
|
Hi! I'm noticing that oreka is taking up to 99% cpu on my fedora core 8 based setup. Is this normal? |
|
From: Henri H. <he...@or...> - 2008-12-02 16:44:46
|
Conor, Have you tried it without any IP filter? Could you please send me offline the orkaudio.log files for both scenario 1 and 2? (Please zip before sending) Henri _____ From: Conor McTernan [mailto:con...@gm...] Sent: 02 December 2008 04:41 To: ore...@li... Subject: [Oreka-user] Allowed IP confusion and call reporting I have been experiencing some problems with how Oreka/OrkWeb have been reporting who are making calls. I am also seeing problems with only 1 side of conversations being captured. First some background. I have an asterisk box terminating a T1 line, this sits on my Lan at 192.168.215.5, I then have my PBX (PBX1) on the same LAN at 192.168.215.1. Oreka is installed and running on the asterisk box. My PBX at 215.1 will send all calls through the Asterisk to the PSTN. I have another PBX (PBX2) at another location (lets say the IP is 123.4.5.6). When people place a call from PBX 2 the calls are routed over the internet and sent to PBX1 which then sends the calls to Asterisk. This works fine, calls can go in and out of the Asterisk gateway with no problems. When I attempt to capture calls going through the Asterisk box coming from PBX2 though I can only capture the 'destination' side of the conversation i.e. the person who is being called. I have played around with different configurations in my config.xml but I cannot seem to get it working the way I would wish. Here are the different scenarios I have tried: 1) AllowedIpRanges: Includes the global IP address of PBX2 as well as the Local IP address of the Asterisk box MediaGateways: Includes the LAN IP of PBX1 This config results in both sides of the calls being captured, but the Local Party and RemoteParty show up as their IP addresses (which means I cannot tell who is placing the calls). 2) AllowedIpRanges: Include the Global IP of PBX2, Local IP of Asterisk (215.5) and the local IP of PBX 1 (215.1) MediaGateways: None The result here is that I see the correct details in LocalParty and RemoteParty but I only get 1 side of the call (the person being called). Has anyone experienced this issue before? I suspect I am not understanding something about how to do the config, or maybe I'm leaving something out of my AllowedIpRanges. Any ideas or help are appreciated, Cheers, Conor |
|
From: Conor M. <con...@gm...> - 2008-12-02 09:41:03
|
I have been experiencing some problems with how Oreka/OrkWeb have been reporting who are making calls. I am also seeing problems with only 1 side of conversations being captured. First some background. I have an asterisk box terminating a T1 line, this sits on my Lan at 192.168.215.5, I then have my PBX (PBX1) on the same LAN at 192.168.215.1. Oreka is installed and running on the asterisk box. My PBX at 215.1 will send all calls through the Asterisk to the PSTN. I have another PBX (PBX2) at another location (lets say the IP is 123.4.5.6). When people place a call from PBX 2 the calls are routed over the internet and sent to PBX1 which then sends the calls to Asterisk. This works fine, calls can go in and out of the Asterisk gateway with no problems. When I attempt to capture calls going through the Asterisk box coming from PBX2 though I can only capture the 'destination' side of the conversation i.e. the person who is being called. I have played around with different configurations in my config.xml but I cannot seem to get it working the way I would wish. Here are the different scenarios I have tried: 1) AllowedIpRanges: Includes the global IP address of PBX2 as well as the Local IP address of the Asterisk box MediaGateways: Includes the LAN IP of PBX1 This config results in both sides of the calls being captured, but the Local Party and RemoteParty show up as their IP addresses (which means I cannot tell who is placing the calls). 2) AllowedIpRanges: Include the Global IP of PBX2, Local IP of Asterisk (215.5) and the local IP of PBX 1 (215.1) MediaGateways: None The result here is that I see the correct details in LocalParty and RemoteParty but I only get 1 side of the call (the person being called). Has anyone experienced this issue before? I suspect I am not understanding something about how to do the config, or maybe I'm leaving something out of my AllowedIpRanges. Any ideas or help are appreciated, Cheers, Conor |
|
From: Henri H. <he...@or...> - 2008-11-17 17:06:20
|
Neal, This question should have been posted on oreka-devel. Please try: $ automake --add-missing Henri _____ From: Neal Piche [mailto:phi...@gm...] Sent: 15 November 2008 21:05 To: ore...@li... Subject: [Oreka-user] compiling orkaudio trying to compile from svn. I'm using ubuntu 8.10 and I am getting this error at the make stage, can anyone help? libtool: Version mismatch error. This is libtool 2.2.4 Debian-2.2.4-0ubuntu4, but the libtool: definition of this LT_INIT comes from an older release. libtool: You should recreate aclocal.m4 with macros from libtool 2.2.4 Debian-2.2.4-0ubuntu4 libtool: and run autoconf again. this happens when encoding orkbasecxx |
|
From: Neal P. <phi...@gm...> - 2008-11-16 02:05:34
|
trying to compile from svn. I'm using ubuntu 8.10 and I am getting this error at the make stage, can anyone help? libtool: Version mismatch error. This is libtool 2.2.4 Debian-2.2.4-0ubuntu4, but the libtool: definition of this LT_INIT comes from an older release. libtool: You should recreate aclocal.m4 with macros from libtool 2.2.4 Debian-2.2.4-0ubuntu4 libtool: and run autoconf again. this happens when encoding orkbasecxx |
|
From: Henri H. <he...@or...> - 2008-11-12 21:54:45
|
Neal, We might produce Ubuntu debs for the upcoming Oreka 1.0 release if we manage to spare a bit of time for that. Henri _____ From: Neal Piche [mailto:phi...@gm...] Sent: 11 November 2008 22:36 To: ore...@li... Subject: [Oreka-user] new debs? hey guys I was wondering if anyone had either something from the svn in a deb that works on ubuntu or perhaps even a repo that someone is keeping for it? Ubuntu updated some of the packages that orkaudio depends on and now it won't start so any help would be great. Thanks Neal |
|
From: Henri H. <he...@or...> - 2008-11-12 21:52:31
|
Mihail, This is a known problem that has been fixed in the latest svn version. Just check out the latest code or wait for the upcoming 1.0 release. Cheers, Henri -----Original Message----- From: M.Komarovskiy [mailto:zo...@bs...] Sent: 11 November 2008 14:52 To: ore...@li... Subject: [Oreka-user] service hostname Hello, I have a problem with changing hostname in service table of mysql database oreka version 0.5-313, after chaging it with mysql> update service set hostname = '192.168.0.2'; playing of recording works fine in orkweb, but in couple of minutes orkweb or orktrak rollback hostname to 'localhost'. Here is mysql server log: 24 Prepare [10] update Service set hostname=?, fileServeProtocol=?, fileServeTcpPort=?, fileServePath=?, ... 24 Query SHOW TABLES FROM `oreka` LIKE 'Service' 24 Query SHOW FULL COLUMNS FROM `service` FROM `oreka` 24 Query SHOW FULL COLUMNS FROM `service` FROM `oreka` LIKE 'serviceClass' 24 Execute [10] update Service set hostname='localhost', fileServeProtocol='http', fileServeTcpPort=8080, ... 24 Query commit 24 Query rollback Maybe there is some way to change this procedure to correct name/ip or way to switch off this kind of table update? Thanks in advance and big thanks to developers for such a great software Best regards, Mihail Komarovskiy zombieatbsxdotru ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ Oreka-user mailing list Ore...@li... https://lists.sourceforge.net/lists/listinfo/oreka-user |
|
From: Neal P. <phi...@gm...> - 2008-11-12 03:36:14
|
hey guys I was wondering if anyone had either something from the svn in a deb that works on ubuntu or perhaps even a repo that someone is keeping for it? Ubuntu updated some of the packages that orkaudio depends on and now it won't start so any help would be great. Thanks Neal |
|
From: M.Komarovskiy <zo...@bs...> - 2008-11-11 20:05:18
|
Hello, I have a problem with changing hostname in service table of mysql database oreka version 0.5-313, after chaging it with mysql> update service set hostname = '192.168.0.2'; playing of recording works fine in orkweb, but in couple of minutes orkweb or orktrak rollback hostname to 'localhost'. Here is mysql server log: 24 Prepare [10] update Service set hostname=?, fileServeProtocol=?, fileServeTcpPort=?, fileServePath=?, ... 24 Query SHOW TABLES FROM `oreka` LIKE 'Service' 24 Query SHOW FULL COLUMNS FROM `service` FROM `oreka` 24 Query SHOW FULL COLUMNS FROM `service` FROM `oreka` LIKE 'serviceClass' 24 Execute [10] update Service set hostname='localhost', fileServeProtocol='http', fileServeTcpPort=8080, ... 24 Query commit 24 Query rollback Maybe there is some way to change this procedure to correct name/ip or way to switch off this kind of table update? Thanks in advance and big thanks to developers for such a great software Best regards, Mihail Komarovskiy zombieatbsxdotru |
|
From: wraith d <wra...@gm...> - 2008-11-04 20:42:08
|
yes, that`s right.
Now everything it`s okay.
Thanks.
2008/11/4 Henri Herscher <he...@or...>
> This is pure SIP traffic, there is no RTP there, you are probably
> monitoring your SIP proxy, not your PSTN gateway. Just use wireshark to
> figure out if you have RTP or not on this link.
>
> Henri
>
>
> ------------------------------
>
> *From:* wraith d [mailto:wra...@gm...]
> *Sent:* 04 November 2008 10:22
> *To:* ore...@li...
> *Subject:* [Oreka-user] Hi all
>
>
>
> Hello.
>
> I`ve recently installed on an Windows Station the orekaudio software.
> The windows pc has 2 ethernet card. One for accessing the network, and the
> second for sniffing the voip traffic.
> The second ethernet is connected in an Cisco 3524 switch in an span
> interface.
> I see the traffic between mediagateways and SIP endpoints, but i don`t get
> any wav or native files.
> Also my tapelist.log file is empty.
>
> what could be the problem?
>
> Here is an output from orkaudio.log file :
>
> OrkAudio service starting
>
> 2008-11-04 16:43:38,109 INFO root:93 - Loaded plugin:
> ./plugins/RtpMixer.dll
> 2008-11-04 16:43:38,109 INFO immediateProcessing:53 - thread starting -
> queue size:10000
> 2008-11-04 16:43:38,125 INFO root:87 - Loaded plugin:
> audiocaptureplugins/VoIp.dll
> 2008-11-04 16:43:38,125 ERROR root:138 - Failed to start http server on
> port:20000
> 2008-11-04 16:43:38,125 ERROR root:46 - Failed to start command line server
> on port:10000
> 2008-11-04 16:43:38,125 INFO packet:835 - Initializing VoIP plugin
> 2008-11-04 16:43:38,187 INFO packet:753 - Available pcap devices:
> 2008-11-04 16:43:38,187 INFO packet:760 - *
> \Device\NPF_GenericDialupAdapter - Generic dialup adapter
> 2008-11-04 16:43:38,203 INFO packet:760 - *
> \Device\NPF_{9956A723-1254-42FB-B514-65A92D2D2F2F} - Intel(R) PRO/100 VE
> Network Connection (Microsoft's Packet Scheduler)
> 2008-11-04 16:43:38,203 INFO packet:781 - Successfully opened device. pcap
> handle:1362050
> 2008-11-04 16:43:38,218 INFO packet:760 - *
> \Device\NPF_{3122178E-0562-427B-AC0C-D528B196AD02} - Intel(R) PRO/100 S
> Desktop Adapter (Microsoft's Packet Scheduler)
> 2008-11-04 16:43:38,218 INFO packet:617 - Start Capturing: pcap
> handle:1362050
> 2008-11-04 16:43:43,218 INFO sip:351 - INVITE: sender:192.168.0.63from:0365128167 RTP:
> 192.168.0.63,60000 to:01321314943 rcvr:217.10.195.230
> callid:264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63<callid%3A264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63>
> 2008-11-04 16:43:43,234 INFO rtpsessions:567 - AVD: created by SIP INVITE
> 2008-11-04 16:43:43,234 INFO sip:351 - INVITE: sender:192.168.0.63from:0365128167 RTP:
> 192.168.0.63,60000 to:01321314943 rcvr:217.10.195.230
> callid:264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63<callid%3A264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63>
> 2008-11-04 16:43:43,250 INFO sip:351 - INVITE: sender:217.10.195.230from:0365128167 RTP:
> 192.168.0.63,60000 to:01321314943 rcvr:217.10.195.231
> callid:264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63<callid%3A264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63>
> 2008-11-04 16:44:14,062 INFO sip:351 - INVITE: sender:10.201.100.232from:0365121644 RTP:
> 10.201.100.232,53456 to:0356465196 rcvr:217.10.195.230
> callid:4b7fbcb65004e22124f96333643298f6@10.201.100.232<callid%3A4b7fbcb65004e22124f96333643298f6@10.201.100.232>
> 2008-11-04 16:44:14,062 INFO rtpsessions:567 - AVE: created by SIP INVITE
> 2008-11-04 16:44:14,562 INFO sip:351 - INVITE: sender:10.201.100.232from:0365121644 RTP:
> 10.201.100.232,53456 to:0356465196 rcvr:217.10.195.230
> callid:4b7fbcb65004e22124f96333643298f6@10.201.100.232<callid%3A4b7fbcb65004e22124f96333643298f6@10.201.100.232>
> 2008-11-04 16:44:14,578 INFO sip:351 - INVITE: sender:217.10.195.230from:0365121644 RTP:
> 10.201.100.232,53456 to:0356465196 rcvr:217.10.195.231
> callid:4b7fbcb65004e22124f96333643298f6@10.201.100.232<callid%3A4b7fbcb65004e22124f96333643298f6@10.201.100.232>
> 2008-11-04 16:44:19,578 INFO sip:351 - INVITE: sender:10.199.110.146from:0365125362 RTP:
> 10.199.110.146,53460 to:0313165657 rcvr:217.10.195.230
> callid:d4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146<callid%3Ad4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146>
> 2008-11-04 16:44:19,593 INFO rtpsessions:567 - AVF: created by SIP INVITE
> 2008-11-04 16:44:19,593 INFO sip:351 - INVITE: sender:10.199.110.146from:0365125362 RTP:
> 10.199.110.146,53460 to:0313165657 rcvr:217.10.195.230
> callid:d4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146<callid%3Ad4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146>
> 2008-11-04 16:44:19,593 INFO sip:351 - INVITE: sender:217.10.195.230from:0365125362 RTP:
> 10.199.110.146,53460 to:0313165657 rcvr:217.10.195.231
> callid:d4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146<callid%3Ad4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146>
> 2008-11-04 16:44:21,093 INFO sip:351 - INVITE: sender:217.10.195.232from:0766969918 RTP:
> 217.10.195.232,15468 to:0365125660 rcvr:217.10.195.230callid:0082-4CB1-6ED5ACA7-0@51E0E85426775DC8B
> 2008-11-04 16:44:21,093 INFO rtpsessions:567 - AVG: created by SIP INVITE
> 2008-11-04 16:44:21,109 INFO sip:351 - INVITE: sender:217.10.195.230from:0766969918 RTP:
> 217.10.195.232,15468 to:0365125660 rcvr:10.200.101.65callid:0082-4CB1-6ED5ACA7-0@51E0E85426775DC8B
> 2008-11-04 16:44:37,140 INFO sip:243 - BYE:
> callid:69ff6f139732187524e617c46fca9144@10.200.103.211<callid%3A69ff6f139732187524e617c46fca9144@10.200.103.211>
> 2008-11-04 16:44:37,140 INFO sip:243 - BYE:
> callid:69ff6f139732187524e617c46fca9144@10.200.103.211<callid%3A69ff6f139732187524e617c46fca9144@10.200.103.211>
> 2008-11-04 16:44:39,406 INFO sip:351 - INVITE: sender:10.200.103.110from:0365121528 RTP:
> 10.200.103.110,53456 to:03147947686 rcvr:217.10.195.230
> callid:8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110<callid%3A8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110>
> 2008-11-04 16:44:39,406 INFO rtpsessions:567 - AVH: created by SIP INVITE
> 2008-11-04 16:44:39,421 INFO sip:351 - INVITE: sender:10.200.103.110from:0365121528 RTP:
> 10.200.103.110,53456 to:03147947686 rcvr:217.10.195.230
> callid:8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110<callid%3A8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110>
> 2008-11-04 16:44:39,921 INFO sip:351 - INVITE: sender:217.10.195.230from:0365121528 RTP:
> 10.200.103.110,53456 to:03147947686 rcvr:217.10.195.231
> callid:8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110<callid%3A8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110>
> 2008-11-04 16:44:40,421 INFO sip:351 - INVITE: sender:10.201.100.232from:0365121644 RTP:
> 10.201.100.232,53458 to:0356465196 rcvr:217.10.195.230
> callid:c342facf227615e3408736080d0ee9da@10.201.100.232<callid%3Ac342facf227615e3408736080d0ee9da@10.201.100.232>
>
|
|
From: Henri H. <he...@or...> - 2008-11-04 15:46:55
|
This is pure SIP traffic, there is no RTP there, you are probably monitoring
your SIP proxy, not your PSTN gateway. Just use wireshark to figure out if
you have RTP or not on this link.
Henri
_____
From: wraith d [mailto:wra...@gm...]
Sent: 04 November 2008 10:22
To: ore...@li...
Subject: [Oreka-user] Hi all
Hello.
I`ve recently installed on an Windows Station the orekaudio software.
The windows pc has 2 ethernet card. One for accessing the network, and the
second for sniffing the voip traffic.
The second ethernet is connected in an Cisco 3524 switch in an span
interface.
I see the traffic between mediagateways and SIP endpoints, but i don`t get
any wav or native files.
Also my tapelist.log file is empty.
what could be the problem?
Here is an output from orkaudio.log file :
OrkAudio service starting
2008-11-04 16:43:38,109 INFO root:93 - Loaded plugin:
./plugins/RtpMixer.dll
2008-11-04 16:43:38,109 INFO immediateProcessing:53 - thread starting -
queue size:10000
2008-11-04 16:43:38,125 INFO root:87 - Loaded plugin:
audiocaptureplugins/VoIp.dll
2008-11-04 16:43:38,125 ERROR root:138 - Failed to start http server on
port:20000
2008-11-04 16:43:38,125 ERROR root:46 - Failed to start command line server
on port:10000
2008-11-04 16:43:38,125 INFO packet:835 - Initializing VoIP plugin
2008-11-04 16:43:38,187 INFO packet:753 - Available pcap devices:
2008-11-04 16:43:38,187 INFO packet:760 - *
\Device\NPF_GenericDialupAdapter - Generic dialup adapter
2008-11-04 16:43:38,203 INFO packet:760 - *
\Device\NPF_{9956A723-1254-42FB-B514-65A92D2D2F2F} - Intel(R) PRO/100 VE
Network Connection (Microsoft's Packet Scheduler)
2008-11-04 16:43:38,203 INFO packet:781 - Successfully opened device. pcap
handle:1362050
2008-11-04 16:43:38,218 INFO packet:760 - *
\Device\NPF_{3122178E-0562-427B-AC0C-D528B196AD02} - Intel(R) PRO/100 S
Desktop Adapter (Microsoft's Packet Scheduler)
2008-11-04 16:43:38,218 INFO packet:617 - Start Capturing: pcap
handle:1362050
2008-11-04 16:43:43,218 INFO sip:351 - INVITE: sender:192.168.0.63
from:0365128167 RTP:192.168.0.63,60000 to:01321314943 rcvr:217.10.195.230
callid:264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63
<mailto:callid%3A264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63>
2008-11-04 16:43:43,234 INFO rtpsessions:567 - AVD: created by SIP INVITE
2008-11-04 16:43:43,234 INFO sip:351 - INVITE: sender:192.168.0.63
from:0365128167 RTP:192.168.0.63,60000 to:01321314943 rcvr:217.10.195.230
callid:264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63
<mailto:callid%3A264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63>
2008-11-04 16:43:43,250 INFO sip:351 - INVITE: sender:217.10.195.230
from:0365128167 RTP:192.168.0.63,60000 to:01321314943 rcvr:217.10.195.231
callid:264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63
<mailto:callid%3A264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63>
2008-11-04 16:44:14,062 INFO sip:351 - INVITE: sender:10.201.100.232
from:0365121644 RTP:10.201.100.232,53456 to:0356465196 rcvr:217.10.195.230
callid:4b7fbcb65004e22124f96333643298f6@10.201.100.232
<mailto:callid%3A4b7fbcb65004e22124f96333643298f6@10.201.100.232>
2008-11-04 16:44:14,062 INFO rtpsessions:567 - AVE: created by SIP INVITE
2008-11-04 16:44:14,562 INFO sip:351 - INVITE: sender:10.201.100.232
from:0365121644 RTP:10.201.100.232,53456 to:0356465196 rcvr:217.10.195.230
callid:4b7fbcb65004e22124f96333643298f6@10.201.100.232
<mailto:callid%3A4b7fbcb65004e22124f96333643298f6@10.201.100.232>
2008-11-04 16:44:14,578 INFO sip:351 - INVITE: sender:217.10.195.230
from:0365121644 RTP:10.201.100.232,53456 to:0356465196 rcvr:217.10.195.231
callid:4b7fbcb65004e22124f96333643298f6@10.201.100.232
<mailto:callid%3A4b7fbcb65004e22124f96333643298f6@10.201.100.232>
2008-11-04 16:44:19,578 INFO sip:351 - INVITE: sender:10.199.110.146
from:0365125362 RTP:10.199.110.146,53460 to:0313165657 rcvr:217.10.195.230
callid:d4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146
<mailto:callid%3Ad4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146>
2008-11-04 16:44:19,593 INFO rtpsessions:567 - AVF: created by SIP INVITE
2008-11-04 16:44:19,593 INFO sip:351 - INVITE: sender:10.199.110.146
from:0365125362 RTP:10.199.110.146,53460 to:0313165657 rcvr:217.10.195.230
callid:d4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146
<mailto:callid%3Ad4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146>
2008-11-04 16:44:19,593 INFO sip:351 - INVITE: sender:217.10.195.230
from:0365125362 RTP:10.199.110.146,53460 to:0313165657 rcvr:217.10.195.231
callid:d4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146
<mailto:callid%3Ad4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146>
2008-11-04 16:44:21,093 INFO sip:351 - INVITE: sender:217.10.195.232
from:0766969918 RTP:217.10.195.232,15468 to:0365125660 rcvr:217.10.195.230
callid:0082-4CB1-6ED5ACA7-0@51E0E85426775DC8B
2008-11-04 16:44:21,093 INFO rtpsessions:567 - AVG: created by SIP INVITE
2008-11-04 16:44:21,109 INFO sip:351 - INVITE: sender:217.10.195.230
from:0766969918 RTP:217.10.195.232,15468 to:0365125660 rcvr:10.200.101.65
callid:0082-4CB1-6ED5ACA7-0@51E0E85426775DC8B
2008-11-04 16:44:37,140 INFO sip:243 - BYE:
callid:69ff6f139732187524e617c46fca9144@10.200.103.211
<mailto:callid%3A69ff6f139732187524e617c46fca9144@10.200.103.211>
2008-11-04 16:44:37,140 INFO sip:243 - BYE:
callid:69ff6f139732187524e617c46fca9144@10.200.103.211
<mailto:callid%3A69ff6f139732187524e617c46fca9144@10.200.103.211>
2008-11-04 16:44:39,406 INFO sip:351 - INVITE: sender:10.200.103.110
from:0365121528 RTP:10.200.103.110,53456 to:03147947686 rcvr:217.10.195.230
callid:8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110
<mailto:callid%3A8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110>
2008-11-04 16:44:39,406 INFO rtpsessions:567 - AVH: created by SIP INVITE
2008-11-04 16:44:39,421 INFO sip:351 - INVITE: sender:10.200.103.110
from:0365121528 RTP:10.200.103.110,53456 to:03147947686 rcvr:217.10.195.230
callid:8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110
<mailto:callid%3A8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110>
2008-11-04 16:44:39,921 INFO sip:351 - INVITE: sender:217.10.195.230
from:0365121528 RTP:10.200.103.110,53456 to:03147947686 rcvr:217.10.195.231
callid:8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110
<mailto:callid%3A8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110>
2008-11-04 16:44:40,421 INFO sip:351 - INVITE: sender:10.201.100.232
from:0365121644 RTP:10.201.100.232,53458 to:0356465196 rcvr:217.10.195.230
callid:c342facf227615e3408736080d0ee9da@10.201.100.232
<mailto:callid%3Ac342facf227615e3408736080d0ee9da@10.201.100.232>
|
|
From: wraith d <wra...@gm...> - 2008-11-04 15:21:51
|
Hello.
I`ve recently installed on an Windows Station the orekaudio software.
The windows pc has 2 ethernet card. One for accessing the network, and the
second for sniffing the voip traffic.
The second ethernet is connected in an Cisco 3524 switch in an span
interface.
I see the traffic between mediagateways and SIP endpoints, but i don`t get
any wav or native files.
Also my tapelist.log file is empty.
what could be the problem?
Here is an output from orkaudio.log file :
OrkAudio service starting
2008-11-04 16:43:38,109 INFO root:93 - Loaded plugin:
./plugins/RtpMixer.dll
2008-11-04 16:43:38,109 INFO immediateProcessing:53 - thread starting -
queue size:10000
2008-11-04 16:43:38,125 INFO root:87 - Loaded plugin:
audiocaptureplugins/VoIp.dll
2008-11-04 16:43:38,125 ERROR root:138 - Failed to start http server on
port:20000
2008-11-04 16:43:38,125 ERROR root:46 - Failed to start command line server
on port:10000
2008-11-04 16:43:38,125 INFO packet:835 - Initializing VoIP plugin
2008-11-04 16:43:38,187 INFO packet:753 - Available pcap devices:
2008-11-04 16:43:38,187 INFO packet:760 - *
\Device\NPF_GenericDialupAdapter - Generic dialup adapter
2008-11-04 16:43:38,203 INFO packet:760 - *
\Device\NPF_{9956A723-1254-42FB-B514-65A92D2D2F2F} - Intel(R) PRO/100 VE
Network Connection (Microsoft's Packet Scheduler)
2008-11-04 16:43:38,203 INFO packet:781 - Successfully opened device. pcap
handle:1362050
2008-11-04 16:43:38,218 INFO packet:760 - *
\Device\NPF_{3122178E-0562-427B-AC0C-D528B196AD02} - Intel(R) PRO/100 S
Desktop Adapter (Microsoft's Packet Scheduler)
2008-11-04 16:43:38,218 INFO packet:617 - Start Capturing: pcap
handle:1362050
2008-11-04 16:43:43,218 INFO sip:351 - INVITE:
sender:192.168.0.63from:0365128167 RTP:
192.168.0.63,60000 to:01321314943 rcvr:217.10.195.230
callid:264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63<callid%3A264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63>
2008-11-04 16:43:43,234 INFO rtpsessions:567 - AVD: created by SIP INVITE
2008-11-04 16:43:43,234 INFO sip:351 - INVITE:
sender:192.168.0.63from:0365128167 RTP:
192.168.0.63,60000 to:01321314943 rcvr:217.10.195.230
callid:264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63<callid%3A264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63>
2008-11-04 16:43:43,250 INFO sip:351 - INVITE:
sender:217.10.195.230from:0365128167 RTP:
192.168.0.63,60000 to:01321314943 rcvr:217.10.195.231
callid:264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63<callid%3A264bdb8b3e3a96517862bfc11b1b55a6@192.168.0.63>
2008-11-04 16:44:14,062 INFO sip:351 - INVITE:
sender:10.201.100.232from:0365121644 RTP:
10.201.100.232,53456 to:0356465196 rcvr:217.10.195.230
callid:4b7fbcb65004e22124f96333643298f6@10.201.100.232<callid%3A4b7fbcb65004e22124f96333643298f6@10.201.100.232>
2008-11-04 16:44:14,062 INFO rtpsessions:567 - AVE: created by SIP INVITE
2008-11-04 16:44:14,562 INFO sip:351 - INVITE:
sender:10.201.100.232from:0365121644 RTP:
10.201.100.232,53456 to:0356465196 rcvr:217.10.195.230
callid:4b7fbcb65004e22124f96333643298f6@10.201.100.232<callid%3A4b7fbcb65004e22124f96333643298f6@10.201.100.232>
2008-11-04 16:44:14,578 INFO sip:351 - INVITE:
sender:217.10.195.230from:0365121644 RTP:
10.201.100.232,53456 to:0356465196 rcvr:217.10.195.231
callid:4b7fbcb65004e22124f96333643298f6@10.201.100.232<callid%3A4b7fbcb65004e22124f96333643298f6@10.201.100.232>
2008-11-04 16:44:19,578 INFO sip:351 - INVITE:
sender:10.199.110.146from:0365125362 RTP:
10.199.110.146,53460 to:0313165657 rcvr:217.10.195.230
callid:d4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146<callid%3Ad4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146>
2008-11-04 16:44:19,593 INFO rtpsessions:567 - AVF: created by SIP INVITE
2008-11-04 16:44:19,593 INFO sip:351 - INVITE:
sender:10.199.110.146from:0365125362 RTP:
10.199.110.146,53460 to:0313165657 rcvr:217.10.195.230
callid:d4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146<callid%3Ad4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146>
2008-11-04 16:44:19,593 INFO sip:351 - INVITE:
sender:217.10.195.230from:0365125362 RTP:
10.199.110.146,53460 to:0313165657 rcvr:217.10.195.231
callid:d4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146<callid%3Ad4e64b4bdaf1ddd71727e191fb1924fb@10.199.110.146>
2008-11-04 16:44:21,093 INFO sip:351 - INVITE:
sender:217.10.195.232from:0766969918 RTP:
217.10.195.232,15468 to:0365125660
rcvr:217.10.195.230callid:0082-4CB1-6ED5ACA7-0@51E0E85426775DC8B
2008-11-04 16:44:21,093 INFO rtpsessions:567 - AVG: created by SIP INVITE
2008-11-04 16:44:21,109 INFO sip:351 - INVITE:
sender:217.10.195.230from:0766969918 RTP:
217.10.195.232,15468 to:0365125660
rcvr:10.200.101.65callid:0082-4CB1-6ED5ACA7-0@51E0E85426775DC8B
2008-11-04 16:44:37,140 INFO sip:243 - BYE:
callid:69ff6f139732187524e617c46fca9144@10.200.103.211<callid%3A69ff6f139732187524e617c46fca9144@10.200.103.211>
2008-11-04 16:44:37,140 INFO sip:243 - BYE:
callid:69ff6f139732187524e617c46fca9144@10.200.103.211<callid%3A69ff6f139732187524e617c46fca9144@10.200.103.211>
2008-11-04 16:44:39,406 INFO sip:351 - INVITE:
sender:10.200.103.110from:0365121528 RTP:
10.200.103.110,53456 to:03147947686 rcvr:217.10.195.230
callid:8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110<callid%3A8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110>
2008-11-04 16:44:39,406 INFO rtpsessions:567 - AVH: created by SIP INVITE
2008-11-04 16:44:39,421 INFO sip:351 - INVITE:
sender:10.200.103.110from:0365121528 RTP:
10.200.103.110,53456 to:03147947686 rcvr:217.10.195.230
callid:8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110<callid%3A8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110>
2008-11-04 16:44:39,921 INFO sip:351 - INVITE:
sender:217.10.195.230from:0365121528 RTP:
10.200.103.110,53456 to:03147947686 rcvr:217.10.195.231
callid:8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110<callid%3A8282c6d48c9b6fd0c7e5ab930eb6e4ac@10.200.103.110>
2008-11-04 16:44:40,421 INFO sip:351 - INVITE:
sender:10.201.100.232from:0365121644 RTP:
10.201.100.232,53458 to:0356465196 rcvr:217.10.195.230
callid:c342facf227615e3408736080d0ee9da@10.201.100.232<callid%3Ac342facf227615e3408736080d0ee9da@10.201.100.232>
|
|
From: Henri H. <he...@or...> - 2008-11-03 15:23:34
|
Don, This plugin should work but has barely been used by anyone. If it generates lots of files, it is probably because the VAD is badly tuned. Here are the config parms of interest to you: #define AUDIO_SEGMENTATION_PARAM "AudioSegmentation" #define AUDIO_SEGMENTATION_DEFAULT false #define AUDIO_SEGMENT_DURATION_PARAM "AudioSegmentDuration" #define AUDIO_SEGMENT_DURATION_DEFAULT 60 #define LOG_RMS_PARAM "LogRms" #define LOG_RMS_DEFAULT false #define VAD_PARAM "VAD" #define VAD_DEFAULT false #define VAD_HIGH_THRESHOLD_DB_PARAM "VadHighThresholdDb" #define VAD_HIGH_THRESHOLD_DB_DEFAULT -12.2 #define VAD_LOW_THRESHOLD_DB_PARAM "VadLowThresholdDb" #define VAD_LOW_THRESHOLD_DB_DEFAULT -12.5 #define VAD_HOLD_ON_SEC_PARAM "VadHoldOnSec" #define VAD_HOLD_ON_SEC_DEFAULT 4 Henri _____ From: Don Fletcher [mailto:dfl...@Co...] Sent: 31 October 2008 18:12 To: ore...@li... Subject: [Oreka-user] Setting up the SoundDevice section installed oreka on windows laptop all default commented out voip and generator plugin sections, and made these changes <StorageAudioFormat>pcmwav</StorageAudioFormat> <EnableReporting>false</EnableReporting> <AudioSegmentation>false</AudioSegmentation> <AudioSegmentDuration>10</AudioSegmentDuration> <TapeFileNameing>Norm_Sacramento-</TapeFileNameing> <SoundDevicePlugin> <SampleRate>8000</SampleRate> </SoundDevicePlugin> when I started it, it would write files to the ./audiorecordings/2008/mo/day/hr/ dir they were unusable; windows though they were gsm encoded, that is what it said on the properties tab, a run of the program for just a few minutes, generated 100's of files. they all had port[0-9] in the filename. So -- 1) how should the sounddevice recordings behave? How does it tell when a call is happening? It it going to genereate 1000's of files/hour that need to be joined? 2) are there other options in the sounddeviceplug seciont that I can /should set? 3) is my syntax correct on SoundDevicePlugin seciont? Thanks in advance. Don |
|
From: Henri H. <he...@or...> - 2008-11-03 15:19:34
|
Don, Oreka is routinely being run on PIII class machines. Performance entirely depends on the specific machine you have but you might be able to achieve your number with enough RAM and a fast hard drive. Henri -----Original Message----- From: Don Fletcher [mailto:dfl...@Co...] Sent: 02 November 2008 03:20 To: ore...@li... Subject: [Oreka-user] Min. System Requirements How small of a computer can this Oreka run on? Say I want to record 20-30 simultaneous voip calls -- is the RAM/Processor/HD more where the bottle neck will be. Could this run on something like a pentium 3 class machine? or a Via C7 processor? would a C7 system w/1 gb of ram be adequate to run just orkaudio? Thanks Don ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ Oreka-user mailing list Ore...@li... https://lists.sourceforge.net/lists/listinfo/oreka-user |
|
From: Don F. <dfl...@Co...> - 2008-11-02 07:19:57
|
How small of a computer can this Oreka run on? Say I want to record 20-30 simultaneous voip calls -- is the RAM/Processor/HD more where the bottle neck will be. Could this run on something like a pentium 3 class machine? or a Via C7 processor? would a C7 system w/1 gb of ram be adequate to run just orkaudio? Thanks Don |
|
From: Don F. <dfl...@Co...> - 2008-10-31 22:12:11
|
installed oreka on windows laptop all default commented out voip and generator plugin sections, and made these changes <StorageAudioFormat>pcmwav</StorageAudioFormat> <EnableReporting>false</EnableReporting> <AudioSegmentation>false</AudioSegmentation> <AudioSegmentDuration>10</AudioSegmentDuration> <TapeFileNameing>Norm_Sacramento-</TapeFileNameing> <SoundDevicePlugin> <SampleRate>8000</SampleRate> </SoundDevicePlugin> when I started it, it would write files to the ./audiorecordings/2008/mo/day/hr/ dir they were unusable; windows though they were gsm encoded, that is what it said on the properties tab, a run of the program for just a few minutes, generated 100's of files. they all had port[0-9] in the filename. So -- 1) how should the sounddevice recordings behave? How does it tell when a call is happening? It it going to genereate 1000's of files/hour that need to be joined? 2) are there other options in the sounddeviceplug seciont that I can /should set? 3) is my syntax correct on SoundDevicePlugin seciont? Thanks in advance. Don |
|
From: Henri H. <he...@or...> - 2008-10-30 17:10:31
|
This is the line that makes this happen:
session->ReportSipInvite(invite);
Henri
-----Original Message-----
From: wh...@wo... [mailto:wh...@wo...]
Sent: 30 October 2008 13:07
To: he...@or...; ore...@li...
Subject: Re: [Oreka-user] SIP Proxy obfuscation
>From RtpSession.cpp under voip under audiocaptureplugins under
Oreka-0.5-313, you have the following:
void RtpSessions::ReportSipInvite(SipInviteInfoRef& invite)
{
char szFromRtpIp[16];
ACE_OS::inet_ntop(AF_INET, (void*)&invite->m_fromRtpIp, szFromRtpIp,
sizeof(szFromRtpIp));
CStdString ipAndPort = CStdString(szFromRtpIp) + "," +
invite->m_fromRtpPort;
std::map<CStdString, RtpSessionRef>::iterator pair;
pair = m_byIpAndPort.find(ipAndPort);
if (pair != m_byIpAndPort.end())
{
// The session already exists, do nothing
return;
}
pair = m_byCallId.find(invite->m_callId);
if (pair != m_byCallId.end())
{
// The session already exists
RtpSessionRef session = pair->second;
if(!session->m_ipAndPort.Equals(ipAndPort))
{
// The session RTP connection address has changed
// Remove session from IP and Port map
m_byIpAndPort.erase(session->m_ipAndPort);
// ... update
session->m_ipAndPort = ipAndPort;
session->ReportSipInvite(invite);
// ... and reinsert
m_byIpAndPort.insert(std::make_pair(session->m_ipAndPort, session));
LOG4CXX_INFO(m_log, session->m_trackingId + ":
updated with new INVITE data");
}
return;
}
// create new session and insert into both maps
CStdString trackingId = alphaCounter.GetNext();
RtpSessionRef session(new RtpSession(trackingId));
session->m_ipAndPort = ipAndPort;
session->m_callId = invite->m_callId;
session->m_protocol = RtpSession::ProtSip;
session->ReportSipInvite(invite);
m_byIpAndPort.insert(std::make_pair(session->m_ipAndPort, session));
m_byCallId.insert(std::make_pair(session->m_callId, session));
CStdString numSessions = IntToString(m_byIpAndPort.size());
LOG4CXX_DEBUG(m_log, CStdString("ByIpAndPort: ") + numSessions);
LOG4CXX_INFO(m_log, trackingId + ": created by SIP INVITE");
}
I can't seem to find where you are grabbing the from: and the to: and not
the via.
In fact it seems like this line, session->m_ipAndPort = ipAndPort is pulling
only the ip and port number and not actually grabbing the ani@ip.
-wh
>Oreka reports from: and to: by default, not via. You might have another
>problem. I can take a look if you send me a test trace of a call offline.
>Henri
>
>-----Original Message-----
>From: wh...@wo... [mailto:wh...@wo...]
>Sent: 29 October 2008 19:57
>To: ore...@li...
>Subject: [Oreka-user] SIP Proxy obfuscation
>
>I have noticed the following problem.
>
>Should a caller dial from outside a sip proxy (a pbx) to an extension
within
>the sip proxy lan, Oreka does not display the ani specified in From: phrase
>because it is looking at the Via phrase, which in most sip proxies does not
>display the ani.
>
>So all orkaudio shows is the sip proxy ip address for all inbound calls.
>
>The config file under orkaudio has a specification for MediaGateways.
>
>It would be nice if orkaudio had logic that understood that if an IP packet
>had a via phrase with an ip address of the media gateway then orkaudio
would
>then look in the from phrase to pickup the ani information (and even the ip
>address associated with that ani outside of the sip proxy.)
>
>Just a suggestion. If you don't do it, I might.
>
>wh
>
>-------------------------------------------------------------------------
>This SF.Net email is sponsored by the Moblin Your Move Developer's
challenge
>Build the coolest Linux based applications with Moblin SDK & win great
>prizes
>Grand prize is a trip for two to an Open Source event anywhere in the world
>http://moblin-contest.org/redirect.php?banner_id=100&url=/
>_______________________________________________
>Oreka-user mailing list
>Ore...@li...
>https://lists.sourceforge.net/lists/listinfo/oreka-user
>
-------------------------------------------------------------------------
This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the coolest Linux based applications with Moblin SDK & win great
prizes
Grand prize is a trip for two to an Open Source event anywhere in the world
http://moblin-contest.org/redirect.php?banner_id=100&url=/
_______________________________________________
Oreka-user mailing list
Ore...@li...
https://lists.sourceforge.net/lists/listinfo/oreka-user
|
|
From: <wh...@wo...> - 2008-10-30 17:07:07
|
>From RtpSession.cpp under voip under audiocaptureplugins under Oreka-0.5-313, you have the following:
void RtpSessions::ReportSipInvite(SipInviteInfoRef& invite)
{
char szFromRtpIp[16];
ACE_OS::inet_ntop(AF_INET, (void*)&invite->m_fromRtpIp, szFromRtpIp, sizeof(szFromRtpIp));
CStdString ipAndPort = CStdString(szFromRtpIp) + "," + invite->m_fromRtpPort;
std::map<CStdString, RtpSessionRef>::iterator pair;
pair = m_byIpAndPort.find(ipAndPort);
if (pair != m_byIpAndPort.end())
{
// The session already exists, do nothing
return;
}
pair = m_byCallId.find(invite->m_callId);
if (pair != m_byCallId.end())
{
// The session already exists
RtpSessionRef session = pair->second;
if(!session->m_ipAndPort.Equals(ipAndPort))
{
// The session RTP connection address has changed
// Remove session from IP and Port map
m_byIpAndPort.erase(session->m_ipAndPort);
// ... update
session->m_ipAndPort = ipAndPort;
session->ReportSipInvite(invite);
// ... and reinsert
m_byIpAndPort.insert(std::make_pair(session->m_ipAndPort, session));
LOG4CXX_INFO(m_log, session->m_trackingId + ": updated with new INVITE data");
}
return;
}
// create new session and insert into both maps
CStdString trackingId = alphaCounter.GetNext();
RtpSessionRef session(new RtpSession(trackingId));
session->m_ipAndPort = ipAndPort;
session->m_callId = invite->m_callId;
session->m_protocol = RtpSession::ProtSip;
session->ReportSipInvite(invite);
m_byIpAndPort.insert(std::make_pair(session->m_ipAndPort, session));
m_byCallId.insert(std::make_pair(session->m_callId, session));
CStdString numSessions = IntToString(m_byIpAndPort.size());
LOG4CXX_DEBUG(m_log, CStdString("ByIpAndPort: ") + numSessions);
LOG4CXX_INFO(m_log, trackingId + ": created by SIP INVITE");
}
I can't seem to find where you are grabbing the from: and the to: and not the via.
In fact it seems like this line, session->m_ipAndPort = ipAndPort is pulling only the ip and port number and not actually grabbing the ani@ip.
-wh
>Oreka reports from: and to: by default, not via. You might have another
>problem. I can take a look if you send me a test trace of a call offline.
>Henri
>
>-----Original Message-----
>From: wh...@wo... [mailto:wh...@wo...]
>Sent: 29 October 2008 19:57
>To: ore...@li...
>Subject: [Oreka-user] SIP Proxy obfuscation
>
>I have noticed the following problem.
>
>Should a caller dial from outside a sip proxy (a pbx) to an extension within
>the sip proxy lan, Oreka does not display the ani specified in From: phrase
>because it is looking at the Via phrase, which in most sip proxies does not
>display the ani.
>
>So all orkaudio shows is the sip proxy ip address for all inbound calls.
>
>The config file under orkaudio has a specification for MediaGateways.
>
>It would be nice if orkaudio had logic that understood that if an IP packet
>had a via phrase with an ip address of the media gateway then orkaudio would
>then look in the from phrase to pickup the ani information (and even the ip
>address associated with that ani outside of the sip proxy.)
>
>Just a suggestion. If you don't do it, I might.
>
>wh
>
>-------------------------------------------------------------------------
>This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
>Build the coolest Linux based applications with Moblin SDK & win great
>prizes
>Grand prize is a trip for two to an Open Source event anywhere in the world
>http://moblin-contest.org/redirect.php?banner_id=100&url=/
>_______________________________________________
>Oreka-user mailing list
>Ore...@li...
>https://lists.sourceforge.net/lists/listinfo/oreka-user
>
|
|
From: Henri H. <he...@or...> - 2008-10-30 15:08:23
|
Oreka reports from: and to: by default, not via. You might have another problem. I can take a look if you send me a test trace of a call offline. Henri -----Original Message----- From: wh...@wo... [mailto:wh...@wo...] Sent: 29 October 2008 19:57 To: ore...@li... Subject: [Oreka-user] SIP Proxy obfuscation I have noticed the following problem. Should a caller dial from outside a sip proxy (a pbx) to an extension within the sip proxy lan, Oreka does not display the ani specified in From: phrase because it is looking at the Via phrase, which in most sip proxies does not display the ani. So all orkaudio shows is the sip proxy ip address for all inbound calls. The config file under orkaudio has a specification for MediaGateways. It would be nice if orkaudio had logic that understood that if an IP packet had a via phrase with an ip address of the media gateway then orkaudio would then look in the from phrase to pickup the ani information (and even the ip address associated with that ani outside of the sip proxy.) Just a suggestion. If you don't do it, I might. wh ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ Oreka-user mailing list Ore...@li... https://lists.sourceforge.net/lists/listinfo/oreka-user |
143 messages has been excluded from this view by a project administrator.