Re: [asterisk-users] Realtime changes not reflected realtime
Jonathan, 'sip show peers' works just fine... asterisk*CLI sip show peers Name/username Host Dyn Nat ACL Port Status Realtime testcorp4 (Unspecified) D N 0 UNREACHABLE Cached RT testcorp3/testcorp3 192.168.1.100 D N 5061 OK (25 ms) Cached RT Only you see the 'Realtime'-column, and the 'Cached RT'. Jonathan Thurman wrote: I know that "sip show peers" doesn't work, and I believe that qualify does not work without caching (but I haven't tested that). I enable caching because I don't change the names of sip_accounts that frequently, and why have Asterisk hit the database constantly if you aren't changing the information? Asterisk will then save all of the results in RAM, and only do a look-up for an unknown account. If you have a web interface for updating information you could always use AMI to issue the prune/reload after committing the changes. -Jonathan -- _ -- 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] Bug or feature: cdr_odbc.conf.sample
Hello, From asterisk-1.6.1.18/configs/cdr_odbc.conf.sample : ; ; cdr_odbc.conf ; ;[global] ;dsn=MySQL-test ;username=username ;password=password ;loguniqueid=yes ;dispositionstring=yes ;table=cdr ;cdr is default table name ;usegmtime=no ; set to yes to log in GMT Though, reading from https://issues.asterisk.org/view.php?id=15021, it seems that lines username= and password= in cdr_odbc.conf are not used anymore (the fields in res_odbc.conf are used instead). My question are : 1. Are those lines still used in 1.6.1.X ? 2. If those lines are not used anymore, would you think more appropriate to remove them from cdr_odbc.conf.sample ? Regards -- _ -- 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] Amazon EC2 SIP floods - you can help
Hi, We all know most people are reporting that Amazon hasn't been helpful at all. A few people say they've received answers, but most are getting smoke screen PR BS. You can vote this up on Slashdot, send the message: SIP Attacks From Amazon EC2 Going Unaddressed: http://bit.ly/bOkNNx Send this message out to Amazon, I am positive that once it reaches the right person, they will do the right thing. The more places you send links to, the more likely it is that someone will wake up. We are a minority, unlike Toyota or airline customers. We need to yell louder. If you are on any social networks at all, Facebook, Twitter, Linkedin or vertical networks that are visible on Google, please consider posting at least a line about the lack of Amazon cooperation or a link to one of the many articles about this issue. If nothing is done, all cloud providers will do nothing and it'll become a bigger nightmare. If Amazon sets the standard for cooperation, the others will likely need to follow. Be sure to include Amazon EC2 once or more in any message you send, as this is what makes it rise to the top. The EC2 robot on Twitter is even stupidly repeating all the complaints :) There's a VUC discussion you can listen to on your commute to work: http://vuc.li/9n7Qxl /r -- _ -- 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] Amazon EC2 SIP floods - you can help
On Sun, Apr 18, 2010 at 12:10:32PM +0200, Randy R wrote: Hi, We all know most people are reporting that Amazon hasn't been helpful at all. A few people say they've received answers, but most are getting smoke screen PR BS. You can vote this up on Slashdot, send the message: SIP Attacks From Amazon EC2 Going Unaddressed: http://bit.ly/bOkNNx It seems that at least Slashdot is responsive: http://it.slashdot.org/story/10/04/17/2059256/SIP-Attacks-From-Amazon-EC2-Going-Unaddressed -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-stat - Bugs
Hi, With latest asterisk-stat (2.0.1) : 1. In call-log.php file, there are lines with src=images/clear.gif but there is no such images/clear.gif file. This produces : [Sun Apr 18 14:03:14 2010] [error] [client 192.168.102.102] File does not exist: /var/www/asterisk-stat/images/clear.gif, referer: http://192.168.102.240/asterisk-stat/cdr.php?s=1 Has someone found an elegant workaround ? 2. I'm also getting : [Sun Apr 18 14:13:30 2010] [error] [client 192.168.102.102] File does not exist: /var/www/css, referer: http://192.168.102.240/asterisk-stat/cdr.php?s=1t=order=calldatesens=DESCcurrent_page=0 Looking at cdr.php, I could remove this error changing /css/print.css to images/print.css. Suggestions ? Regards. -- _ -- 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] Slightly OT: OMA DM Solution
Hi, anyone knows an open OMA DM tool that would be able to configure Nokia phones (mainly the sip-stuff of the e-series) for use with asterisk? Anything open i could find was the device manager from funambol, which was last updated in 2006 :-( Regards, Jay -- _ -- 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] Amazon EC2 SIP floods - you can help
Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Apr 18, 2010 at 12:10:32PM +0200, Randy R wrote: Hi, We all know most people are reporting that Amazon hasn't been helpful at all. A few people say they've received answers, but most are getting smoke screen PR BS. You can vote this up on Slashdot, send the message: SIP Attacks From Amazon EC2 Going Unaddressed: http://bit.ly/bOkNNx It seems that at least Slashdot is responsive: http://it.slashdot.org/story/10/04/17/2059256/SIP-Attacks-From-Amazon-EC2-Going-Unaddressed -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Amazon EC2 SIP floods - you can help
On Sun, Apr 18, 2010 at 1:56 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: It seems that at least Slashdot is responsive: http://it.slashdot.org/story/10/04/17/2059256/SIP-Attacks-From-Amazon-EC2-Going-Unaddressed Yes, there's a lot of talk here, some of it sympathetic, some less so, but at least there's discussion. I expect a response from Amazon at some point, but not until the visibility level becomes painful. One other person in the Slashdot posts threatens to stop buying things from Amazon. More of those will likely surface. I a related question, if the IP addresses were spoofed, how could a response be directed back? Don't the register attempts, because they need a response necessarily carry the correct source IP? /r -- _ -- 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] Realtime changes not reflected realtime
On Sun, Apr 18, 2010 at 12:30 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Jonathan, 'sip show peers' works just fine... Sorry, I wasn't clear. It has been my experience in 1.6.1.x that 'sip show peers' does not work without rtcachefriends=yes for realtime implementations. asterisk*CLI sip show peers Name/username Host Dyn Nat ACL Port Status Realtime testcorp4 (Unspecified) D N 0 UNREACHABLE Cached RT testcorp3/testcorp3 192.168.1.100 D N 5061 OK (25 ms) Cached RT Only you see the 'Realtime'-column, and the 'Cached RT'. With rtcachefriends enabled it does show if the peer is Cached RT or a static peer in sip.conf. -Jonathan -- _ -- 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] Amazon EC2 SIP floods - you can help
-BEGIN PGP SIGNED MESSAGE- Hash: SHA256 Randy R wrote: On Sun, Apr 18, 2010 at 1:56 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: It seems that at least Slashdot is responsive: http://it.slashdot.org/story/10/04/17/2059256/SIP-Attacks-From-Amazon-EC2-Going-Unaddressed Yes, there's a lot of talk here, some of it sympathetic, some less so, but at least there's discussion. I expect a response from Amazon at some point, but not until the visibility level becomes painful. One other person in the Slashdot posts threatens to stop buying things from Amazon. More of those will likely surface. I a related question, if the IP addresses were spoofed, how could a response be directed back? Don't the register attempts, because they need a response necessarily carry the correct source IP? /r Yes, If the IP addresses were spoofed, it would be simply a DoS attack. Stu -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.10 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIcBAEBCAAGBQJLyyd4AAoJEFKVLITDJSGSnZQQAJTR/VNudd6ZIsTYX3mxlGRC 0l85Z31Jh0ek0u+eNceuQc430yqFHWnK79Bvun/PK7Fz6RbiY5v+h3L5gkNMLpFy i21qTLJGua0MtbPPh3VktzHRle4r4Ph/darbMmwpUCtnBq38cjzTJvpDXIgtNwW/ yHhXIgEsQBhqs5xsPB9yZoPK7hBR3i9gmMi4aNbXg+mHIq7oYEe+ko1U0J/vjHXg 8zH78hzl6RZJfXFAoVb29htt+zII4A+SAH9fZAr72L3IOY4FCRYonT0ttktmTsMk V+wLXJyU2CnUb9w7v5DDG+EgdVrQFxdqN3pGuxfIXBvzu4bC6NAIRpfqqBV4yAAu TmR1+bCOCSbf5giBXJP77HobghzoOsRxhZVa7TMp5OKWuKa+v4zbpwY7YJRMDYOb vd42mSEWlOMr+he0SUnSNNusvqB0jmWIt/8lWEZl4/prpnyym0TWun7Z4Z0GjJLx +OHJLFEwo0T0f2Vf+og1wDBW9e/Tf2c1l2w6SrVHTL+0Hz3+2sh8pt389PPcQmuD PLqmG5mJC1ZuNhYhTknn1mT/NCtpgV4RrLJRcHM/46noEKAdy3DGs79Uwzw0jj3v XUaGRgEpGYz85YF0WaxMnOkSTMW1pgg8sTirHCGZttTcTHybJV2tDfEEEedx11Z5 jPjnd1SbFqhNG9KEsAGr =b2RY -END PGP SIGNATURE- -- _ -- 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] Amazon EC2 SIP floods - you can help
On Sun, Apr 18, 2010 at 5:38 PM, Stuart Sheldon s...@actusa.net wrote: I a related question, if the IP addresses were spoofed, how could a response be directed back? Don't the register attempts, because they If the IP addresses were spoofed, it would be simply a DoS attack. This is what I thought, so when people say yeah, but they could be spoofed this isn't a valid argument. A huge number of requests going to your server with an originating EC2 IP needs to be shut down first, questions asked after. Only Amazon can fix this. They have not only the IP info but also full customer data, including banking info. What possible excuse can they provide? Is this why they are silent? There's no good excuse other than, it would cut into our profits. Maybe we could get GigaOm interested or some other high-visibility blog. /r -- _ -- 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] Amazon EC2 SIP floods - you can help
-BEGIN PGP SIGNED MESSAGE- Hash: SHA256 Randy R wrote: On Sun, Apr 18, 2010 at 5:38 PM, Stuart Sheldon s...@actusa.net wrote: I a related question, if the IP addresses were spoofed, how could a response be directed back? Don't the register attempts, because they If the IP addresses were spoofed, it would be simply a DoS attack. This is what I thought, so when people say yeah, but they could be spoofed this isn't a valid argument. A huge number of requests going to your server with an originating EC2 IP needs to be shut down first, questions asked after. Only Amazon can fix this. They have not only the IP info but also full customer data, including banking info. What possible excuse can they provide? Is this why they are silent? There's no good excuse other than, it would cut into our profits. Maybe we could get GigaOm interested or some other high-visibility blog. /r For what it's worth, here is my Blog Article from the incident... http://www.stuartsheldon.org/blog/2010/04/sip-brute-force-attack-originating-from-amazon-ec2-hosts/ Stu -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.10 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIcBAEBCAAGBQJLyy0SAAoJEFKVLITDJSGSMaYQAKVTy6en4zsbekcjXTSjMo6z SSwBL95mSpgGRU6nAOKIjs5UUczFS8MtReag7hqW7e1ZtwwlXz88KP+c7yNZVw9+ 6HIjAf+PdaxRmDQ/bUpcXy+4Nnl6RRzVnE5oY33/ZWJrAjBfLb/eQCFQOqAdgxDr xsTGCPts/CJWeQrni6g4pdYFf3P4BvxsyoGw5vpF8rXipujaK1V0zxT6dE+XDNYZ aqrLlZtGvF7oTLtYCAt6g/C7VG7RJDNbuxGKG0q8GfHeU3xXEjYytH6jq26yiCSi FvP6vH0CzOInyYohPEXuxej2rLADf6c3JqXidadXX87l5XLb947pooMK+gmyRv8m AjsoOryMs43V48q5y1F25LVV8pnw83xEUZyxfa4/JNx4Fr4PvuMdVs0UDZbjWdCD ncf47IVQKztWfM3vcbyFXyfgDHrAnGUwZ/VxPpQ9/0VGsrC8V9rujQCI3UVk2/7v RHFK97ddmPvrAr8Gml+wnjTROSyY5n8ds762ZfyN3rel7e7w5gynpa+G9pcNqgSX MzdKRiC10hF4X6ZMXOski1UIXm+x7r+8uY8p+/8l6A4sdXohCUhXTcYLMnDBzgob fsmxb6WKKkaGTLv7jWLukfZVYcppk+B4M8hFgAvVqMWBRI3eZmZTKvmzDs9yjaqw kcF4NwJOpLXsG3w9vs7F =kLEJ -END PGP SIGNATURE- -- _ -- 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] Amazon EC2 SIP floods - you can help
On Apr 18, 2010, at 1:14 PM, Randy R wrote: On Sun, Apr 18, 2010 at 6:02 PM, Stuart Sheldon s...@actusa.net wrote: For what it's worth, here is my Blog Article from the incident... http://www.stuartsheldon.org/blog/2010/04/sip-brute-force-attack-originating-from-amazon-ec2-hosts/ Saw it early on Stu, and quoted your excellent summary: “I’m sorry, you have reached a company that doesn’t care that we are attacking you…” /r There's also a link to it from the VoIP Tech Chat article. ---fred http://qxork.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] Amazon EC2 SIP floods - you can help
On Sun, Apr 18, 2010 at 6:02 PM, Stuart Sheldon s...@actusa.net wrote: For what it's worth, here is my Blog Article from the incident... http://www.stuartsheldon.org/blog/2010/04/sip-brute-force-attack-originating-from-amazon-ec2-hosts/ Saw it early on Stu, and quoted your excellent summary: “I’m sorry, you have reached a company that doesn’t care that we are attacking you…” /r -- _ -- 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] Amazon EC2 SIP floods - you can help
On Sun, Apr 18, 2010 at 7:17 PM, Fred Posner f...@teamforrest.com wrote: There's also a link to it from the VoIP Tech Chat article. And we are also linking to Fred's original story which says it all about Amazon: http://www.voiptechchat.com/voip/457/amazon-ec2-sip-brute-force-attacks-on-rise/ It has been suggested that posters on Twitter use hashtags. I'm not big on them myself, but #EC2, #SIP and #amazon might be appropriate. /r -- _ -- 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] Bug or feature: cdr_odbc.conf.sample
On Sunday 18 April 2010 04:10:11 Olivier wrote: From asterisk-1.6.1.18/configs/cdr_odbc.conf.sample : ; ; cdr_odbc.conf ; ;[global] ;dsn=MySQL-test ;username=username ;password=password ;loguniqueid=yes ;dispositionstring=yes ;table=cdr ;cdr is default table name ;usegmtime=no ; set to yes to log in GMT Though, reading from https://issues.asterisk.org/view.php?id=15021, it seems that lines username= and password= in cdr_odbc.conf are not used anymore (the fields in res_odbc.conf are used instead). My question are : 1. Are those lines still used in 1.6.1.X ? No. 2. If those lines are not used anymore, would you think more appropriate to remove them from cdr_odbc.conf.sample ? Removed, as of revision 257770. Thank you for the reminder. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] problems originating an outgoing IAX2 call
Dear all
i'm trying to originate an outgoing call with the command originate,
from Asterisk's CLI i'm typing:
CLI originate IAX2/my-iax-provider/number2call application wait 10
[Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr:
prepending 40 to prefs
-- Call accepted by 62.149.202.150 (format ilbc)
-- Format for call is ilbc
-- Hungup 'IAX2/my-iax-provider-5647'
... and nothing happend the hangup is given after 3-4 seconds of the command
But, if i try to call a dialplan extenstion from a local IAX user the
call works properly
[outgoing_voipvoice]
exten = _X.,1,Dial(IAX2/my-iax-provider/${EXTEN})
-- Accepting AUTHENTICATED call from 82.56.46.69:
requested format = gsm,
requested prefs = (),
actual format = gsm,
host prefs = (),
priority = mine
-- Executing [number2c...@outgoing_voipvoice:1]
Dial(IAX2/localuser-3519, IAX2/my-iax-provider/number2call) in new
stack
[Apr 18 19:34:22] DEBUG[32577]: chan_iax2.c:4000 create_addr:
prepending 2 to prefs
-- Called my-iax-provider/number2call
-- Call accepted by 62.149.202.150 (format ilbc)
-- Format for call is ilbc
-- IAX2/my-iax-provider-25 is making progress passing it to
IAX2/localuser-3519
-- IAX2/my-iax-provider-25 is ringing
-- IAX2/my-iax-provider-25 is making progress passing it to
IAX2/localuser-3519
-- IAX2my-iax-provider-25 stopped sounds
-- IAX2/my-iax-provider-25 answered IAX2/localuser-3519
-- Operating with different codecs 2[0x2 (gsm)] 1024[0x400 (ilbc)]
, can't native bridge...
-- Hungup 'IAX2/my-iax-provider-25'
== Spawn extension (outgoing_voipvoice, number2call, 1) exited
non-zero on 'IAX2/localuser-3519'
-- Hungup 'IAX2/localuser-3519'
I'm having the same problem using the dial from console:
CLI console dial number2c...@outgoing_voipvoice
-- Executing [number2c...@outgoing_voipvoice:1]
Dial(Console/dsp, IAX2/ my-iax-provider/number2call) in new stack
[Apr 18 19:39:42] DEBUG[32645]: chan_iax2.c:4000 create_addr:
prepending 40 to prefs
-- Called my-iax-provider/number2call
-- Call accepted by 62.149.202.150 (format ilbc)
-- Format for call is ilbc
-- IAX2/my-iax-provider-361 is circuit-busy
-- Hungup 'IAX2/my-iax-provider-361'
== Everyone is busy/congested at this time (1:0/1/0)
Have you got any idea?
Thanks to all in advance
--
/*/
nik600
http://www.kumbe.it
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[asterisk-users] The best way to stop an ongoing call
Hello, I'm evaluating how I should implement the following feature (with asterisk 1.6.1) : - party A calls party B - A and B are talking to each other - supervisor C detects the call is on-going for too long (as, for example, others are waiting to use the phone) - C dials a special command that would play a short music to A and/or B and then cut the call 30 seconds later. At the moment, I would use : - System, cron and soft hangup to stop ongoing call - ChanSpy to barge into ongoing call plus a dynamic feature to playback a pre-recorded message when supervisor dials a DTMF sequence. I also wondered if if should instead move each party to a conference room with a timer. Suggestions ? Regards -- _ -- 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] AGI, FASTAGI or Windows Voice Server
Sorry if u understood this my english is so limited and not so good , my apologize it was not my intention. *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Sat, 17 Apr 2010 18:19:53 -0400 From: stot...@first-notification.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server On Sat, Apr 17, 2010 at 1:48 PM, Edwin Quijada listas_quij...@hotmail.com wrote: Just a shot in the dark, have you tried ExternalIVR? It was originally developed for Unwired Buyer (Ebay), I believe Digium and Unwired Buyer teamed up on this one. This option NO. Another option would be FastAGI to your windows server. You write an app for the windows box that interacts with the ATT application and then pipe the audio back to your asterisk box somehow. First thought is app_bridge or meetme. This is the idea just I dont know how to do. You can give any direction to start first. I am looking for information about app_bridge*---* *-Edwin Quijada *-Developer DataBase king *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com*-809-849-8087*---* This option NO. is quite a rude reply when someone is giving you ideas for free. Maybe you can say why it is not an option but your response was rude and makes me not want to help you anymore. I can tell you are an ESOL by the way you write, so maybe you don't understand the best way to communicate. Also, if you tried FTP, then did you not post that first. What else have you tried? Why waste people's time when you have tried things that didn't work but don't convey them? Did you try Samba? As far as app_bridge, there is plenty of documentation, let me waste more of my time.. http://tinyurl.com/y73mp9s Sounds like you should pay for the Linux version or paid Asterisk support. I really appreciate helping you, thanks, Steve Totaro _ -- _ -- 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] incoming ghost call
On Fri, Apr 16, 2010 at 11:32:00AM -0430, Danny Dias wrote: Hello asterisk users... We are having a little problem in our installation, we are using Asterisk 1.4.21.2 and zaptel 1.4.11 with a Digium TDM410P (3FXO + 1FXS), the problem is that when we disconnect the line from any of the fxo ports, we receive an incoming ghost call (using zap/x channel) it rings on the phone but we cant hear nothing...it's always doing the same everytime we disconnect the lines from the fxo We tried with a Sangoma card, and the problem went away, but we must use this digium card, we've tried with answer/hangup on polarityswitch with all the options, and we cant make this work ok, what should we do? Can you try latest Zaptel? I think some similar issues with that driver were resolved there. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Amazon EC2 SIP floods - you can help
On Sun, Apr 18, 2010 at 12:25 PM, Randy R randulo2...@gmail.com wrote: On Sun, Apr 18, 2010 at 7:17 PM, Fred Posner f...@teamforrest.com wrote: There's also a link to it from the VoIP Tech Chat article. And we are also linking to Fred's original story which says it all about Amazon: http://www.voiptechchat.com/voip/457/amazon-ec2-sip-brute-force-attacks-on-rise/ It has been suggested that posters on Twitter use hashtags. I'm not big on them myself, but #EC2, #SIP and #amazon might be appropriate. /r -- _ -- 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 Just a thought or my worst nightmare. i wonder if it isn't a hyperkit / hyperrootkit. A malicious variant of BluePill on a Virtual Machine that can spread through all other VM's on a machine because it becomes the hypervisor. Since a SAN is often used to move images from one machine to another, an infected vm fired up on a different machine could spread quickly. -- _ -- 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] Changing storm-prevention behaviour in logger.conf
On Apr 18, 2010, at 12:40 AM, Barry Miller wrote: On Sat, Apr 17, 2010 at 11:14:23PM +0200, Remco Bressers wrote: Dear List, According to https://issues.asterisk.org/view.php?id=14905 there is a storm prevention mechanism in newer Asterisks. If i look in my logfile, i see : [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Registration from ' sip:x...@xxx.xxx.xxx.xxx' failed for 'xx.xx.xx.xx' - Wrong password [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message repeated 3 times This IS a good thing to do, but i want to disable this behaviour. We are using fail2ban to ban scripts and people from the Asterisk system. On version 1.4.23 this worked fine, but now this mechanism is in place, i cannot use fail2ban anymore. Is there any option to disable this behaviour, or even better, add it to logger.conf so anybody can decide what to do? I just want all logging and it seems impossible now. Maybe a patch on the source? If you use a newer version of rsyslogd to do your logging, there is a global configuration directive: $RepeatedMsgReduction off that will do what you are asking. The issue #14905 patch you mention is not in 1.6.2.x. Hi, Well, this sounds fair, but this happened after an upgrade to 1.4.29 from 1.4.23. Nothing else changed in my setup after that. My logger.conf : [general] dateformat=%F %T [logfiles] console = notice,warning,error messages = notice,warning,error This tells me i'm not using the syslog feature at all and /var/log/asterisk/messages is generated by Asterisk and not by syslogd Please help. Regards, Remco -- _ -- 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] meetme / upgrade to 1.6.2.6
I read that I need to run 1.6.2.6 (at least 1.6) to use the MeetMe application since I don't have a zdummy timing driver. Anyway, I want to upgrade my machine to 1.6.2.6. Does anyone have the exact steps? I see a lot of references on the web but any other links from our community may be preferred! Thank you Tom -- _ -- 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] Amazon EC2 SIP floods - you can help
On Sun, Apr 18, 2010 at 8:16 PM, Rob Townley rob.town...@gmail.com wrote: Just a thought or my worst nightmare. i wonder if it isn't a hyperkit / hyperrootkit. A malicious variant of BluePill on a Virtual Machine that can spread through all other VM's on a machine because it becomes the hypervisor. Since a SAN is often used to move images from one machine to another, an infected vm fired up on a different machine could spread quickly. All the more reason for Amazon to get off their ass and look into this. r -- _ -- 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] Slightly OT: OMA DM Solution
Jay- anyone knows an open OMA DM tool that would be able to configure Nokia phones (mainly the sip-stuff of the e-series) for use with asterisk? Anything open i could find was the device manager from funambol, which was last updated in 2006 :-( I'm not sure what you're trying to do, but I can tell that we use pjsip-based applications to allow E-series cellphones to talk to Asterisk. Here is a data flow diagram example: ftp://ftp.signalogic.com/documentation/Boards/SigC64xx/Mobile_VoIP_Server_Acceleration_CallFlow_Diagram.pdf The SIP proxy isn't needed if you're not doing high capacity channels, load-balancing, encryption, transcoding, etc. Is there some specific OMA thing -- like say PoC -- that you're thinking is hard to find? Otherwise pjsip seems to handle basic mobile VoIP very well. -Jeff -- _ -- 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] Slightly OT: OMA DM Solution
Hiya Jeff, thanks for your answer, 2010/4/18 Jeff Brower jbro...@signalogic.com Jay- anyone knows an open OMA DM tool that would be able to configure Nokia phones (mainly the sip-stuff of the e-series) for use with asterisk? Anything open i could find was the device manager from funambol, which was last updated in 2006 :-( I'm not sure what you're trying to do, but I can tell that we use pjsip-based applications to allow E-series cellphones to talk to Asterisk. Here is a data flow diagram example: I'm trying to remotly*configure* E-Series device, OMA DM is OMA DeviceManagment, pretty much what the (european, at least) carrier do to configure GPRS/MMS settings on the phone... They already have a good (best on any handset, IMHO) SIP Application that very well integrated... And as i said, funambol offers an opensource DM Server, but it's outdated and a real bitch to install. All other solutions are commerical and VERY expensive, but what worries me more, all of them that i contacted where not willing to give even a handset/time-limited trial out... Thanks anyway, Kind Regards, Jay -- _ -- 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] kamailio
Hi guys, I want to integrate with two asterisk servers a kamailio sip server. Any of you know some good tutorial for this? Thanks in advance! Regards. -- jabber: trip...@12jabber.com blog: http://impresionesdeunloco.wordpress.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] AGI, FASTAGI or Windows Voice Server
I figured as much. ESOL=English as a Second Language. Apology accepted. Have you tried creating the file on the windows server, running sox to your specifications and then moving the file to a samba share? The key to this is moving the files at different stages. The first sound file is being created while the call is in progress. When the call is finished, move the file to a different location to process, after processing, move it to it's final destination so it can be played. Thanks, Steve T On Sun, Apr 18, 2010 at 2:00 PM, Edwin Quijada listas_quij...@hotmail.comwrote: Sorry if u understood this my english is so limited and not so good , my apologize it was not my intention. *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- Date: Sat, 17 Apr 2010 18:19:53 -0400 From: stot...@first-notification.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server On Sat, Apr 17, 2010 at 1:48 PM, Edwin Quijada listas_quij...@hotmail.com wrote: Just a shot in the dark, have you tried ExternalIVR? It was originally developed for Unwired Buyer (Ebay), I believe Digium and Unwired Buyer teamed up on this one. This option NO. Another option would be FastAGI to your windows server. You write an app for the windows box that interacts with the ATT application and then pipe the audio back to your asterisk box somehow. First thought is app_bridge or meetme. This is the idea just I dont know how to do. You can give any direction to start first. I am looking for information about app_bridge *---* *-Edwin Quijada *-Developer DataBase king *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* This option NO. is quite a rude reply when someone is giving you ideas for free. Maybe you can say why it is not an option but your response was rude and makes me not want to help you anymore. I can tell you are an ESOL by the way you write, so maybe you don't understand the best way to communicate. Also, if you tried FTP, then did you not post that first. What else have you tried? Why waste people's time when you have tried things that didn't work but don't convey them? Did you try Samba? As far as app_bridge, there is plenty of documentation, let me waste more of my time.. http://tinyurl.com/y73mp9s Sounds like you should pay for the Linux version or paid Asterisk support. I really appreciate helping you, thanks, Steve Totaro -- -- _ -- 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] Changing storm-prevention behaviour in logger.conf
On Sun, Apr 18, 2010 at 08:21:57PM +0200, Remco Bressers wrote: On Apr 18, 2010, at 12:40 AM, Barry Miller wrote: On Sat, Apr 17, 2010 at 11:14:23PM +0200, Remco Bressers wrote: Dear List, According to https://issues.asterisk.org/view.php?id=14905 there is a storm prevention mechanism in newer Asterisks. If i look in my logfile, i see : [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Registration from ' sip:x...@xxx.xxx.xxx.xxx' failed for 'xx.xx.xx.xx' - Wrong password [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message repeated 3 times This IS a good thing to do, but i want to disable this behaviour. We are using fail2ban to ban scripts and people from the Asterisk system. On version 1.4.23 this worked fine, but now this mechanism is in place, i cannot use fail2ban anymore. Is there any option to disable this behaviour, or even better, add it to logger.conf so anybody can decide what to do? I just want all logging and it seems impossible now. Maybe a patch on the source? If you use a newer version of rsyslogd to do your logging, there is a global configuration directive: $RepeatedMsgReduction off that will do what you are asking. The issue #14905 patch you mention is not in 1.6.2.x. Hi, Well, this sounds fair, but this happened after an upgrade to 1.4.29 from 1.4.23. Nothing else changed in my setup after that. My logger.conf : [general] dateformat=%F %T [logfiles] console = notice,warning,error messages = notice,warning,error This tells me i'm not using the syslog feature at all and /var/log/asterisk/messages is generated by Asterisk and not by syslogd Hi. First, I'm sorry I didn't look more closely at your in your example. Of course you're not using syslog, but rather asterisk's own logging. Second, I just downloaded 1.4.29. The patch that does the message repeated stuff is just not there, as Tilghman said. Is it possible that someone applied that patch to your source? Have you tried downloading the 1.4.29 tarball again and recompiling? If you installed asterisk as a package from somebody's repo, I can't really say, but it seems highly unlikely that the patch would be present. I hope this helps a little bit. -- Barry -- _ -- 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] Extensions Reload | Asterisk Freezes ? 1.4
Good day.. We have what I consider to be a large dialplan (-= 1501 extensions (2559 priorities) in 99 contexts. =-) If we have more than 10 or so channels up (all SIP, no TDM) and issue the extensions reload command.. quite often, asterisk will completely freeze up... requiring us to either kill and restart the process or restart the box... I should probably also share that when watching the log files, at the time the extensions reload command executes... there are no exceptions and/or issues reported while parsing the dialplan... Has anyone in the community experienced this.. and/or have any suggestions ? -- _ -- 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] meetme / upgrade to 1.6.2.6
I guess what you meant, is you don't have a physical card to provide the timing needed by Meetme. Then, if you are looking for dahdi to use kernel timer, then you need not to upgrade Aterisk but Dahdi to 2.3.0 Alyed 2010/4/18 Thomas Perron thomas.per...@gmail.com I read that I need to run 1.6.2.6 (at least 1.6) to use the MeetMe application since I don't have a zdummy timing driver. Anyway, I want to upgrade my machine to 1.6.2.6. Does anyone have the exact steps? I see a lot of references on the web but any other links from our community may be preferred! Thank you Tom -- _ -- 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] X-lite direct sip call - Is it possible?
You can't do that with Xlite, try Sjphone instead. Alyed 2010/4/17 bruce bruce bruceb...@gmail.com Hi Guys, Wondering if anyone has tried to make a direct SIP peer to peer call using x-lite without any registrations of any sort. I can't seem to find the setting. Thanks, bruce -- _ -- 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] meetme / upgrade to 1.6.2.6
You got him wrong. He actually want to know the steps to upgrade to version 1.6.2 so he do can a conference bridge using confbridge instead of of meetme because he does not have dahdi installed. He just want to know how to upgrade from an older version to version 1.6.2 On Sun, Apr 18, 2010 at 11:52 PM, Alyed al...@vivoxie.com wrote: I guess what you meant, is you don't have a physical card to provide the timing needed by Meetme. Then, if you are looking for dahdi to use kernel timer, then you need not to upgrade Aterisk but Dahdi to 2.3.0 Alyed 2010/4/18 Thomas Perron thomas.per...@gmail.com I read that I need to run 1.6.2.6 (at least 1.6) to use the MeetMe application since I don't have a zdummy timing driver. Anyway, I want to upgrade my machine to 1.6.2.6. Does anyone have the exact steps? I see a lot of references on the web but any other links from our community may be preferred! Thank you Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kamailio
Hi what do you want to integrate, Media Services or Loadbalance ? Ram On Mon, Apr 19, 2010 at 4:14 AM, Hector Muñoz hectormun...@gmail.comwrote: Hi guys, I want to integrate with two asterisk servers a kamailio sip server. Any of you know some good tutorial for this? Thanks in advance! Regards. -- jabber: trip...@12jabber.com blog: http://impresionesdeunloco.wordpress.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
