Re: [asterisk-users] Verizon-Vonage Lawsuit
Dovid B wrote: snip ROTFL. The US patent system is treated with contempt in Hong Kong? You have no idea how EXTREME legislation in Hong Kong against IP 'theft' is in Hong Kong. /snip I find this hard to believe since most hack attempts to my box's originate from IP's in China. What exactly would attacks from China have to do with the legal system in Hong Kong? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vonage fraud controls
And if they get you black-listed you can always signup with Verizon... On 4/8/07, Dean Collins [EMAIL PROTECTED] wrote: There's no way for them to tell if you have asterisk on the fxo port BUT they will terminate your account if you hook it up as the outbound for an office pumping call after call through it. What did you expect? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Salvatore Giudice Sent: Saturday, 7 April 2007 8:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Vonage fraud controls Has anyone tried pushing calls to a Vonage ATA attached to an FXO card in Asterisk and had your account terminated by Vonage? I'm curious as to whether they will stop your service if you push too many calls through their ATA in a specific period of time. Thanks in advance for the info, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702)979-2906 Fax: (212) 279-2906 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 30
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon-Vonage Lawsuit
Steve Underwood wrote: Dovid B wrote: snip ROTFL. The US patent system is treated with contempt in Hong Kong? You have no idea how EXTREME legislation in Hong Kong against IP 'theft' is in Hong Kong. /snip I find this hard to believe since most hack attempts to my box's originate from IP's in China. What exactly would attacks from China have to do with the legal system in Hong Kong? Steve Not sure, but maybe Verizon is responsible for the problem? Maybe I slept too much when I took logic as a colege freshman. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adding Noise or background noise
Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to trunk2 to make the call quality bad. Mainly I want to achieve bad call quality on trunk2 by adding some noise /echo /latency or something. Please guide me. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager Originate and Var to long
Hi, I use Originate to make a call. I have problems to bring my vars into the channel. Are there restrictions more then only 24 vars at mentioned at www.voip-info.org? Any workaround to get this running? WARNING[4641]: manager.c:1365 get_input: Dumping long line with no return from 127.0.0.1: regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a variable for SIP response codes?
Hi all,
I want to implement certain actions based on SIP response codes. Is there a
similar variable such as ${DIALSTATUS} that comes back with the relevant SIP
response code for a call?
--- Thanks
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Re: [asterisk-users] Is there a variable for SIP response codes?
Eric Bishop wrote:
Hi all,
I want to implement certain actions based on SIP response codes. Is there a
similar variable such as ${DIALSTATUS} that comes back with the relevant
SIP
response code for a call?
I believe there is SIPGetHeader, but Asterisk tries to translate
whatever code it gets from the specific technology (PRI, SIP, IAS2,
MGCP, SCCP, H323, etc) into an Asterisk HANGUPCAUSE which is mostly
Q.931 codes. HANGUPCAUSE will not tell you the SIP response code, but
it will tell you much more than DIALSTATUS will.
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Re: [asterisk-users] Is there a variable for SIP response codes?
Once the call is hung up it is too late. I need to interpret the SIP
response codes prior to hangup so I can play an appropriate recorded voice
announcement.
On 4/9/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Eric Bishop wrote:
Hi all,
I want to implement certain actions based on SIP response codes. Is
there a
similar variable such as ${DIALSTATUS} that comes back with the relevant
SIP
response code for a call?
I believe there is SIPGetHeader, but Asterisk tries to translate
whatever code it gets from the specific technology (PRI, SIP, IAS2,
MGCP, SCCP, H323, etc) into an Asterisk HANGUPCAUSE which is mostly
Q.931 codes. HANGUPCAUSE will not tell you the SIP response code, but
it will tell you much more than DIALSTATUS will.
___
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Re: [asterisk-users] Manager Originate and Var to long
you can easily increment the buffer size changing include/asterisk/manager.h #define AST_MAX_MANHEADER_LEN 256 chage that line for something like this #define AST_MAX_MANHEADER_LEN 512 and recompile Asterisk, Is the only way I know, Regards On 4/8/07, Thomas Winter [EMAIL PROTECTED] wrote: Hi, I use Originate to make a call. I have problems to bring my vars into the channel. Are there restrictions more then only 24 vars at mentioned at www.voip-info.org? Any workaround to get this running? WARNING[4641]: manager.c:1365 get_input: Dumping long line with no return from 127.0.0.1: regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a variable for SIP response codes?
I am assuming this:
Call comes in, the Dial happens and for whatever reason the destination
cannot be reached. You then want to play a message to the caller.
Just put the g option on the end of Dial and then check the
HANGUPCAUSE. The destination has already hungup, but the caller has not.
The extensions.conf.sample has something similar in the (I think)
[macro-stdexten]
Eric Bishop wrote:
Once the call is hung up it is too late. I need to interpret the SIP
response codes prior to hangup so I can play an appropriate recorded voice
announcement.
On 4/9/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Eric Bishop wrote:
Hi all,
I want to implement certain actions based on SIP response codes. Is
there a
similar variable such as ${DIALSTATUS} that comes back with the
relevant
SIP
response code for a call?
I believe there is SIPGetHeader, but Asterisk tries to translate
whatever code it gets from the specific technology (PRI, SIP, IAS2,
MGCP, SCCP, H323, etc) into an Asterisk HANGUPCAUSE which is mostly
Q.931 codes. HANGUPCAUSE will not tell you the SIP response code, but
it will tell you much more than DIALSTATUS will.
___
___
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RE: [asterisk-users] Adding Noise or background noise
From: Arun Kumar [EMAIL PROTECTED] Date: Sun, 8 Apr 2007 05:25:58 -0700 Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to trunk2 to make the call quality bad. Mainly I want to achieve bad call quality on trunk2 by adding some noise /echo /latency or something. Please guide me. This is got to be the strangest requirement I've seen - a penalty box. But if you must, one way to add noise could be to bring the parties to a conference, then add a third party to the conf. Another possibility is to use frequent announcements (don't have to be real announcements, but could be simple, brief noise) with L option in Dial(). I haven't seen L announcements working properly, though. Yuan Liu thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 31
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding Noise or background noise
Yuan LIU wrote: From: Arun Kumar [EMAIL PROTECTED] Date: Sun, 8 Apr 2007 05:25:58 -0700 Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to trunk2 to make the call quality bad. Mainly I want to achieve bad call quality on trunk2 by adding some noise /echo /latency or something. Please guide me. This is got to be the strangest requirement I've seen - a penalty box. Sounds like another half-baked calling card operation with tiered pricing. The rest of the world should subscribe to the Asterisk users list, then they'd see where their money is actually going. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan-capi-HEAD and Asterisk 1.4.2
On Tue, 3 Apr 2007, Armin Schindler wrote: On Tue, 3 Apr 2007, Peer Oliver Schmidt wrote: Hello Armin, thanks a lot for your help. Can you please do the same with 'showcapimsgs=2'? It may give more info on the commands itself, maybe some parameters are wrong here. Here you go. 17:23:17 is the magic time. This log below shows no error in parameters, but the problem is still the same: the fcpci driver doesn't respond and I cannot tell why. Can you please try HEAD version of SVN trunk (443)? It seems that the Fritz driver has a bug when registering at its CAPI interface. Armin Apr 3 17:23:09 server42 kernel: [263323.308388] fcpci: AVM FRITZ!Card PCI driver, revision 0.7.2 Apr 3 17:23:09 server42 kernel: [263323.308411] fcpci: (fcpci built on Feb 27 2007 at 21:22:25) Apr 3 17:23:09 server42 kernel: [263323.308421] fcpci: -- 32 bit CAPI driver -- Apr 3 17:23:10 server42 kernel: [263323.311559] PCI: Found IRQ 10 for device :00:0e.0 Apr 3 17:23:10 server42 kernel: [263323.311602] fcpci: AVM FRITZ!Card PCI found: port 0xdcc0, irq 10 Apr 3 17:23:10 server42 kernel: [263323.311613] fcpci: Loading... Apr 3 17:23:10 server42 kernel: [263323.311625] fcpci: Driver 'fcpci' attached to fcpci-stack. (152) Apr 3 17:23:10 server42 kernel: [263323.539987] fcpci: Stack version 3.11-07 Apr 3 17:23:10 server42 kernel: [263323.541140] kcapi: Controller 1: fcpci-dcc0-10 attached Apr 3 17:23:10 server42 kernel: [263323.541154] kcapi: card 1 fcpci-dcc0-10 ready. Apr 3 17:23:10 server42 kernel: [263323.541833] fcpci: Loaded. Apr 3 17:23:12 server42 kernel: [263325.975634] capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) Apr 3 17:23:17 server42 kernel: [263330.892916] kcapi: put [0x1] FACILITY_REQ ID=001 #0x0001 LEN=0018 Apr 3 17:23:17 server42 kernel: [263330.892926] Controller/PLCI/NCCI = 0x1 Apr 3 17:23:17 server42 kernel: [263330.892933] FacilitySelector = 0x3 Apr 3 17:23:17 server42 kernel: [263330.892939] FacilityRequestParameter = 00 00 00 Apr 3 17:23:17 server42 kernel: [263330.892946] Apr 3 17:23:17 server42 kernel: [263330.893153] kcapi: got [0x1] FACILITY_CONF ID=001 #0x0001 LEN=0026 Apr 3 17:23:17 server42 kernel: [263330.893163] Controller/PLCI/NCCI = 0x1 Apr 3 17:23:17 server42 kernel: [263330.893169] Info= 0x0 Apr 3 17:23:17 server42 kernel: [263330.893176] FacilitySelector = 0x3 Apr 3 17:23:17 server42 kernel: [263330.893182] FacilityConfirmationParameter = 00 00 06 00 00\37703 00 00 Apr 3 17:23:17 server42 kernel: [263330.893190] Apr 3 17:23:17 server42 kernel: [263330.900689] kcapi: put [0x1] LISTEN_REQ ID=001 #0x0002 LEN=0026 Apr 3 17:23:17 server42 kernel: [263330.900699] Controller/PLCI/NCCI = 0x1 Apr 3 17:23:17 server42 kernel: [263330.900706] InfoMask= 0x Apr 3 17:23:17 server42 kernel: [263330.900713] CIPmask= 0x1fff03ff Apr 3 17:23:17 server42 kernel: [263330.900720] CIPmask2= 0x0 Apr 3 17:23:17 server42 kernel: [263330.900726] CallingPartyNumber = default Apr 3 17:23:17 server42 kernel: [263330.900733] CallingPartySubaddress = default Apr 3 17:23:17 server42 kernel: [263330.900739] -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan-capi-HEAD and Asterisk 1.4.2
Hello Armin (and happy easter), thanks for you continuing support. Can you please try HEAD version of SVN trunk (443)? Did checkout the 443. It works without any verbosity. THANK YOU! I'll buy you a beer, if you ever happen to come to the northern part of Germany. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio Gain Settings
On Sat, 2007-04-07 at 23:52 -0500, Eric ManxPower Wieling wrote: snip The device doing the IP/TDM conversion should be the device that sets the gains correctly. The same applies to echo canceling. As I stated, this started with the warning of Novice Question :-). Eric, can you elaborate on the above? Is the device you are referring to within Asterisk or somewhere else in VOIP land? I am not sure what to do with this information. If it matters - the clipping behavior I see is in voices recorded on Asterisk 1.4.2 from a call placed over Packet8 and routed back to my Asterisk box through NuFone.net. Same happens from a POTS call routed back to my Asterisk box through NuFone.net. Thanks, -- Bob Smither [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon-Vonage Lawsuit
Dovid B wrote: snip ROTFL. The US patent system is treated with contempt in Hong Kong? You have no idea how EXTREME legislation in Hong Kong against IP 'theft' is in Hong Kong. /snip I find this hard to believe since most hack attempts to my box's originate from IP's in China. Welcome to China. Most Hong Kong'ers loathe mainland chinese and if they could, they would never get a job in China. I get plenty of hack attempts too from China however I doubt that is due to the same sentiment in China. If you want to find someone to blame, please look no further than the US where your chicken boners are in league with crackers and virus writers to create botnets to send their spam. This is of course besides the ignorance of those who own computers in China (man, computers there are infested with virii, worms and trojans) that run that most secure of operating systems Microsoft Windows and those who actually get paid by chicken boners to host their crap. Oh, there are plenty of hack attempts from Korea too. Are you going to add Korea to the list of 'IP' violators too? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] intermittent choppy sound over wifi link
I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider - IAX2 over the Internet - 20Mb fiber connection - router - Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running 3.1.19(SE) firmware), also tested with X-lite softest - PIX 506 (although I have tried multiple routers and direct connection to the radio try to fix the problem) - 1 mile 802.11b link to AP - 15 mile 802.11b link Backhaul - router - Asterisk My Asterisk version is Asterisk 1.2.12.1, Zaptel 1.2.9.1. Ping times are ~10ms, jitter is under 10 with an average of 5. QoS is enabled in the router for SIP, RTP and IAX2 traffic going to and from the Asterisk box. When I experience the choppiness the ATA reports packet loss on the web interface (Call 1 Packets Lost: ). I can run something such as ping plotter from the same leg of the network that the Asterisk box is on while this is happening and there is not even a small glitch of lost packets on the network but the ATA displays otherwise. The only thing I have come up with thus far is possible retransmissions on the wireless connection (and due to the type of gear, I'm not able to see this data). We are way out in the country with no other real providers even close so I'm doubting interference although I suppose it is a possibility keeping an open mind. My question is can anyone point me to any possible reasons this would be happening? Also can anyone tell me other reasons other than real lost packets that the ATA would show this? My only guess on that was packets that never got an ACK due to server congestion or some other reason other than actual loss. Any insight appreciated! Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] intermittent choppy sound over wifi link
Curt Shaffer wrote: I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider - IAX2 over the Internet - 20Mb fiber connection - router - Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running 3.1.19(SE) firmware), also tested with X-lite softest - PIX 506 (although I have tried multiple routers and direct connection to the radio try to fix the problem) - 1 mile 802.11b link to AP - 15 mile 802.11b link Backhaul - router - Asterisk My Asterisk version is Asterisk 1.2.12.1, Zaptel 1.2.9.1. Ping times are ~10ms, jitter is under 10 with an average of 5. QoS is enabled in the router for SIP, RTP and IAX2 traffic going to and from the Asterisk box. When I experience the choppiness the ATA reports packet loss on the web interface (Call 1 Packets Lost: ). I can run something such as ping plotter from the same leg of the network that the Asterisk box is on while this is happening and there is not even a small glitch of lost packets on the network but the ATA displays otherwise. The only thing I have come up with thus far is possible retransmissions on the wireless connection (and due to the type of gear, I’m not able to see this data). We are way out in the country with no other real providers even close so I’m doubting interference although I suppose it is a possibility keeping an open mind. My question is can anyone point me to any possible reasons this would be happening? Also can anyone tell me other reasons other than real lost packets that the ATA would show this? My only guess on that was packets that never got an ACK due to server congestion or some other reason other than actual loss. The most likely culprit is jitter. Any insight appreciated! Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
