Re: [Asterisk-Users] Unlocking Vonage ATA 186
Right, which the security hole mentioned for 2.14 doesn't even apply to this method. This isn't a 'security hole' per se, it's just how the data is stored inside the device. Vonage is running the latest 2.16-2 firmware. No longer applicable. cameron. On Wed, 24 Dec 2003, Doug Shubert wrote: this security hole has been around for some time http://www.securiteam.com/securitynews/5PP0G0K75U.html Lion Templin wrote: In the process of investigating a Cisco ATA 186 that was locked by Vonage, I found that you can still unlock the device yourself. But there's a catch. The device's design has a great plus: a DIP32 *socketed* SST28SF040A flash chip. I found an 8 digit unlock code at 0x03FA71-0x03FA78. I do not know if that is a standard location. If you have the equipment, you're in luck. But IMHO, the $15 fee is more than reasonable .. and certainly less than what it would cost to get a device to read/write these flash chips. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 1003 http://www.pulver.com/fwd/ ext. 83740 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- = lion is Lion J Templin [EMAIL PROTECTED] = = 612-605-3613 x3001 FWD 94117 = ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unlocking Vonage ATA 186
If you are willing to snarf the data out of the prom I agree it will work. I may have missed the beginning of the thread,,, but is someone in the business of recovering Vonage ATAs? I may be in the market soon... It would be interesting to see how much of the contents of the prom are readable. cameron. On Thu, 25 Dec 2003, Lion Templin wrote: Right, which the security hole mentioned for 2.14 doesn't even apply to this method. This isn't a 'security hole' per se, it's just how the data is stored inside the device. Vonage is running the latest 2.16-2 firmware. No longer applicable. cameron. On Wed, 24 Dec 2003, Doug Shubert wrote: this security hole has been around for some time http://www.securiteam.com/securitynews/5PP0G0K75U.html Lion Templin wrote: In the process of investigating a Cisco ATA 186 that was locked by Vonage, I found that you can still unlock the device yourself. But there's a catch. The device's design has a great plus: a DIP32 *socketed* SST28SF040A flash chip. I found an 8 digit unlock code at 0x03FA71-0x03FA78. I do not know if that is a standard location. If you have the equipment, you're in luck. But IMHO, the $15 fee is more than reasonable .. and certainly less than what it would cost to get a device to read/write these flash chips. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 1003 http://www.pulver.com/fwd/ ext. 83740 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 BTXML
Has anyone devloped some perl scripts for sending Directory and Services information to the Cisco 7960? The older thread I googled is giving me fits. Thank you, cameron. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to H.323 without gatekeeper
Andrew Thompson wrote:
- Original Message -
From: Pavel Litvinenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 19, 2003 8:42 AM
Subject: Re: [Asterisk-Users] Asterisk to H.323 without gatekeeper
bam wrote:
I've read through the archives and have picked up that * does not need
a gatekeeper to talk directly with an H323 handset to send and receive
calls.
I'm trying to go PSTN*-H323 and all the examples that I can
find use a gatekeeper. Are there any examples or hints for doing it
without the gatekeeper?
many thanks in advance
Brian
[your_context]
exten = _9XX,1,Dial,H323/78632${EXTEN:[EMAIL PROTECTED]|30
exten = _9XX,2,Busy
exten = _9XX,102,Busy
What's the 78632? Is that something you have to dial, like country/area code
+ 6 digits?
I have to pass called number in e164 format ... 7 - russia, 8632 -
Rostov-on-Don.
This is just an example ... of course there should be your own extentions :)
-
Andrew Thompson http://aktzero.com/
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The opinions
stated above are yours. You cannot imagine why you ever felt otherwise.
___
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--
-
Best Regards,
Pavel Litvinenko.
ICQ: 16224754
Ph: (8632) 923962, 923640
sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Encryption
My first words, merry xmas to everyone... And may your voip packets be safer in 2004 :P All details supplied below might be subject to my complete retardness and incompetence because its early and I didn't have any coffee yet :) Steven Critchfield wrote: On Wed, 2003-12-24 at 19:11, Mahoney, Matt wrote: Hi, Does asterisk support any kind of voice encryption? Not right now. As I understand it, it is a problem with the fact that each packet would have to be able to be decrypted even if packets in the stream are lost. In the long run it is going to be a question of how secure do you need it to be. Afaik, the good encryption methods are stream based, and the loss of packets would kill the decryption. Most block and stream ciphers can recover from loss... And if you don't use block chaining, you have no loss at all... (but it's less secure, however this is always relative to what you try to protect) (With block chaining, you lose the next block) (Afaik block ciphers are more secure, but well... i'm no cryptographer, nor cryptanalyst... or mathematician... or ...) Things like plaintext attacks... are still possible... think about the same voice data going through an encrypted channel (even trunk), and then ending up on the sip phone of our attacker... or some unencrypted voice channel (sip, mgcp, etc)... so he then knows a part of the plain text of the encrypted trunk... possibly revealing some of the keys' info... possibly revealing all the calls over that trunk during the use of that exact key. It is questionable how feasible such an attack would be, but this is not the question we should ask ourselves... So, how would you encrypt a trunk? per channel? that would seem horrible considering you would need a key exchanged for each channel etc., and the overhead would go up quite a bit... Would you want to use stream ciphers? (we have blocks of voice data? not much streaming in * to my knowledge) Would you want to use a different key for voice and commands? Considering implementation... you will get even more overhead if you decide to stay backwards compatible with iax2, unless you do some really funky (ugly) things, which are prone to errors in the implementation. Add to that the problem of finding some encryption method that can deal with small packets without adding a lot of overhead. Each voice packet in IAX with GSM compression isn't very long. The voice data is only 33 bytes. The overhead? Well the block ciphers come in 2 tastes... 64 and 128bit block sizes (mind you... block size != key size) 64: *DES, Blowfish, ... 128: AES, Twofish, ... So with a blockcipher the overhead would be: 33 times 8 = 264 bits 64: 5 blocks = 320 bits = 56 bits overhead per packet 128: 3 blocks = 348 bits = 120 bits overhead per packet (Yes, this is exactly worst case since you are 1 byte over the exact fit limit...) (264 - 8 = 256) These are just some of my thoughts, please don't pin yourself by just looking at the best cipher... which is considered to be AES-CTR (block) by many people... as you see it has 120bits of overhead when used with GSM... (If the 33 bytes figure presented before is right) Cipher capabilities could be exchanged just like codec capabilities... and if a device (IAXy?) only happens to support plain AES... so be it, but please don't restrict the protocol to that :) Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Encryption
Steven Critchfield wrote: On Wed, 2003-12-24 at 19:11, Mahoney, Matt wrote: Hi, Does asterisk support any kind of voice encryption? Not right now. As I understand it, it is a problem with the fact that each packet would have to be able to be decrypted even if packets in the stream are lost. In the long run it is going to be a question of how secure do you need it to be. Afaik, the good encryption methods are stream based, and the loss of packets would kill the decryption. Add to that the problem of finding some encryption method that can deal with small packets without adding a lot of overhead. Each voice packet in IAX with GSM compression isn't very long. The voice data is only 33 bytes. The need to tolerate packet losss means a continuous stream cypher can't be used. That reduces the quality of the ciphering, but it can still be pretty good. You know the packet sequence number (at least with RTP you do), so some forms of sequentially changing encryption can still be used. SRTP has been through the process of trying to deal with this in the most effective manner, but doesn't seem to be widely used right now. Free implementations exists - see srtp.sourceforge.net. I guess it should be adaptable to IAX. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] return of the transfer to a busy number
Hello, Can such thing be done through dialplan , that say I transfer a call to an extension but it is busy, so that this call returns back to me. Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call pickup via *8 from ata186 (SIP)
Hello, Does call pickup works with subj? at the same pbx it works with MGCP but bit ata-186 with SIP it doesnt work, just nothing happens. Anyone have it working? Also it seems that when typing reload on the console, the asterisk doesnt reread the mgcp.conf. Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Encryption
On Thu, 25 Dec 2003, Michael Sandee wrote: Most block and stream ciphers can recover from loss... And if you don't use block chaining, you have no loss at all... (but it's less secure, however this is always relative to what you try to protect) (With block chaining, you lose the next block) (Afaik block ciphers are more secure, but well... i'm no cryptographer, nor cryptanalyst... or mathematician... or ...) It opens up a *LOT* of attacks, most significantly data injection, if you aren't really careful. Here's how I would write the protocol... You have a bunch of input packets which consist of some sort of UDP payload that represents the voice on the channel. These are D (for Data) Each connection has a unique ID. This ID should be based on time and be unique. An example would be Time of connection origination in seconds since 1970 concatinated with a random number. So this number would probably be a 64 bit number - the first four bytes would be the seconds since 1970, the next four bytes would be random. This is CID. Each packet would then have a sequence number - not for the traditional reassmbly reasons, though, simply to keep old packets from being reinjected. This probably isn't necessarily though if * uses sequence numbers in IAX currently. I'm assuming that it is already in the data stream now. To encrypt, you would take a block cipher. The result packet would be (. is string encapsulation, E() is encrypt): P = CID . E(CID . D) To decrypt you would split the packet into CID and C (Ciphertext). And then you would (E'() is decrypt): CID . D = E'(C) You would compare the plain text CID with the value in the encrypted packet, logging an attack if they don't match. You would also need to throw out old CIDs when they tried to initiate a connection. And also old packet sequence numbers. To prevent replay attacks. This would let the packets arrive out of sequence, handle a missing packet fine, and also ensure that two packets containing exactly the same data did not have the same ciphertext (as they would have different sequence numbers and CIDs). These are just some of my thoughts, please don't pin yourself by just looking at the best cipher... which is considered to be AES-CTR (block) by many people... as you see it has 120bits of overhead when used with GSM... (If the 33 bytes figure presented before is right) Cipher capabilities could be exchanged just like codec capabilities... and if a device (IAXy?) only happens to support plain AES... so be it, but please don't restrict the protocol to that :) It would be nice, though, if the cipher something like AES, though, since that would meet government requirements for encryption in the US. It might give us more users. 3DES would work - right now - but not in a few years as 3DES is in the process of being phased out. I do understand the overhead issues, though. I would say that I would have uses for this technology tomorrow if it was cheap and affordable, had a well-engineered protocol (which mine probably isn't - I just threw it out to show one way of trying to solve this problem), etc. I work with some organizations that really do need encrypted voice but can't afford commercial encrypted telephones. It would also be a good way for me to get VoIP into those organizations. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Encryption
I understand AES can do this. bkw On Thu, 25 Dec 2003, Steve Underwood wrote: Steven Critchfield wrote: On Wed, 2003-12-24 at 19:11, Mahoney, Matt wrote: Hi, Does asterisk support any kind of voice encryption? Not right now. As I understand it, it is a problem with the fact that each packet would have to be able to be decrypted even if packets in the stream are lost. In the long run it is going to be a question of how secure do you need it to be. Afaik, the good encryption methods are stream based, and the loss of packets would kill the decryption. Add to that the problem of finding some encryption method that can deal with small packets without adding a lot of overhead. Each voice packet in IAX with GSM compression isn't very long. The voice data is only 33 bytes. The need to tolerate packet losss means a continuous stream cypher can't be used. That reduces the quality of the ciphering, but it can still be pretty good. You know the packet sequence number (at least with RTP you do), so some forms of sequentially changing encryption can still be used. SRTP has been through the process of trying to deal with this in the most effective manner, but doesn't seem to be widely used right now. Free implementations exists - see srtp.sourceforge.net. I guess it should be adaptable to IAX. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX NOTICE and WARNING messages
Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File chan_iax.c, Line 2878 (register_verify) empty registration from 192.168.0.30 NOTICE[1116941120]: File chan_iax.c, Line 2878 (register_verify) empty registration from 192.168.0.30 WARNING[1142106560]: File chan_iax2.c, Line 470 (iax_error_output): Information element length exceeds message size WARNING[1142106560]: File chan_iax2.c, Line 4191 (socket_read): undecodable frame received I receive the same exact error messages on the second server, but with the .5 IP Address in the NOTICE Messages. I have looked through the .c files to see if I can figure out what is happening, but can't come up with anything. Are these types of messages normal? Additionally, In the iax.conf files, I have set both servers to use A and U law codecs. I have attached parts of my config below. Any help would be greatly appreciated! Best Regards, Brent Franks - The servers have the same exact config, however the registrations are obviously different. Server 1 Has this: register = hunasterisk:[EMAIL PROTECTED]:5036 [holasterisk] type=friend auth=md5 username=holasterisk secret= context=local host=dynamic defaultip=192.168.0.30 qualify=yes ;trunk=yes While Server 2 has: register = holasterisk:[EMAIL PROTECTED]:5036 [hunasterisk] type=friend auth=md5 username=hunasterisk secret= context=local host=dynamic defaultip=192.168.0.5 qualify=yes ;trunk=yes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX NOTICE and WARNING messages
Use IAX2 :) bkw On Thu, 25 Dec 2003, Brent Franks wrote: Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File chan_iax.c, Line 2878 (register_verify) empty registration from 192.168.0.30 NOTICE[1116941120]: File chan_iax.c, Line 2878 (register_verify) empty registration from 192.168.0.30 WARNING[1142106560]: File chan_iax2.c, Line 470 (iax_error_output): Information element length exceeds message size WARNING[1142106560]: File chan_iax2.c, Line 4191 (socket_read): undecodable frame received I receive the same exact error messages on the second server, but with the .5 IP Address in the NOTICE Messages. I have looked through the .c files to see if I can figure out what is happening, but can't come up with anything. Are these types of messages normal? Additionally, In the iax.conf files, I have set both servers to use A and U law codecs. I have attached parts of my config below. Any help would be greatly appreciated! Best Regards, Brent Franks - The servers have the same exact config, however the registrations are obviously different. Server 1 Has this: register = hunasterisk:[EMAIL PROTECTED]:5036 [holasterisk] type=friend auth=md5 username=holasterisk secret= context=local host=dynamic defaultip=192.168.0.30 qualify=yes ;trunk=yes While Server 2 has: register = holasterisk:[EMAIL PROTECTED]:5036 [hunasterisk] type=friend auth=md5 username=hunasterisk secret= context=local host=dynamic defaultip=192.168.0.5 qualify=yes ;trunk=yes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sip phones on the same extension?
Hello. I think I understand your suggestion, but don't understand how that's any different than the one I came up with. What I want, is to be able to define a specific extension, and then have any external SIP phones register with that extension that want to. It's important that multiple phones be able to register with the same extension simultaneously. Then, I can define something like: exten = 300,1,Dial(SIP/300,15|t) and all phones registered to extension SIP/300 will ring. The number of phones existing on that extension at any given time is unknown, and Asterisk should be able to keep a list of all devices which are currently registered on a given extension, even if it has seen another device register to the same extension. To guard against number stealing, one could restrict the registration of a given phone number to a single password, but allow that password to be used as often and from where ever. So, for example, if my extension is 300, and my password is JustForFun, I should be able to program any number of SIP phones to register as extension 300, and as long as they know the magic password, JustForFun, Asterisk will permit all of them to register as SIP/300. Then, if someone calls 300, they'll all ring simultaneously, and which ever phone gets picked up first, gets the call. This doesn't appear to be how Asterisk works at the moment. Am I wrong about this? -Brian Message: 9 From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip phones on the same extension? Date: Wed, 24 Dec 2003 13:24:53 -0600 Reply-To: [EMAIL PROTECTED] In sip.conf: [phone1] type=peer host=dynamic [phone2] type=peer host=dynamic [phone3] type=peer host=dynamic in extensions.conf: [default] exten = 0,1,Dial(SIP/phone1SIP/phone2SIP/phone3,30,T) -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sip phones on the same extension?
Because the one you came up with isn't possible with asterisk at this time. On Thu, 25 Dec 2003, Brian Buhrow wrote: Hello. I think I understand your suggestion, but don't understand how that's any different than the one I came up with. What I want, is to be able to define a specific extension, and then have any external SIP phones register with that extension that want to. It's important that multiple phones be able to register with the same extension simultaneously. Then, I can define something like: exten = 300,1,Dial(SIP/300,15|t) and all phones registered to extension SIP/300 will ring. The number of phones existing on that extension at any given time is unknown, and Asterisk should be able to keep a list of all devices which are currently registered on a given extension, even if it has seen another device register to the same extension. To guard against number stealing, one could restrict the registration of a given phone number to a single password, but allow that password to be used as often and from where ever. So, for example, if my extension is 300, and my password is JustForFun, I should be able to program any number of SIP phones to register as extension 300, and as long as they know the magic password, JustForFun, Asterisk will permit all of them to register as SIP/300. Then, if someone calls 300, they'll all ring simultaneously, and which ever phone gets picked up first, gets the call. This doesn't appear to be how Asterisk works at the moment. Am I wrong about this? -Brian Message: 9 From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip phones on the same extension? Date: Wed, 24 Dec 2003 13:24:53 -0600 Reply-To: [EMAIL PROTECTED] In sip.conf: [phone1] type=peer host=dynamic [phone2] type=peer host=dynamic [phone3] type=peer host=dynamic in extensions.conf: [default] exten = 0,1,Dial(SIP/phone1SIP/phone2SIP/phone3,30,T) -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX NOTICE and WARNING messages
Hi Brian, Thanks for the help. I've changed the ports on the registration statements, however I still receive the last two error messages. The first two are gone. WARNING[1142106560]: File chan_iax2.c, Line 470 (iax_error_output): information element length exceeds message size WARNING[1142106560]: File chan_iax2.c, Line 4191 (socket_read): undecodable frame received Both error messages are showing up on both servers. Thanks again, Brent -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Thursday, December 25, 2003 2:02 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX NOTICE and WARNING messages Use IAX2 :) bkw On Thu, 25 Dec 2003, Brent Franks wrote: Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File chan_iax.c, Line 2878 (register_verify) empty registration from 192.168.0.30 NOTICE[1116941120]: File chan_iax.c, Line 2878 (register_verify) empty registration from 192.168.0.30 WARNING[1142106560]: File chan_iax2.c, Line 470 (iax_error_output): Information element length exceeds message size WARNING[1142106560]: File chan_iax2.c, Line 4191 (socket_read): undecodable frame received I receive the same exact error messages on the second server, but with the .5 IP Address in the NOTICE Messages. I have looked through the .c files to see if I can figure out what is happening, but can't come up with anything. Are these types of messages normal? Additionally, In the iax.conf files, I have set both servers to use A and U law codecs. I have attached parts of my config below. Any help would be greatly appreciated! Best Regards, Brent Franks - The servers have the same exact config, however the registrations are obviously different. Server 1 Has this: register = hunasterisk:[EMAIL PROTECTED]:5036 [holasterisk] type=friend auth=md5 username=holasterisk secret= context=local host=dynamic defaultip=192.168.0.30 qualify=yes ;trunk=yes While Server 2 has: register = holasterisk:[EMAIL PROTECTED]:5036 [hunasterisk] type=friend auth=md5 username=hunasterisk secret= context=local host=dynamic defaultip=192.168.0.5 qualify=yes ;trunk=yes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] time to build an open phone?
How about to build an ip phone with this IC? http://focus.ti.com/docs/apps/catalog/general/applications.jhtml?templateId= 969path=templatedata/cm/general/data/bband_ipphone_tnetv1001 - Original Message - From: Bob Knight [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 4:30 PM Subject: [Asterisk-Users] time to build an open phone? Open software seems to work. Why don't we try it with hardware. 1. pick an embedded processor. It should have a nice linux gui support (like x jtag debugger). 2. pick a linux based cad system we all have easy access to and place schematics under cvs. 3. pick some type of gpio or serial interface for keyboard/display. 4. pick some basic functionality. 5. code it up. A stripped down *. Let everyone do their own thing with the expensive part. Tooling/packaging. We could let Digium be the distributor, so they are not left out of the loop. A board set would be offered with NO support. If Digium wants no part of it, we just build them on our own for our own use or sell them on ebay. What we would provide is schematics and source code. Everyone can take this to their favorite fab house and crank em out. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Sip phones on the same extension?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Buhrow Sent: Thursday, December 25, 2003 2:15 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Sip phones on the same extension? Hello. I think I understand your suggestion, but don't understand how that's any different than the one I came up with. What I want, is to be able to define a specific extension, and then have any external SIP phones [...] The difference is, his suggestion works. Yours doesn't. If you register multiple SIP devices in the way you suggest, only one of them will ring. It appears to me that the one that is fastest to respond will work, but I only tried the setup briefly before doing a bit of research that told me it wasn't the way this is done in *. Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 troubles
Anton V Kirichenko wrote: No, I did't bought any license from Digium. But as I say at my previous post, only _some part_ of my g729 calls are failed ! I think if I need license for G729 at Asterisk then all of my calls must to fails. Is it right ? It depends on what you are trying to do with G729. If you are just calling between two G729 endpoints then you don't need a license. If you are trying to call Asterisk's voicemail, any application that generates sound (like MeetMe) or trying to call the PSTN, then you need a G729 license. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] return of the transfer to a busy number
Anton Yurchenko wrote: Hello, Can such thing be done through dialplan , that say I transfer a call to an extension but it is busy, so that this call returns back to me. exten = 123,1,Dial(Zap/5,30) ; ring Zap/5 for 30 seconds exten = 123,2,VoiceMailMain(u123) ; No answer voicemail exten = 123,102,Wait(2) exten = 123,103,Dial(Local/$(CALLERIDNUM}) You can also do FLASH, dial number, if it's busy FLASH twice, if it's not hang up. This is for Zap channels. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X101P stopped working. Newbie seeks help
Hi I received the Asterisk Dev Kit Lite on Saturday and got a small PBX working fairly quickly using the config files supplied with the kit. This was on my desktop machine and the sound card stopped working. I turned the system off and moved the sound card to another slot to give it a different IRQ. On reboot, the modules zaptel, wcfxo, and wcusb all load OK and asterisk starts OK but there is no dialtone on the extension and the X101P never answers. The command show channels shows nothing. I moved cards around, rebuilt, reloaded, rebooted... nothing worked. I bought a new system and did a fresh RH9 install. Did a cvs checkout of zaptel, asterisk, zapata, and libpri. The problem persists on the new machine. The new box is on a separate spur on the firewall so I'll be glad to give root access to anyone willing to help. Any ideas on what to do next? Thanks Bob Smith bsmith at linuxtoys dot org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can't get oss console working.
I've been trying to get a console channel working without success. The sound card, which is built into the motherboard, is a VIA Technologies, Inc. VT82C686 AC97 Audio Controller. Using the oss drivers (vi82cxxx_audio) in kernel 2.4.23 and chan_oss, I just get beeps and screeches. Using alsa drivers (snd-via82cxx) and chan_oss (using the alsa oss emulation), playing sound works, but anything recorded via the mic has a weird scratchy echo over it. If I use a program other than asterisk to record (such as rec or arecord), the recording sounds fine. It works even if I'm playing a sound at the same time, so its not a duplex issue with the card. So I attempted to use chan_alsa with the alsa drivers. I discovered it will segfault with the current version of alsa unless you add #define ALSA_PCM_OLD_HW_PARAMS_API #define ALSA_PCM_OLD_SW_PARAMS_API to the top of chan_alsa.c. Even then, though, it doesn't work right. Playing sound works, but recording doesn't work at all. You just get nothing. I also tried an Ensoniq ES1371 AudioPCI sound card. With alsa drivers, it behaves just like the VT82C686. With oss drivers, it mostly works, except the sound level of things recorded with the mic is way way too low, no matter how I adjust the mixer. So, any suggestions? Does anyone have a working console with asterisk, and if so, what combination of channel, sound card, and sound card drivers are you using? Thanks. Here is more information about my system, in case its useful. Redhat 9 Kernel 2.4.23 1.2Ghz celeron 640mb ram X100P and TDM400P interface cards asterisk from cvs on 12/24/03 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Encryption
Steve == Steve Underwood [EMAIL PROTECTED] writes: Steve SRTP has been through the process of trying to deal with this Steve in the most effective manner, but doesn't seem to be widely Steve used right now. Free implementations exists - see Steve srtp.sourceforge.net. I guess it should be adaptable to IAX. It should be adaptable, and is a good starting point. In the short term, anyone who needs encryption between n * boxen, and controls all of them, should give ipsec a test. They've also had to deal with encrypting a non-reliable packet-stream, and have been working on it for several years now (The major dists should all support it in their kernels; if you compile your own it is not difficult to add to 2.4 kernels -- 2.6.0 already has it.) -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] time to build an open phone?
only problem is the protocol stack isn't open. Good chip CW_ASN wrote: How about to build an ip phone with this IC? http://focus.ti.com/docs/apps/catalog/general/applications.jhtml?templateId= 969path=templatedata/cm/general/data/bband_ipphone_tnetv1001 - Original Message - From: Bob Knight [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 4:30 PM Subject: [Asterisk-Users] time to build an open phone? Open software seems to work. Why don't we try it with hardware. 1. pick an embedded processor. It should have a nice linux gui support (like x jtag debugger). 2. pick a linux based cad system we all have easy access to and place schematics under cvs. 3. pick some type of gpio or serial interface for keyboard/display. 4. pick some basic functionality. 5. code it up. A stripped down *. Let everyone do their own thing with the expensive part. Tooling/packaging. We could let Digium be the distributor, so they are not left out of the loop. A board set would be offered with NO support. If Digium wants no part of it, we just build them on our own for our own use or sell them on ebay. What we would provide is schematics and source code. Everyone can take this to their favorite fab house and crank em out. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X101P stopped working. Newbie seeks help
This is probably not the problem, but if you have a slow machine (500MHz), you might try putting a couple seconds of sleep between each stage of the startup (modprobe, ztcfg, asterisk etc) and see if it helps. One of my machines is slow, and I had the same problem with the demo kit. Or, instead of starting everything automatically at reboot, do each step manually (with -vv on ztcfg) and see if you get any errors. Good luck Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Smith Sent: Thursday, December 25, 2003 8:26 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X101P stopped working. Newbie seeks help Hi I received the Asterisk Dev Kit Lite on Saturday and got a small PBX working fairly quickly using the config files supplied with the kit. This was on my desktop machine and the sound card stopped working. I turned the system off and moved the sound card to another slot to give it a different IRQ. On reboot, the modules zaptel, wcfxo, and wcusb all load OK and asterisk starts OK but there is no dialtone on the extension and the X101P never answers. The command show channels shows nothing. I moved cards around, rebuilt, reloaded, rebooted... nothing worked. I bought a new system and did a fresh RH9 install. Did a cvs checkout of zaptel, asterisk, zapata, and libpri. The problem persists on the new machine. The new box is on a separate spur on the firewall so I'll be glad to give root access to anyone willing to help. Any ideas on what to do next? Thanks Bob Smith bsmith at linuxtoys dot org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling from * to fwd
Hi i was trying to call 17009978275 which is my Fwd
line on my notebook from Asterisk and i keep getting this message on the
console.
-- Executing Dial("Zap/2-1", "[EMAIL PROTECTED]/[EMAIL PROTECTED]")
in new stack -- Called
@iaxtel.com/[EMAIL PROTECTED]WARNING[1150495040]: File chan_iax2.c, Line
4547 (socket_read): I don't know how to authenticate rob to
69.73.19.178 -- Hungup
'IAX2[69.73.19.178:4569]/2'
Is there anything i'm missing here? Any insight
would be very great ful
thanks.
Re: [Asterisk-Users] time to build an open phone?
Bruce Ferrell wrote: only problem is the protocol stack isn't open. Good chip We would only use that code for examples of how to bolt in the bottom end drivers. We would roll out our own os/scheduler, a little * code and drivers. I have not found a data sheet for the 1001 yet, but I did look at the 1050. Great looking chip. Just a few questions. Any idea how much it cost? It does have a jtag debug interface. Do you know of any gui debuggers running on linux for this chip? We really need a nice friendly debug environment to make it as easy to write/load/debug code as doing it for linux. CW_ASN wrote: How about to build an ip phone with this IC? http://focus.ti.com/docs/apps/catalog/general/applications.jhtml?templateId= 969path=templatedata/cm/general/data/bband_ipphone_tnetv1001 - Original Message - From: Bob Knight [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 4:30 PM Subject: [Asterisk-Users] time to build an open phone? Open software seems to work. Why don't we try it with hardware. 1. pick an embedded processor. It should have a nice linux gui support (like x jtag debugger). 2. pick a linux based cad system we all have easy access to and place schematics under cvs. 3. pick some type of gpio or serial interface for keyboard/display. 4. pick some basic functionality. 5. code it up. A stripped down *. Let everyone do their own thing with the expensive part. Tooling/packaging. We could let Digium be the distributor, so they are not left out of the loop. A board set would be offered with NO support. If Digium wants no part of it, we just build them on our own for our own use or sell them on ebay. What we would provide is schematics and source code. Everyone can take this to their favorite fab house and crank em out. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sip phones on the same extension?
Hello. I'm sorry I wasn't clear. In the original question I asked, I said that I found the same work around that was suggested on this list. Since the suggestion was there, and since I had posted my original work around in my original message, I thought there was something that I was missing with respect to the work around itself, and I was asking for clarification. The solution I have working at the moment, is exactly the one which was offered up. However, I don't like it, because it's a solution which doesn't scale. I was trying to assertain if Asterisk would do what I was envisioning, and which SER does very well, and if the fact that I couldn't think of a way was merely due to my lack of knowledge about Asterisk. It sounds like Asterisk doesn't work like this right now. Do folks think they'd find such a feature useful if I coded it up and sent it back to Digium? -thanks -Brian Message: 5 Date: Thu, 25 Dec 2003 13:20:51 -0600 (CST) From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Sip phones on the same extension? Reply-To: [EMAIL PROTECTED] Because the one you came up with isn't possible with asterisk at this time. On Thu, 25 Dec 2003, Brian Buhrow wrote: Hello. I think I understand your suggestion, but don't understand how that's any different than the one I came up with. What I want, is to be able to define a specific extension, and then have any external SIP phones register with that extension that want to. It's important that multiple phones be able to register with the same extension simultaneously. Then, I can define something like: exten = 300,1,Dial(SIP/300,15|t) and all phones registered to extension SIP/300 will ring. The number of phones existing on that extension at any given time is unknown, and Asterisk should be able to keep a list of all devices which are currently registered on a given extension, even if it has seen another device register to the same extension. To guard against number stealing, one could restrict the registration of a given phone number to a single password, but allow that password to be used as often and from where ever. So, for example, if my extension is 300, and my password is JustForFun, I should be able to program any number of SIP phones to register as extension 300, and as long as they know the magic password, JustForFun, Asterisk will permit all of them to register as SIP/300. Then, if someone calls 300, they'll all ring simultaneously, and which ever phone gets picked up first, gets the call. This doesn't appear to be how Asterisk works at the moment. Am I wrong about this? -Brian Message: 9 From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip phones on the same extension? Date: Wed, 24 Dec 2003 13:24:53 -0600 Reply-To: [EMAIL PROTECTED] In sip.conf: [phone1] type=peer host=dynamic [phone2] type=peer host=dynamic [phone3] type=peer host=dynamic in extensions.conf: [default] exten = 0,1,Dial(SIP/phone1SIP/phone2SIP/phone3,30,T) -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) Help With 7960's Speed-dials
Cisco part number: CON-SNT-CP7960 SMARTNET 8x5xNBD Svc, IP Phone 7960, Bus Set (w/ User Lic) $8.00 On Tue, 23 Dec 2003 15:52:32 -0800, Paul Mahler wrote If you purchase a new telephone, the warranty is more like $15. It's more for used phones. Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Tuesday, December 23, 2003 2:17 PM To: asterisk- [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) Help With 7960's Speed-dials 7914's don't work with SIP. SCCP only. And why do people keep talking about this 8 dollar CCO account ... Its a service contract on the Cisco ATA-186. The one for the 79XX's are over 80.00/yr bkw On Tue, 23 Dec 2003, Lists wrote: How do you reset the unit without pulling out the plug. The easiest way to get the info you are looking for, is to get an 8 buck CCO account. On Tue, 23 Dec 2003, Adthrawn wrote: Hi, Has anybody been successful in running the 7914 expansion unit for the Cisco 7960G IP phone? For anybody unaware of what the expansion unit does, it provides 14 additional buttons, with an LCD display. The idea, is that with an expansion unit (a 7960 can take upto 2 of these units), a user can either assign more speed-dial's, or can monitor line status/account status. So, you can either register a speed-dial or register another account. The problem I've found so far, is that speed-dials are not programmed on the phone, but are instead handled by the Call Manager software (not on a user basis, but on a phone, MAC address basis). Likewise, plugging the 7914 unit into the phone on an Asterisk PBX, just brings up 14 red lights (the buttons also light-up red, blue or green), which according to the hidden technical documentation, indicates that the Call Manager is not registering the unit. I can't work out if it's short of firmware embedded in the Call Manager, whether it's searching for a configuration file on the TFTP (Cisco phones need a TFTP to get their settings and SIP firmware), whether it's not happy with the phone being a SIP version, or whether I'm doing something wrong. I've had to learn about the 7960's configuration the hard way, and despite their useless technical documents, have managed to configure most settings. There's quite a bit of extra configuration for the 7960 I'd love to get to, and would like help or advice on. Things like directory services, screen logo, the 7914 and more! If anybody is interested, I have resources and files to; convert from Cisco Skinny/MGCP to a SIP version, how to configure the VoiceMail indicator lamp, special key combinations to reset the unit (without pulling the plug out) and locking/unlocking the preferences, configuring the voicemail speed-dial Any help or advice, please let me know! Regards, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: FAX detection Problem
This was a problem that got introduced some time around (I believe) 12/15/2003 -- It might have been fixed by now, you should do a fresh cvs checkout and try again. If not, do a cvs checkout from around 12/08/2003, that version worked for me. (btw, the issue is some updates that were applied to dsp.c) Thanks, Pat - Original Message - From: Hisham Allam [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 8:40 AM Subject: [Asterisk-Users] Fw: FAX detection Problem Hi, I am using asterisk with PRI TE410P card. Everything work fine, except that every time I receive a call, I get File chan_zap.c, Line 3546 (zt_read): Fax detected although they are just normal calls. How can i set the threshold of fax detection. What might be wrong that tone_detect function always detect a fax tone. Help please Hisham. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 BTXML
On Wed, 24 Dec 2003, Cameron Palmer wrote: Has anyone devloped some perl scripts for sending Directory and Services information to the Cisco 7960? The older thread I googled is giving me fits. My understanding is that the SIP images only use BTXML internally. I think the MGCP can use BTXML-driven services. The directory should be available (An example is http://lists.digium.com/pipermail/asterisk-users/2003-May/013013.html) For 'Services' you have to use a subset of the CMXML (I think maybe v3.0?) with SIP, it doesn't support all the latest stuff. Good luck, Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Red Alarm on X100P
I'm having sporadic problems with my X100P card. When trying trying to call out I get a message on the console that the channel is busy even though I have bounced asterisk. It seems the card grabs the line and doesn't let go. When I try to call the number I get a busy signal. If I perform a rmmod wcfxo and restart asterisk everything retruns to normal. I have a TDP400P installed along with the X100P. Any ideas what may be happening. Log Entries at time of Alarm Dec 25 21:42:13 WARNING[663568]: File chan_zap.c, Line 4468 (ss_thread): CallerID returned with error on channel 'Zap/1-1' Dec 25 21:42:31 WARNING[663568]: File chan_zap.c, Line 2779 (zt_handle_event): Detected alarm on channel 1: Red Alarm Dec 25 21:42:50 WARNING[8192]: File config.c, Line 579 (cfg_process): No '=' (equal sign) in line 34 of mgcp.conf Dec 25 21:42:50 WARNING[8192]: File chan_iax2.c, Line 5453 (set_config): Ignoring port for now Dec 25 21:42:50 WARNING[8192]: File chan_oss.c, Line 429 (soundcard_init): Unable to open /dev/dsp: No such device Dec 25 21:42:50 WARNING[8192]: File cdr_unixodbc.c, Line 196 (unixodbc_load_module): cdr_unixodbc: Unable to load config for unixODBC CDR's: cdr_unixodbc.conf Dec 25 21:43:11 NOTICE[122896]: File app_dial.c, Line 506 (dial_exec): Unable to create channel of type 'ZAP' Dec 25 21:43:21 WARNING[122896]: File pbx.c, Line 1829 (ast_pbx_run): Timeout, but no rule 't' in context 'home' - 1 Interrupts CPU0 0: 54778940 XT-PIC timer 1: 5130 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 0 XT-PIC usb-ohci 5: 106532071 XT-PIC wcfxo 8: 1 XT-PIC rtc 10: 107102671 XT-PIC eth0, wcfxs 11: 0 XT-PIC ide2 12:277 XT-PIC PS/2 Mouse 14: 107761 XT-PIC ide0 NMI: 1 ERR: 0 rgds burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
