Re: [Asterisk-Users] x100p dropping incoming calls

[Jeffrey Gustafson]

On Tue, 2004-02-17 at 07:44, dkwok wrote:
> I have been experiencing hung up when answering incoming calls through 
> x100p.

I've noticed that in one system that I've been running x101p's on I was
getting a lot of hangups.  Sometimes there was loud buzzing.  It turns
out that the systems (a Dell) has a power supply right over the PCI
bus.  I moved the HDD, and x101p's to a traditional ATX system on the
same phone lines and the problems went away.
...Jeff


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RE: [Asterisk-Users] Cisco 7940 no audio

[Craig Waddington]

Title: Message








Thanks for the help, I am currently running
the latest sip image, it seems to have fixed a lot of bugs..

 

I did a full rebuild of the server and
used the stable cvs, all is working perfectly now. I am actually amazed at the
quality of the call using the diva card/capi through isdn.

 

 

 

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Sent: 18 April 2004 04:44
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Cisco 7940 no audio



 



Try upgrading to SIP 6.3.  I heard
from someone on the IRC Channel about this problem and 6.3 resolved it





 





 





-gcc





-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington
Posted At: Friday, April 16, 2004
1:04 PM
Posted To: Asterisk User Group
Conversation: Cisco 7940 no audio
Subject: [Asterisk-Users] Cisco
7940 no audio





When we receive or make a call to the outside – they
can hear us, but we cant hear them.

 

It may work 1 of 20 times. I have set canreinvite=no 
and looked at many sites but cannot track down this problem.

 

Current setup:

 

Isdn Eicon Diva card / Capi -> Asterisk à network.

 

I have tried adjusting the RTP port in rtp.conf with the
Cisco default ports, no luck.

 

Anyone had this problem, and has a fix?

 

Thanks.

Re: [Asterisk-Users] Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)

[Chris Maresca]

I'm not surprised.  My wife is an editor at on of CMP's more cluefull
publications, and, even there, she's fighting against general stupidity
evey day.

Most journalists, esp. tech journalists, are pretty clueless.

Chris.

On Sat, 17 Apr 2004, JR Richardson wrote:

> * Brethren,
> 
> It's a sad day in our community.  Please join me in a moment of silence for
> the death of responsible journalism.  Silence.good
> enough.
> 
> This article goes on to tell about Pingtel's announcement of forming the
> "first open source community aimed at creating SIP based servers".
> 
> http://www.networkmagazine.com/shared/article/showArticle.jhtml?articleId=18
> 900066&classroom=
> 
> Yipeee!
> 
> I am embarrassed and appalled with the lack of recognition or mention of all
> the tremendous work in this "already existing community".  I'm writing a
> letter to the editor and encourage all of you to do the same.
> 
> Send your comments to [EMAIL PROTECTED]
> 
> Mark Spencer,
> 
> I caught your presentation at the Linux-Kongress
> http://graphics.cs.uni-sb.de/VCORE/recordings.html .  I want to personally
> thank you for validating all that I have been doing to promote Asterisk to
> anyone and everyone who will take the time to listen.  When I talk about it
> and show off working systems in action, people get excited and are quite
> impressed.  Having your 30 min presentation to go along with my demo
> increases my credibility 10 fold.
> 
> Thank you all for contributing to this great community.
> 
> JR
> 
> 
> 
> 


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[Asterisk-Users] FWD<->NAT<->* config info

[William J Mandra]

Here is my sip.conf and extensions.conf which allow me inbound and outbound
calling between * and Freeworld Dialup, with * behind a NAT.

;
; SIP Configuration for Asterisk
;

[general]
disallow=all
allow=ulaw
port=5060   ; Port to bind to
bindaddr=0.0.0.0; Address to bind SIP channel to
externip=xxx.xxx.xxx.xxx
localnet=172.16.1.0
localmask=255.255.255.0
context=inbound-sip ; Default context for incoming calls
maxexpirey=180
defaultexpirey=160
tos=reliability
srvlookup=yes
register => 290805:[EMAIL PROTECTED]/290805
register => 293440:[EMAIL PROTECTED]/293440

[fwd1]
type=friend
secret=secret
username=293440
fromuser=293440
fromdomain=fwd.pulver.com
host=fwd.pulver.com
dtmfmode=inband
nat=yes
canreinvite=no

[fwd2]
type=friend
secret=secret
username=290805
fromuser=290805
fromdomain=fwd.pulver.com
host=fwd.pulver.com
dtmfmode=inband
nat=yes
canreinvite=no

[phone17]
disallow=all
allow=ulaw
type=friend
host=dynamic
defaultip=172.16.1.17
dtmfmode=inband
secret=voip17
mailbox=2206
context=home
callerid="Bill Mandra" <2206>
nat=no

[phone18]
disallow=all
allow=ulaw
type=friend
host=dynamic
defaultip=172.16.1.18
dtmfmode=inband
secret=voip18
mailbox=2201
context=home
callerid="Kitchen" <2204>
nat=no

;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;

[general]
static=yes
writeprotect=no

[globals]
DIALOUTANALOG=Zap/1
MAINPHONE=Zap/2
JESSICA=Zap/3
CHRISTOPHER=Zap/4
PORCH=Zap/5
KITCHEN=SIP/phone18
BILL=SIP/phone17

FWDUSERID1=290805
FWD1USERNAME=William Mandra
FWDUSERID2=293440
FWD2USERNAME=Donna Mandra
FWDPREFIX=*

HOMENUMBER=XX

BILLCELLPHONE=9XX
MOMCELLPHONE=1XX
;JESSCELLPHONE=1XX

;
; Macros
;

[macro-fastbusy]
exten => s,1,Answer
exten => s,2,Wait,1
exten => s,3,Playback(ss-noservice)
exten => s,4,Wait(30)
exten => s,5,Hangup

[macro-dialoutsip]
exten => s,1,SetCallerID(${FWDUSERID2})
exten => s,2,SetCIDName(${FWD2USERNAME})
exten => s,3,Dial(SIP/[EMAIL PROTECTED],70)
exten => s,4,Macro(fastbusy)
exten => s,5,Hangup
exten => s,104,Macro(fastbusy)
exten => s,105,Wait,3
exten => s,106,Playtones(congestion)
exten => s,107,Wait,30
exten => s,108,Hangup

[macro-billcellfwdoutsip2]
exten => s,1,SetCallerID(${ARG2})
exten => s,2,Dial(SIP/[EMAIL PROTECTED],20)
exten => s,3,Goto(local,2206,4)
exten => s,102,Goto(local,2206,4

;
; Outbound
;

[operator]
exten => 0,1,Dial(${DIALOUTANALOG}/${EXTEN},70)
exten => 0,2,Macro(fastbusy)
exten => 0,102,Playback(ss-noservice)
exten => 0,103,Macro(fastbusy)

[e911]
exten => 911,1,Dial(${DIALOUTANALOG}/${EXTEN})
exten => 911,2,Macro(fastbusy)
exten => 911,102,Playback(ss-noservice)
exten => 911,103,Macro(fastbusy)

[forced-analog]
exten => _9.,1,Dial(${DIALOUTANALOG}/${EXTEN:1},70)
exten => _9.,2,Macro(fastbusy)
exten => _9.,102,Macro(fastbusy)

[fwd1-out]
exten => _8.,1,SetCallerID(${FWDUSERID2})
exten => _8.,2,SetCIDName(${FWD2USERNAME})
exten => _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)
exten => _8.,4,Macro(fastbusy)
exten => _8.,5,Hangup

[fwd2-out-pvt]
exten => _7.,1,SetCallerID(${FWDUSERID1})
exten => _7.,2,SetCIDName(${FWD1USERNAME})
exten => _7.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)
exten => _7.,4,Macro(fastbusy)
exten => _7.,5,Hangup

[information]
exten => 411,1,Dial(${DIALOUTANALOG}/${EXTEN},70)
exten => 411,2,Macro(fastbusy)
exten => 411,102,Playback(ss-noservice)
exten => 411,103,Macro(fastbusy)

[pstn-local]
exten => _1973.,1,Dial(${DIALOUTANALOG}/${EXTEN:1})
exten => _1973.,2,Macro(fastbusy)
exten => _1973.,102,Macro(dialoutsip,${EXTEN})

exten => _1201.,1,Dial(${DIALOUTANALOG}/${EXTEN})
exten => _1201.,2,Macro(fastbusy)
exten => _1201.,102,Macro(dialoutsip,${EXTEN})

[pstn-local-toll]
exten => _11973.,1,Dial(${DIALOUTANALOG}/${EXTEN:1})
exten => _11973.,2,Marco(fastbusy)
exten => _11973.,102,Macro(dialoutsip,${EXTEN:1})

exten => _11201.,1,Dial(${DIALOUTANALOG}/$EXTEN:1})
exten => _11201.,2,Macro(fastbusy)
exten => _11201.,102,Macro(dialoutsip,${EXTEN:1})

[toll-free]
exten => _1888.,1,Dial(${DIALOUTANALOG}/${EXTEN})
exten => _1888.,2,Macro(fastbusy)
exten => _1888.,102,Macro(dialoutsip,${EXTEN})

exten => _1877.,1,Dial(${DIALOUTANALOG}/$EXTEN})
exten => _1877.,2,Macro(fastbusy)
exten => _1877.,102,Macro(dialoutsip,${EXTEN})

exten => _1855.,1,Dial(${DIALOUTANALOG}/{${EXTEN})
exten => _1855.,2,Macro(fastbusy)
exten => _1855.,102,Macro(dialoutsip,${EXTEN})

exten => _1800.,1,Dial(${DIALOUTANALOG}/${EXTEN})
exten => _1800.,2,Macro(fastbusy)
exten => _1800.,102,Macro(dialoutsip,${EXTEN})

[long-distance]
exten => _1XX,1,Macro(dialoutsip,${EXTEN})
exten => _1XX,2,Macro(fastbusy)
exten => _1XX,102,Dial(${DIALOUTANALOG}/${EXTEN})
exten => _1XX,103,Macro(fastbusy)

[home]
include => operator
include => e911
include => forced-analog
include => fwd1-out
include => fwd2-out-pvt
include => information
include => local
include => pstn-local
include => pstn

RE: [Asterisk-Users] Cisco 7940 no audio

[AstGrp]

Title: Message



Try 
upgrading to SIP 6.3.  I heard from someone on the IRC Channel about this 
problem and 6.3 resolved it
 
 
-gcc

-Original Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Craig 
WaddingtonPosted At: Friday, April 16, 2004 1:04 PMPosted 
To: Asterisk User GroupConversation: Cisco 7940 no 
audioSubject: [Asterisk-Users] Cisco 7940 no 
audio

  
  When we receive or make a call to 
  the outside – they can hear us, but we cant hear 
  them.
   
  It may work 1 of 20 times. I have 
  set canreinvite=no  and looked at many sites but cannot track down this 
  problem.
   
  Current 
  setup:
   
  Isdn Eicon Diva card / Capi -> 
  Asterisk à 
  network.
   
  I have tried adjusting the RTP 
  port in rtp.conf with the Cisco default ports, no 
  luck.
   
  Anyone had this problem, and has a 
  fix?
   
  Thanks.

Re: [Asterisk-Users] VOIP Spam

[Duane]

Tracy R Reed wrote:

No, I haven't. And you are right it is highly unlikely. Knowing that
someone was going to want to get a key signed, putting the bogus info
where they would find it, tricking someone into calling you and giving
them a bogus key, etc. is all very difficult. I think we are going to have
to give up the notion of 100% security and accept the very small chance
(orders of magnitude smaller than now) of someone being fooled if we ever
want to get this stuff deployed.
ongoing man in the middle attacks aren't impossible, the FBI's carnivore 
system is all over the place and in theory could not only sniff but 
inject... Then again there are other methods at the disposal of 
governments...

Since most cpu's out there in the world spend 80% of their time idle doing
nothing anyway I don't think it would be quite this bad. :)
What about asterisk servers that are already under load, this would 
multiply the effect, yes most servers would idle most of the time, but 
if you have periods of peak activity this would compound any existing 
problems you get from this...

Ah. I haven't given too much thought about how it interacts with phone
systems yet. I'll ponder this one.
I believe there is an RFC on PGP use in browsers, I don't know of anyone 
actually implementing it however...

Very cool. I am reading up on this stuff.
We wanted a method of dynamic routing so we didn't have an ever growing 
list of extensions and IAX/SIP items not to mention getting away from 
single points of failure that if a service is down you're out of luck, 
it seemed like enum.164 is the only solution to this problem. We wanted 
to do things in such away we could be relatively certain the person we 
were calling was who we were expecting and not a telemarketer etc etc 
that had hijacked a heap of numbers... As far as I'm aware no other enum 
system (even ITU's) currently implements anything that comes close to 
what we were after...

Indeed. It was just an example of the mail vendors successfully forcing
something on everyong.
The thing is it didn't stop normal text posts, so yes it tacked added 
functionality on top without denying the existing system, you're 
suggestion doesn't take that into account...

That is fine. The mail administrator can read everything they type into
the server anyhow. He can bug their keyboard if he wants. 
Not if you encrypt email at the mail client... He can't bug a remote 
keyboard... Some of the PKI hardware devices are implemented in a 
keyboard and when access the certificate the keyboard direct key strokes 
 directly to the hardware reader rather then via the PC...

I doubt they would because it would make spamming much more expensive.
Some might but it makes it much less likely and kills their profits which
removes the incentive.
What cost? It's trivial to generate both PGP and self signed PKI keys 
using openssl toolkit, spammers could easily pay someone to grab a new 
domain/email/certificate daily, $10 in wages? If they get $1000 in 
profit from $10 in expenses they'd do it...

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
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Re: [Asterisk-Users] VOIP Spam

[Tracy R Reed]

On Sun, Apr 18, 2004 at 11:13:27AM +1000, Duane spake thusly:
> But have you ever met face to face with an employee from a CA and 
> verified they were an employee or just grabbed the info from their 
> website and assumed there was no man in the middle attack sending you an 
> alternate key/fingerprint (yes I know this is highly unlikely however 
> high profile targets would be possible at some point, how lucky do you 
> feel? :)

No, I haven't. And you are right it is highly unlikely. Knowing that
someone was going to want to get a key signed, putting the bogus info
where they would find it, tricking someone into calling you and giving
them a bogus key, etc. is all very difficult. I think we are going to have
to give up the notion of 100% security and accept the very small chance
(orders of magnitude smaller than now) of someone being fooled if we ever
want to get this stuff deployed.

> If we make up some number, I have seen figures for websites can't seem 
> to find them at present, anyways say a TLS/SSL operation uses 8x more 
> CPU power then a non-TLS connection, this means if you are running a 
> voip to pstn service or in an office environment with a large amount of 
> handsets/calls you need 8x more servers or 8x less clients so there is 
> definitely a cost involved there even if CPUs etc are cheaper...

Since most cpu's out there in the world spend 80% of their time idle doing
nothing anyway I don't think it would be quite this bad. :)

> As for hostname matching, you run an enum check on a phone number, it 
> returns a URL... say iaxtel.com... you connect to it and it then says 

Ah. I haven't given too much thought about how it interacts with phone
systems yet. I'll ponder this one.

> Umm just a side note, we have a working enum.164 website/dns ( 
> http://e164.org ) service that now does pstn verification (due 
> diligence) by calling you and reading out a pin number, currently a 
> little rough and we need a few IVR records (which will within the next 
> few days), and need to update the documentation on the website, however 
> it does seem to work reasonably well...

Very cool. I am reading up on this stuff.

> Most HTML emails have a non-html component as well, and the amount of 
> people that dislike html emails I don't see this as a good comparison ;)

Indeed. It was just an example of the mail vendors successfully forcing
something on everyong.

> You can't enforce crypto from a MTA/MUA point of view, there is a whole 
> bunch of complications if you force certificates on people like you'd 
> have to get them a public/private key pair and then well it wouldn't be 
> so private...

That is fine. The mail administrator can read everything they type into
the server anyhow. He can bug their keyboard if he wants. 

> The reason they would is to beat the virus/spam filters currently in 
> operation at a MTA level, they would be rendered useless, at present all 
> you need is a valid email address to get a certificate issued from a CA 
> with their root certificate in most/all current email clients...

I doubt they would because it would make spamming much more expensive.
Some might but it makes it much less likely and kills their profits which
removes the incentive.

-- 
Tracy Reed The attachment is a digital signature.
http://copilotconsulting.com   More info: http://copilotconsulting.com/sig


pgp0.pgp
Description: PGP signature

Re: [Asterisk-Users] FW: Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)

[tmpm]

I fired off a note to them at both addresses as well, and asked them check 
out astricon as well..links included. (clues to the clueless mode...)

I read their article on Pingtel going OS while sitting on the porcelain 
furniture the other morning, and thought they might be about to discover 
fire and the wheel, the post here reminded me to follow up on mentioning * 
to them...Im sure they have been enlightened by now (thump). Now it remains 
to see if they're going to cover a product that doesn't purchase pricey 
space in their rag.

Marc



At 11:51 4/17/2004, you wrote:
The [EMAIL PROTECTED] appears to be broken.  I dug around the magazine
contacts and found Doug Allen, senior editor, you can send comments to
[EMAIL PROTECTED] .  I didn't get a bounce back from that e-mail so I assume
it made it to the editor.
JR



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Re: [Asterisk-Users] VOIP Spam

[Duane]

Tracy R Reed wrote:
Same way I know someones key is theirs by the pgp fingerprint. It's well
publicized and they use it quite a bit. 
But have you ever met face to face with an employee from a CA and 
verified they were an employee or just grabbed the info from their 
website and assumed there was no man in the middle attack sending you an 
alternate key/fingerprint (yes I know this is highly unlikely however 
high profile targets would be possible at some point, how lucky do you 
feel? :)

Not sure what you mean by matching an email address against a hostname but
a lot of the crypto accelerator cards implement fundamentals that could be
used in either system and more specific hardware devices would certainly
come along if more people used it. But with the speed and SIMD capability
modern cpu's I'm not too concerned either way.
If we make up some number, I have seen figures for websites can't seem 
to find them at present, anyways say a TLS/SSL operation uses 8x more 
CPU power then a non-TLS connection, this means if you are running a 
voip to pstn service or in an office environment with a large amount of 
handsets/calls you need 8x more servers or 8x less clients so there is 
definitely a cost involved there even if CPUs etc are cheaper...

As for hostname matching, you run an enum check on a phone number, it 
returns a URL... say iaxtel.com... you connect to it and it then says 
I'm able to provide encryption here is my public certificate, you grab 
the certificate and it has [EMAIL PROTECTED], which doesn't match 
iaxtel.com, or even if it was [EMAIL PROTECTED] how do you know that 
email account should be able to say I validate this server is the one 
you should be talking to and that DNS hasn't been hijacked? PGP can't 
easily deal with this, and if you start connecting to foreign asterisk 
servers via enum services how can you validate them without prior 
relationships? While PKI may be flawed it is better then the current 
alternatives at present...

Umm just a side note, we have a working enum.164 website/dns ( 
http://e164.org ) service that now does pstn verification (due 
diligence) by calling you and reading out a pin number, currently a 
little rough and we need a few IVR records (which will within the next 
few days), and need to update the documentation on the website, however 
it does seem to work reasonably well...

If the MUA authors forced the issue everyone would use crypto. Look at
what Outlook did for html mail. Encrypted spam would be difficult for the
spammers to do. It would consume huge resources, make spam a lot more
expensive, and if they signed the spam with a trusted key such that my MUA
trusted them you can be sure the signer would revoke his signature lest he
get the signatures on his own key revoked by someone.
Most HTML emails have a non-html component as well, and the amount of 
people that dislike html emails I don't see this as a good comparison ;)

You can't enforce crypto from a MTA/MUA point of view, there is a whole 
bunch of complications if you force certificates on people like you'd 
have to get them a public/private key pair and then well it wouldn't be 
so private...

Some very interesting points. Especially about encrypted spam confounding
the government. Although I doubt they would encrypt spam it does add chaff
to the wheat to help hide us all. Just like the everyone sending their
letters in envelopes instead of on postcards analogy.
The reason they would is to beat the virus/spam filters currently in 
operation at a MTA level, they would be rendered useless, at present all 
you need is a valid email address to get a certificate issued from a CA 
with their root certificate in most/all current email clients...

On a per capita basis it's not nearly as often as computers get broken
into. :) Whenever anyone bothers to try to physically secure their stuff
they usually do a pretty good job. Not so with computers.
maybe cars being stolen was a better suggestion, break a window and 
you're in unless they have an alarm (computers can also have "alarms" in 
this sense)

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
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Re: [Asterisk-Users] VOIP Spam

[Tracy R Reed]

On Sun, Apr 18, 2004 at 10:22:08AM +1000, Duane spake thusly:
> Just a little matter of key distribution, how do you know the CA key 
> given to you is actually the CA? Especially since Thawte no longer does 
> PGP key signing and verisign is making too much money from PKI...

Same way I know someones key is theirs by the pgp fingerprint. It's well
publicized and they use it quite a bit. 

> The are a number of issues with the PGP model, it contains an email 
> address, how do you match that against a hostname? As far as I know 
> there is no hardware devices to store pgp keys, or accelerator cards 
> (crypto does chew through a bit of CPU) both devices exist for PKI 
> certificates/keys...

Not sure what you mean by matching an email address against a hostname but
a lot of the crypto accelerator cards implement fundamentals that could be
used in either system and more specific hardware devices would certainly
come along if more people used it. But with the speed and SIMD capability
modern cpu's I'm not too concerned either way.

> Mozilla Foundation, it's developers and direct support staff 
> (paid/unpaid) are currently reviewing about a dozen or so CAs for 
> inclusion in their browser, CAcert is one of them, which will be good 
> for the community if we can get in, as we provide all certificates for 
> free...

Very cool.

> This would be good and bad, if you force the issue you will end up with 
> 2 things, less people being able to email you, and in the very long term 
> encrypted spam so we end up with them beating scanners that way...

If the MUA authors forced the issue everyone would use crypto. Look at
what Outlook did for html mail. Encrypted spam would be difficult for the
spammers to do. It would consume huge resources, make spam a lot more
expensive, and if they signed the spam with a trusted key such that my MUA
trusted them you can be sure the signer would revoke his signature lest he
get the signatures on his own key revoked by someone.

> and runs the program in the zip file infecting themselves... So I 
> foresee a lot of missuses from crypto as much as anything else if/when 
> the general populace gets into it...

Some very interesting points. Especially about encrypted spam confounding
the government. Although I doubt they would encrypt spam it does add chaff
to the wheat to help hide us all. Just like the everyone sending their
letters in envelopes instead of on postcards analogy.

> So that's why people still get broken into and all their contents stolen :)

On a per capita basis it's not nearly as often as computers get broken
into. :) Whenever anyone bothers to try to physically secure their stuff
they usually do a pretty good job. Not so with computers.

-- 
Tracy Reed The attachment is a digital signature.
http://copilotconsulting.com   More info: http://copilotconsulting.com/sig


pgp0.pgp
Description: PGP signature

Re: [Asterisk-Users] VOIP Spam

[Duane]

Tracy R Reed wrote:

I prefer the PGP model because it includes the CA model. That is to say
that you can still have a CA within the PGP model. Both myself and my
colleague from Africa could pay a central CA we both trust (Verisign,
Thawte, whoever) to sign our keys and connect us in the web of trust. 
Just a little matter of key distribution, how do you know the CA key 
given to you is actually the CA? Especially since Thawte no longer does 
PGP key signing and verisign is making too much money from PKI...

The are a number of issues with the PGP model, it contains an email 
address, how do you match that against a hostname? As far as I know 
there is no hardware devices to store pgp keys, or accelerator cards 
(crypto does chew through a bit of CPU) both devices exist for PKI 
certificates/keys...

Yep. We end up with collusion which prevents competition in the CA space.
It's a shame common browsers only support a few select CA's.
Mozilla Foundation, it's developers and direct support staff 
(paid/unpaid) are currently reviewing about a dozen or so CAs for 
inclusion in their browser, CAcert is one of them, which will be good 
for the community if we can get in, as we provide all certificates for 
free...

I think huge improvements are needed in software to handle this. We really
need to encourage everyone to use signatures etc. and make them so
prevalent that email programs etc. will simply refuse to accept or display
non-signed and authenticated messages/connections/whatever.
This would be good and bad, if you force the issue you will end up with 
2 things, less people being able to email you, and in the very long term 
encrypted spam so we end up with them beating scanners that way...

It's a balancing act, push things one way you have to even them up the 
other...

There will be 3 consequences from mass encryption adoption, encrypted 
spam, and forcing governments to do due diligence as they will no longer 
be able to simply passively collect any traffic passing their monitoring 
devices, they'd have to go back to a situation of only targeting people 
they really had to, this is obviously a good thing, and even the 
encrypted spam, while being annoying would tick any gov surveillance off 
due to sheer number of spam messages that could be encrypted that would 
be the equivalent of noise to them... 3rd is a little more serious, 
since most people wouldn't care about due diligence with crypto they 
wouldn't care if they did it right or who they accepted, this is clearly 
visible from the latest virus trends where they exploit human ignorance, 
greed and stupidity not exploiting computer software. What else could it 
be called where a person opens a zip file, uses a password in the email, 
and runs the program in the zip file infecting themselves... So I 
foresee a lot of missuses from crypto as much as anything else if/when 
the general populace gets into it...

Indeed but that is a far better situation than we are in now. We know very
well how to deal with physical security due to thousands of years of doing
so.
So that's why people still get broken into and all their contents stolen :)

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
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Re: [Asterisk-Users] VOIP Spam

[Tracy R Reed]

On Sun, Apr 18, 2004 at 09:31:48AM +1000, Duane spake thusly:
> be sure more are issued on a correct basis. PGP model if you lived in 
> say Africa and wanted to communicate with someone in South America with 
> little or no prior relationship and you wanted to be sure the 
> communication wouldn't be intercepted you have 2 choices, fly to meet 
> each other or gain trust you both are who you say you are from an 
> impartial 3rd party that if it did it's job correct would be correct.

I prefer the PGP model because it includes the CA model. That is to say
that you can still have a CA within the PGP model. Both myself and my
colleague from Africa could pay a central CA we both trust (Verisign,
Thawte, whoever) to sign our keys and connect us in the web of trust. 

> *BUT*, and it's a very big but, there is 2 or 3 flaws in the PKI model, 
> firstly there is a crap load of money usually involved, where there is 
> money there is usually corruption, at this stage of the game the PKI 
> industry has had very little over all impact, something like 0.3% of web 

Yep. We end up with collusion which prevents competition in the CA space.
It's a shame common browsers only support a few select CA's.

> PGP model would obviously be an advantage in this case, but most people 
> don't have a clue about security practises and get so many pop-up 
> warning messages they simply click ok to whatever comes up.

I think huge improvements are needed in software to handle this. We really
need to encourage everyone to use signatures etc. and make them so
prevalent that email programs etc. will simply refuse to accept or display
non-signed and authenticated messages/connections/whatever.

> The other flaw is safe keeping of certificates, unless you have a 
> hardware device, the more difficult you make it for someone to break 
> digital security will only make them turn round and break physical 
> security...

Indeed but that is a far better situation than we are in now. We know very
well how to deal with physical security due to thousands of years of doing
so.

-- 
Tracy Reed The attachment is a digital signature.
http://copilotconsulting.com   More info: http://copilotconsulting.com/sig


pgp0.pgp
Description: PGP signature

[Asterisk-Users] Capi & MSN routing.

[Craig Waddington]

Kudos to the CAPI developers.

 

I would like to have multiple MSN’s on my ISDN Bri
lines.

 

I see all the cool features but cannot find any examples or
guides to build from.

 

Currently running Diva Eicon Cards with CAPI from http://www.junghanns.net

 

I would like to route calls to sip phones via msn.

 

Set up callgroups etc.

 

Can anyone share some some examples I can build from. 

 

I want to use some of the features the capi drivers support.

RE: [Asterisk-Users] Snom 200 Admin Password

[Christian Stredicke]

We did not see so far that a provider would pay the phone so it's only fair
that the user has the ability for example to change the provider. The user
owns the phone!

Btw take a look at http://www.snom.com/faq_en.php,
http://www.snom.com/faq/FAQ-04-03-24-sf.pdf. 

CS

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Chris Orme
> Sent: Saturday, April 17, 2004 11:39 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Snom 200 Admin Password
> 
> The tftp suggestion you received is well worth trying :-)  I didn't know
> that was possible.
> 
> Although I wonder if it works as it might mean that carriers deploying the
> snom may not be able to properly lock their phones perhaps??
> 
> Chris
> 
> On Sat, 17 Apr 2004, WipeOut wrote:
> 
> > Chris Orme wrote:
> >
> > >Hi.
> > >
> > >Did you buy the phone or get it second hand ?   If second hand do you
> have
> > >any paperwork from the person you bought it from and did they buy it
> > >through official distribution?
> > >
> > >If you got it through distribution I would am fairly sure your vendor
> > >might be able to help ?
> > >
> > >I have a rough idea of how it would be possible but I would think
> you'll
> > >probably have to prove ownership as this password is how carriers lock
> > >their phones.  If you got it from a carrier I imagine you might
> possibly
> > >have to pay them an unlock charge so you can change carriers.
> > >
> > >Or did you accidently set the admin password?
> > >
> > >
> > >Chris
> > >
> > >
> > >
> > I bought the phone new about a year ago so its not provider locked..
> >
> > I set the password to be nothing (I think) and then I set admin mode
> > off, then when I tried to get into the admin area I couldn't, it would
> > seem that either there is a bug that doesn't allow a blank password or
> > it did not set it to be blank..
> >
> > I will have to get hold of the distributor next week..
> >
> > Later..
> >
> > >On Sat, 17 Apr 2004, WipeOut wrote:
> > >
> > >
> > >
> > >>Hi,
> > >>
> > >>I have a Snom 200 that has had admin mode switched off and I have no
> > >>idea when the admin password has been set to.. Does anyone know of a
> way
> > >>to reset the phone to factory defaults??
> > >>
> > >>Later..
> > >>___
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> > >>
> > >>
> > >
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> > >
> > >
> >
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Re: [Asterisk-Users] VOIP Spam

[Duane]

Nicholas Bachmann wrote:

A web of trust is different from the chain of trust I'm talking about.  
In a web of trust, a key is signed by lots of different people; ideally, 
everybody can trust everybody.  In a chain of trust, each member only 
knows and trusts the adjacent members.
CAcert doesn't operate a web of trust in the PGP sense, for someone to 
issue "trust" points to other people they must already have a certain 
amount of trust points themselves. Both PKI and PGP models will fail, 
not because of the technology but because of the people factor. The PKI 
model *can* be to a larger is a slightly more resilient, in general no 
CA would have reason to issue false certificates and *usually* you can 
be sure more are issued on a correct basis. PGP model if you lived in 
say Africa and wanted to communicate with someone in South America with 
little or no prior relationship and you wanted to be sure the 
communication wouldn't be intercepted you have 2 choices, fly to meet 
each other or gain trust you both are who you say you are from an 
impartial 3rd party that if it did it's job correct would be correct.

*BUT*, and it's a very big but, there is 2 or 3 flaws in the PKI model, 
firstly there is a crap load of money usually involved, where there is 
money there is usually corruption, at this stage of the game the PKI 
industry has had very little over all impact, something like 0.3% of web 
servers (not websites) are protected with a "valid" certificate issued 
by a "valid" CA, the number of invalid and self signed and non-"valid" 
signed certificates is closer to 1.3%. There are a lot of websites that 
should use some form of crypto to protect against passive listening. 
Another major flaw is PKI based on issued certificates from any CA would 
be worthless in protecting a person in the country where governments 
repress free speech by arresting and killing their citizens. In the UK I 
believe the government has laws in place so they can demand your private 
key, and the US could coerce by legal means to force CAs to issue false 
certificates and then stick a gag order of them.

PGP model would obviously be an advantage in this case, but most people 
don't have a clue about security practises and get so many pop-up 
warning messages they simply click ok to whatever comes up.

The other flaw is safe keeping of certificates, unless you have a 
hardware device, the more difficult you make it for someone to break 
digital security will only make them turn round and break physical 
security...

Passwords are inherently bad and there are numerous articles on people 
giving their work/email passwords away for a cheap pen...

Sort of... CAcert.org is a Certificate Authority.  A CA just signs 
public keys, while a key server stores a copy of them.  What I'm talking 
about is more like http://pgp.mit.edu/.
Working on it, we actually have a finger daemon setup/running to reply 
with certificates if you send it a exact request that matches an entry 
in the database, weather hostname or email address...

I've penned an internet-draft on what we've done which can be read here:

http://www.cacert.org/index.php?id=26&prob=8

I keep meaning to post it to the IETF as a informational document...

But we're not looking at certificates; we're looking at public/private 
keypairs.  Phones can generated the keypairs, but how does the phone 
prove to the key server that it is an authorized phone?  With just a 
simple password?
The PIX sends a certificate signing request and holds onto the private 
key, the CA then replies with a signed certificate and the PIX stores 
that with the private key...

When grabbing a certificate it doesn't matter if it's authorised to or 
not, because it has the private key so only it can decode data sent to 
it using the public certificate...

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
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Re: [Asterisk-Users] Different UK Caller ID question!

[Linus Surguy]

> Can a standard BT phone that supports CID (Such as a BT Decor 310) pick up
the
> CID information that asterisk passes out to analog lines or would I have
to
> get an analog phone with CID from the states?

Most of the BT brand caller display phones support both the BT/UK standard
means of transmitting caller ID before ring and the US method of
transmitting it afterwards, so you should be OK with Asterisk.

Linus

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[Asterisk-Users] Different UK Caller ID question!

[Darren Poulson]

Here's a bit of a twist to the common UK Caller ID question... (Which I've got 
working nicely thanks to some slight changes in Jonathan McHarg's scripts off 
the asterisk-dev mailing list, and a Pace modem from ebay!)

Can a standard BT phone that supports CID (Such as a BT Decor 310) pick up the 
CID information that asterisk passes out to analog lines or would I have to 
get an analog phone with CID from the states?

Thanks,

Darren.
-- 
Darren Poulson - Unix Admin
PGP Key at: http://www.22balmoralroad.net/~daz/pgp.key
Don't ever be the first, don't ever be the last and don't ever volunteer to do 
anything
-- Murphy's Military Laws n°26
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Re: [Asterisk-Users] FW: Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)

[tmpm]

I fired off a note to them at both addresses as well, and asked them check 
out astricon as well..links included. (clues to the clueless mode...)

I read their article on Pingtel going OS while sitting on the porcelain 
furniture the other morning, and thought they might be about to discover 
fire and the wheel, the post here reminded me to follow up on mentioning * 
to them...Im sure they have been enlightened by now (thump). Now it remains 
to see if they're going to cover a product that doesn't purchase pricey 
space in their rag.

Marc



At 11:51 4/17/2004, you wrote:
The [EMAIL PROTECTED] appears to be broken.  I dug around the magazine
contacts and found Doug Allen, senior editor, you can send comments to
[EMAIL PROTECTED] .  I didn't get a bounce back from that e-mail so I assume
it made it to the editor.
JR



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[Asterisk-Users] SIP incoming distinctive ring

[Thomas B. Clark]

I now have a virtual number in addition to my main number from 
broadvoice.  Enabling distinctive ring in their portal results in the 
following SIP header being received:

Alert-Info: 

It looks like Asterisk has the ability to generate an outgoing 
Alert-Info header (actually Alert-info, not sure if it makes a 
difference), but does not understand incoming headers.

I cannot find any discussion of such a feature. Would there be any other 
interest in it?

Alternatively, if there were any way to read the headers into variables 
from within Asterisk, I could then use that information to make 
decisions within extensions.conf. Is there some way to do this that I'm 
not thinking of?
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[Asterisk-Users] E100P for Bandwidth Termination

[Azher Amin]

Hi,

I have a query from a client that can he use the E100P card to terminate
the 2Mbps bandwidth in a linux box, thus reducing the cost of cisco
router ?? 

The other end is a cisco 2620 router with E1 VWIC-1MFC.

Can anyone explain if its possible with Asterisk and further any
configuration help. Applreciated.

Regards
Azher Amin
---
http://www.consulttech.com.pk




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Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan

[Linus Surguy]

> Linus,  I assuming that for incoming service something like .
> 
> exten => ,1,AbsoluteTimeout(3600)
> exten => ,2,Dial(SIP/sipdevice,120)  (maybe with ,r)
> exten => ,3,Congestion
> exten => ,103,Busy
> 
> [where  is whatever the incoming service number is
> presented as.]
> 
> would probably not cause someones telco to charge a customer if the call
> wasn't picked up after 120 seconds or was busy immediately as the only
> possible 'answer' would be generated by the dialled device picking up?

Exactly!

Linus

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Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan

[Chris Orme]

Thanks Olle !  

I'm getting on better without the ,r now when making outgoing SIP calls
though not confident I've got rid of any ringing with all devices on test 
yet.  
I put the ,r in irrationally when the iaxy wasn't ringing out not that I
think it necessarily helped anyway - taking it out things are starting to
slowly improve with intelligent devices.

I'm pretty sure it does cause ringing from the start but isn't it called
'ringback' so there might be more to it?

Experimentation is doing me proud today :-)  Maybe it always does.

Once I get good results maybe I should post what I ended up with in case
it helps anyone else out. :-) as the original post info is nowhere close
to what is needed.

Linus,  I assuming that for incoming service something like .

exten => ,1,AbsoluteTimeout(3600)
exten => ,2,Dial(SIP/sipdevice,120)  (maybe with ,r)
exten => ,3,Congestion
exten => ,103,Busy

[where  is whatever the incoming service number is
presented as.]

would probably not cause someones telco to charge a customer if the call
wasn't picked up after 120 seconds or was busy immediately as the only
possible 'answer' would be generated by the dialled device picking up?

Thanks!

Chris

On Sat, 17 Apr 2004, Olle E. Johansson wrote:

> Chris Orme wrote:
> 
> >>>exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
> 
> Isn't the 'r' forcing a 'ringing' signal from start, regardless
> of what the device you are calling are signalling. If you are calling
> a SIP device, that device might return 'busy' and that's propably
> why you first hear 'ringing' and then a 'busy' signal.
> 
> I would like app_dial gurus to explain the 'r' option a bit
> more so we can document it better.
> 
> /O
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Re: [Asterisk-Users] VOIP Spam

[Nicholas Bachmann]

Duane wrote:

Nicholas Bachmann wrote:

1. It's a chain of trust: it's hard for Bob to verify Alice's 
signature directly
   -Not impossible to fix


CAcert.org's whole purpose is cheap, easily obtainable security... It 
employs a web of trust in the website frame work to build up and 
distribute face to face identification checks...
A web of trust is different from the chain of trust I'm talking about.  
In a web of trust, a key is signed by lots of different people; ideally, 
everybody can trust everybody.  In a chain of trust, each member only 
knows and trusts the adjacent members.


2. A central registry must be created that's free and open for 
providers to use but secure enough to verify members.


Again CAcert.org fulfils this criteria...
Sort of... CAcert.org is a Certificate Authority.  A CA just signs 
public keys, while a key server stores a copy of them.  What I'm talking 
about is more like http://pgp.mit.edu/.

   -Think about the global IP address distribution agencies
3. Phones must get private keys securely.


Last one is as much a technical issue as a people issue, although PIX 
firewalls implement (forget the acronym) where they send a request to 
a CA and the CA sends back a certificate, I keep meaning to implement 
it for CAcert but I lack a PIX for dev & testing...
But we're not looking at certificates; we're looking at public/private 
keypairs.  Phones can generated the keypairs, but how does the phone 
prove to the key server that it is an authorized phone?  With just a 
simple password?

Nick

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Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan

[Olle E. Johansson]

Chris Orme wrote:

exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
Isn't the 'r' forcing a 'ringing' signal from start, regardless
of what the device you are calling are signalling. If you are calling
a SIP device, that device might return 'busy' and that's propably
why you first hear 'ringing' and then a 'busy' signal.
I would like app_dial gurus to explain the 'r' option a bit
more so we can document it better.
/O
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Re: [Asterisk-Users] Snom 200 Admin Password

[WipeOut]

Pertti Pikkarainen wrote:

There is a way.
Right after reboot, and when you see the first text,  hit any key
and you will get a 'boot menu'.  Give the phone an ip-address and 
define a tftp-server.
The bootfile must be named snom200.bin ( e.g rename the latest snom sw ).

After you have succesfully got it to download the code,
the phone is also resetted to factory defaults.   You will see erasing 
flash etc.
If the download fails the phone will use the sw it has got and there 
will be no change
in the config either.

--Pertti


Thanks Dude!! This worked so I have my phone back.. :)

Later..
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Re: [Asterisk-Users] Snom 200 Admin Password

[Ryan Thrash]

I *think* the default password is "" (all zeros).

HTH,
Ryan
On Apr 17, 2004, at 10:38 AM, WipeOut wrote:

Pertti Pikkarainen wrote:

There is a way.
Right after reboot, and when you see the first text,  hit any key
and you will get a 'boot menu'.  Give the phone an ip-address and 
define a tftp-server.
The bootfile must be named snom200.bin ( e.g rename the latest snom 
sw ).

After you have succesfully got it to download the code,
the phone is also resetted to factory defaults.   You will see 
erasing flash etc.
If the download fails the phone will use the sw it has got and there 
will be no change
in the config either.

--Pertti

Hmm.. That would mean I would have to setup a TFTP server which is a 
hassle.. :) I was hoping that there was some key combination or a 
reset busson hidden somewhere..


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Re: [Asterisk-Users] VOIP Spam

[Duane]

Nicholas Bachmann wrote:

1. It's a chain of trust: it's hard for Bob to verify Alice's signature 
directly
   -Not impossible to fix
CAcert.org's whole purpose is cheap, easily obtainable security... It 
employs a web of trust in the website frame work to build up and 
distribute face to face identification checks...

2. A central registry must be created that's free and open for providers 
to use but secure enough to verify members.
Again CAcert.org fulfils this criteria...

   -Think about the global IP address distribution agencies
3. Phones must get private keys securely.
Last one is as much a technical issue as a people issue, although PIX 
firewalls implement (forget the acronym) where they send a request to a 
CA and the CA sends back a certificate, I keep meaning to implement it 
for CAcert but I lack a PIX for dev & testing...

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
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Re: [Asterisk-Users] asterisk database support

[gaillac harry]

Sorry for echo I just wait for a reply :)

I looked at Voip-info but does a GUI is provided to insert datas in
tables ???


Le sam 17/04/2004 à 17:36, Brancaleoni Matteo a écrit :
> I hear some echo there  :)
> 
> simply, you can define sip friends from a database.
> just create the table, enable SIP_FRIENDS into channels
> Makefile and read chan_sip.c how to set
> db access (db access data must be into sip.conf)
> 
> but, firstofall, you must be familiar with sip.conf
> and friends/user/peer definition in order to understand
> how it works...
> 
> matteo
> 
> Il sab, 2004-04-17 alle 17:21, gaillac harry ha scritto:
> > Hello,
> > 
> > Is it possible to use a database for provisionning sip clients?
> > 
> > CVS provides sip-friends.sql in order to create tables (not database)
> > what may i do with that tables?
> > 
> > Regards
> > 
> > Harry
> > 
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Re: [Asterisk-Users] VOIP Spam

[Nicholas Bachmann]

Duane wrote:

Tom Green wrote:

Brian,

Encrypted SIP messages can be sent using TLS. However,
I don't think it is realistic to expect everyone
calling you to have a public/private key pair.

I don't quite agree.

SMTP servers that support SMTP-TLS and have valid certs + config do 
exactly that already...
But I think Tom's point is that SMTP-TLS is not very common.

However, a PKI for VoIP would be much easier, and much more manageable, 
than PKI for email.  Each provider would have to maintain a key server 
that stored keys for their users.  Then, a public, central registry of 
provider keys would be needed.  The main challenge would be getting 
private keys into phones.

Alice ---> Alice's Provider (AP Co.) -> 
Bob's Provider (BP Co.) > Bob
 [Signed by Alice]   [Alice's 
Verified Sig][Alice's Verified 
Sig]   

[Signed by AP Co.]  [AP Co.'s Verified Sig]

 [Signed by BP Co.]

In this system, Alice would sign and send her SIP messages to her 
provider's  SIP proxy.  Her provider, AP Co., proxy would verify the 
signature with its own key server, and, if valid, would sign it with the 
AP Co, key and pass it on to BP Co.'s proxy server.  The BP Co. proxy 
could then check AP Co.'s signature, sign the message, and pass it to 
Bob.  Bob, then, must only check that the message is signed by the 
user's provider.

There are, of course, weaknesses in this plan.  To name a few:
1. It's a chain of trust: it's hard for Bob to verify Alice's signature 
directly
   -Not impossible to fix
2. A central registry must be created that's free and open for providers 
to use but secure enough to verify members.
   -Think about the global IP address distribution agencies
3. Phones must get private keys securely.

Nick

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Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan

[Linus Surguy]

> > > exten => _00.,1,AbsoluteTimeout(3600)
> > > exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
> > > exten => _00.,3,Answer
> > > exten => _00.,4,Hangup
> >
> > I can't help with presenting busy to the SIP devices, but if you have
the
> > above on any sort of PSTN gateway you are going to annoy the PSTN
users - as
> > if the number selected is busy or otherwise unavailable you will still
> > 'Answer' the PSTN call, causing the person calling to pay whatever call
> > establishment charges/minimum charges appropriate to their tariff.

> Thanks for pointing that out.
>
> Luckily Asterisk has a 'billed seconds' field in the cdr which is 0 when a
> number is unavailable or busy despite showing the call as 'answered'.
>
> A view could be taken that 0 length billed seconds calls need not be
> billed with a minimum connection charge... perhaps.
> Not ideal though.

Thats not quite the point, I was saying that if instead of this being SIP ->
Asterisk, this was PSTN -> Asterisk then you would have cost the caller real
money, as their telco, BT, AT&T or whoever would have charged the caller as
a result of your 'Answer'. Asterisk's own billing records are not relevant
to this.

Linus

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Re: [Asterisk-Users] Snom 200 Admin Password

[Chris Orme]

If there is one I'd love to know about it feel free to let me know as I
use snoms a lot.

The only other ways I know that might be possible are more complicated
than the tftp method.

Good luck!

On Sat, 17 Apr 2004, WipeOut wrote:

> Pertti Pikkarainen wrote:
> 
> >
> > There is a way.
> > Right after reboot, and when you see the first text,  hit any key
> > and you will get a 'boot menu'.  Give the phone an ip-address and 
> > define a tftp-server.
> > The bootfile must be named snom200.bin ( e.g rename the latest snom sw ).
> >
> > After you have succesfully got it to download the code,
> > the phone is also resetted to factory defaults.   You will see erasing 
> > flash etc.
> > If the download fails the phone will use the sw it has got and there 
> > will be no change
> > in the config either.
> >
> > --Pertti
> >
> Hmm.. That would mean I would have to setup a TFTP server which is a 
> hassle.. :) I was hoping that there was some key combination or a reset 
> busson hidden somewhere..
> 
> Later..
> 
> >
> > Chris Orme wrote:
> >
> >> Hi.
> >>
> >> Did you buy the phone or get it second hand ?   If second hand do you 
> >> have
> >> any paperwork from the person you bought it from and did they buy it
> >> through official distribution?
> >>
> >> If you got it through distribution I would am fairly sure your vendor
> >> might be able to help ?
> >>
> >> I have a rough idea of how it would be possible but I would think you'll
> >> probably have to prove ownership as this password is how carriers lock
> >> their phones.  If you got it from a carrier I imagine you might possibly
> >> have to pay them an unlock charge so you can change carriers.
> >>
> >> Or did you accidently set the admin password?
> >>
> >>
> >> Chris
> >>
> >> On Sat, 17 Apr 2004, WipeOut wrote:
> >>
> >>  
> >>
> >>> Hi,
> >>>
> >>> I have a Snom 200 that has had admin mode switched off and I have no 
> >>> idea when the admin password has been set to.. Does anyone know of a 
> >>> way to reset the phone to factory defaults??
> >>>
> >>> Later..
> >>> ___
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> >>
> >>
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[Asterisk-Users] FW: Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)

[JR Richardson]

The [EMAIL PROTECTED] appears to be broken.  I dug around the magazine
contacts and found Doug Allen, senior editor, you can send comments to
[EMAIL PROTECTED] .  I didn't get a bounce back from that e-mail so I assume
it made it to the editor.

JR




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Re: [Asterisk-Users] Snom 200 Admin Password

[Chris Orme]

The tftp suggestion you received is well worth trying :-)  I didn't know
that was possible.

Although I wonder if it works as it might mean that carriers deploying the
snom may not be able to properly lock their phones perhaps??

Chris

On Sat, 17 Apr 2004, WipeOut wrote:

> Chris Orme wrote:
> 
> >Hi.
> >
> >Did you buy the phone or get it second hand ?   If second hand do you have
> >any paperwork from the person you bought it from and did they buy it
> >through official distribution?
> >
> >If you got it through distribution I would am fairly sure your vendor
> >might be able to help ?
> >
> >I have a rough idea of how it would be possible but I would think you'll
> >probably have to prove ownership as this password is how carriers lock
> >their phones.  If you got it from a carrier I imagine you might possibly
> >have to pay them an unlock charge so you can change carriers.
> >
> >Or did you accidently set the admin password?
> >
> >
> >Chris
> >
> >  
> >
> I bought the phone new about a year ago so its not provider locked..
> 
> I set the password to be nothing (I think) and then I set admin mode 
> off, then when I tried to get into the admin area I couldn't, it would 
> seem that either there is a bug that doesn't allow a blank password or 
> it did not set it to be blank..
> 
> I will have to get hold of the distributor next week..
> 
> Later..
> 
> >On Sat, 17 Apr 2004, WipeOut wrote:
> >
> >  
> >
> >>Hi,
> >>
> >>I have a Snom 200 that has had admin mode switched off and I have no 
> >>idea when the admin password has been set to.. Does anyone know of a way 
> >>to reset the phone to factory defaults??
> >>
> >>Later..
> >>___
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> >  
> >
> 
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Re: [Asterisk-Users] Snom 200 Admin Password

[Brancaleoni Matteo]

Wipeout,
> I bought the phone new about a year ago so its not provider locked..
> 
> I set the password to be nothing (I think) and then I set admin mode 
> off, then when I tried to get into the admin area I couldn't, it would 
> seem that either there is a bug that doesn't allow a blank password or 
> it did not set it to be blank..
> 
> I will have to get hold of the distributor next week..
> 
> Later..

if I don't remember wrong, the default pw is .
btw, just download the firmware from snom website,
put it onto a tftp server and rename it as snom200.bin

then reboot the phone, and as soon as it powers up
(don't let the phone boot at all), press a key.
it will prompt an ip addr,netmask,gw and tftp server addr.
fill the values and go on.

the phone will load the firmware, and everything
will be set @ default values.

Matteo.

-- 
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl

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Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan

[Chris Orme]

Hi Linus,

Thanks for pointing that out.

Luckily Asterisk has a 'billed seconds' field in the cdr which is 0 when a
number is unavailable or busy despite showing the call as 'answered'.  

A view could be taken that 0 length billed seconds calls need not be
billed with a minimum connection charge... perhaps.
Not ideal though.

I'm not sure what is the alternative as 'Answer' seems needed to get the
SIP clients attention and tell it what it should do.

Perhaps an immediate Answer before Dial worth trying.  
I'll experiment a little and the mileage of what different SIP devices
do does vary a little bit.

Thanks for replying!  Chris

On Sat, 17 Apr 2004, Linus Surguy wrote:

> > My dialplan is for the outgoing SIP call is:
> >
> > exten => _00.,1,AbsoluteTimeout(3600)
> > exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
> > exten => _00.,3,Answer
> > exten => _00.,4,Hangup
> > exten => _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r)
> > exten => _00.,104,Answer
> > exten => _00.,105,Hangup
> >
> 
> I can't help with presenting busy to the SIP devices, but if you have the
> above on any sort of PSTN gateway you are going to annoy the PSTN users - as
> if the number selected is busy or otherwise unavailable you will still
> 'Answer' the PSTN call, causing the person calling to pay whatever call
> establishment charges/minimum charges appropriate to their tariff.
> 
> Linus
> 
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Re: [Asterisk-Users] Snom 200 Admin Password

[WipeOut]

Pertti Pikkarainen wrote:

There is a way.
Right after reboot, and when you see the first text,  hit any key
and you will get a 'boot menu'.  Give the phone an ip-address and 
define a tftp-server.
The bootfile must be named snom200.bin ( e.g rename the latest snom sw ).

After you have succesfully got it to download the code,
the phone is also resetted to factory defaults.   You will see erasing 
flash etc.
If the download fails the phone will use the sw it has got and there 
will be no change
in the config either.

--Pertti

Hmm.. That would mean I would have to setup a TFTP server which is a 
hassle.. :) I was hoping that there was some key combination or a reset 
busson hidden somewhere..

Later..

Chris Orme wrote:

Hi.

Did you buy the phone or get it second hand ?   If second hand do you 
have
any paperwork from the person you bought it from and did they buy it
through official distribution?

If you got it through distribution I would am fairly sure your vendor
might be able to help ?
I have a rough idea of how it would be possible but I would think you'll
probably have to prove ownership as this password is how carriers lock
their phones.  If you got it from a carrier I imagine you might possibly
have to pay them an unlock charge so you can change carriers.
Or did you accidently set the admin password?

Chris

On Sat, 17 Apr 2004, WipeOut wrote:

 

Hi,

I have a Snom 200 that has had admin mode switched off and I have no 
idea when the admin password has been set to.. Does anyone know of a 
way to reset the phone to factory defaults??

Later..
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Re: [Asterisk-Users] no sound when connected

[Chris Orme]

I assume you've got masquerading working so other hosts inside this
network are ok ?

something like...

/usr/local/sbin/iptables -v -t nat -A POSTROUTING -i eth0 -o eth1 -j
MASQUERADE ?

Definitely sounds more network than asterisk.

How about trying to connect to IAXTEL (which uses IAX2 rather than SIP) to
an external box ?  IAX2 protocol not SIP.  IAX2 is easier to get through
NAT and firewalls than SIP as it's brilliantly designed to just use one
UDP port I think (4569).
You perhaps try other sip destinations incase it's something with FWD
specifically.

you'll also need to keep the connection open.

Here's an extract from my sip.conf as to how I connect to FWD.  It may
help.

I've also got allow=alaw allow=ulaw allow=gsm in the general section of my
sip.conf as recommended by voip-info.org

register=user:[EMAIL PROTECTED]/2030


[fwd]
type=friend
accountcode=fwd
disallow=all
allow=alaw
allow=ulaw
allow=gsm
username=XXX
secret=XXX
host=fwd.pulver.com
qualify=1

you may well need nat=yes or other options.  Also are you registering with
fwd ok?

you could also run 'sip debug' and 'sip show peers' 'sip show users' and
asterisk with -d and - options to try and see what is going on.
Ethereal is also a very useful application for debugging you might want
to try?

I would assume if you are reaching the correct extension then dtmf is ok.
You want the RFC /outband if you're using a lossy codec (ie not A/U law)

Maybe you could also trying to connect to FWD directly with the
grandstream ?

Anyone else any pointers on what he could try - someone must have this
setup ??  Good luck!  Have patience and experiment and you'll crack it :)
SIP+NAT do not make wonderful companions but it is possible for sure :)


Chris

On Sat, 17 Apr 2004, Vlok Stone wrote:

> On Sat, 2004-04-17 at 14:43, Chris Orme wrote:
> > Very much sounds like a firewall issue not allowing voice packets back in 
> > to you (for the received audio) or them not finding you somehow.
> > 
> > Think about how do you connect to the internet.  Perhaps 'it' (whatver
> > device it is doing firewalling/NAT) is configurable through its bios or a
> > web interface or by telnet or ssh.  Depends what 'it' is, but 'it' is
> > likely to be involved in the problem.
> > 
> > You didn't send info of your configuration as to which protocol IAX/SIP
> > you are using and how you are trying to connect so I can't give a
> > specific answer on how to help you.  Or I didn't read closely enough.
> SIP is the protocol. 
> > 
> > I guess it might be:
> > 
> > BT 102 -SIP-> Asterisk on local LAN w/PSTN access/zap cards -SIP??->
> > firewall/router -adsl?-> internet -SIP-> Asterisk (2)
> yes that's the basic layout. firewall is linux w/ 2 nics iptables are
> down. I am able to ping out from the asterisk server. So, it is
> forwarding. 
> iptables -L
> Chain INPUT (policy ACCEPT)
> target prot opt source   destination
> 
> Chain FORWARD (policy ACCEPT)
> target prot opt source   destination
> 
> Chain OUTPUT (policy ACCEPT)
> target prot opt source   destination
> 
> still no sound returns from FWD. 
> I'm sure it's the firewall, but can't figure out what's getting denied. 
> 
> > 
> > But I don't know as you didn't say :-(
> > 
> > I know Asterisk went through time when things weren't easy with
> > Grandstream phones, I don't know what the current state of affairs are
> > and I guess it is all great now if it working now via your Zap.
> > 
> > www.voip-info.org or using google to search the archives of this list
> > might also help, especially if you search perhaps for the name of your
> > firewall or router and asterisk or something like one way audio?  This is
> > what I did when I started.
> > 
> > Also if you have available other SIP clients to try on your network and
> > some patience I'm certain this can be tracked down and sorted out.
> > 
> > It might even be something as simple as 'nat=yes' 'host=dynamic'. There
> > are lots of sample configs on www.voip-info.org as well as those supplied
> > by Asterisk to work through.  Slowly change options from the sample config
> > and with patience you get the hang of things :-)
> I have nat=yes and host=dynamic
> > 
> > Hope that helps a little.  Just trying to put something back for all those
> > that helped me.
> Thank You. You're help is very much appreciated. I hope I may also be of
> assistance soon to others. 
> 
> > Good luck.
> > 
> > Chris
> > 
> > On Sat, 17 Apr 2004, Vlok Stone wrote:
> > 
> > > On Sat, 2004-04-17 at 14:01, Chris Orme wrote:
> > > > Hi Vlok,
> > > > 
> > > > When a call connects is the audio one way ?  Can the remote person hear
> > > > you but you can't hear them ?
> > > yes. 
> > > > 
> > > > Which way is the audio or is it silent in both directions ?
> > > > The echo test?  Is this FWDs echo test or the one running on your
> > > > asterisk box (as that is not outside you LAN is it) ?
> > > ouside
> > > > 
> > > > I'm thinking this could be a NA

Re: [Asterisk-Users] Snom 200 Admin Password

[WipeOut]

Chris Orme wrote:

Hi.

Did you buy the phone or get it second hand ?   If second hand do you have
any paperwork from the person you bought it from and did they buy it
through official distribution?
If you got it through distribution I would am fairly sure your vendor
might be able to help ?
I have a rough idea of how it would be possible but I would think you'll
probably have to prove ownership as this password is how carriers lock
their phones.  If you got it from a carrier I imagine you might possibly
have to pay them an unlock charge so you can change carriers.
Or did you accidently set the admin password?

Chris

 

I bought the phone new about a year ago so its not provider locked..

I set the password to be nothing (I think) and then I set admin mode 
off, then when I tried to get into the admin area I couldn't, it would 
seem that either there is a bug that doesn't allow a blank password or 
it did not set it to be blank..

I will have to get hold of the distributor next week..

Later..

On Sat, 17 Apr 2004, WipeOut wrote:

 

Hi,

I have a Snom 200 that has had admin mode switched off and I have no 
idea when the admin password has been set to.. Does anyone know of a way 
to reset the phone to factory defaults??

Later..
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Re: [Asterisk-Users] asterisk database support

[Brancaleoni Matteo]

I hear some echo there  :)

simply, you can define sip friends from a database.
just create the table, enable SIP_FRIENDS into channels
Makefile and read chan_sip.c how to set
db access (db access data must be into sip.conf)

but, firstofall, you must be familiar with sip.conf
and friends/user/peer definition in order to understand
how it works...

matteo

Il sab, 2004-04-17 alle 17:21, gaillac harry ha scritto:
> Hello,
> 
> Is it possible to use a database for provisionning sip clients?
> 
> CVS provides sip-friends.sql in order to create tables (not database)
> what may i do with that tables?
> 
> Regards
> 
> Harry
> 
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-- 
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Espia - Emmegi Srl

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Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan

[Linus Surguy]

> My dialplan is for the outgoing SIP call is:
>
> exten => _00.,1,AbsoluteTimeout(3600)
> exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
> exten => _00.,3,Answer
> exten => _00.,4,Hangup
> exten => _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r)
> exten => _00.,104,Answer
> exten => _00.,105,Hangup
>

I can't help with presenting busy to the SIP devices, but if you have the
above on any sort of PSTN gateway you are going to annoy the PSTN users - as
if the number selected is busy or otherwise unavailable you will still
'Answer' the PSTN call, causing the person calling to pay whatever call
establishment charges/minimum charges appropriate to their tariff.

Linus

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Re: [Asterisk-Users] no sound when connected

[Vlok Stone]

this is my BT100 phone dtmf. Is this correct.Send DTMF:
in-audio via RTP (RFC2833) xvia SIP INFO
i chose sip info 

On Sat, 2004-04-17 at 14:43, Chris Orme wrote:
> Very much sounds like a firewall issue not allowing voice packets back in 
> to you (for the received audio) or them not finding you somehow.
> 
> Think about how do you connect to the internet.  Perhaps 'it' (whatver
> device it is doing firewalling/NAT) is configurable through its bios or a
> web interface or by telnet or ssh.  Depends what 'it' is, but 'it' is
> likely to be involved in the problem.
> 
> You didn't send info of your configuration as to which protocol IAX/SIP
> you are using and how you are trying to connect so I can't give a
> specific answer on how to help you.  Or I didn't read closely enough.
> 
> I guess it might be:
> 
> BT 102 -SIP-> Asterisk on local LAN w/PSTN access/zap cards -SIP??->
> firewall/router -adsl?-> internet -SIP-> Asterisk (2)
> 
> But I don't know as you didn't say :-(
> 
> I know Asterisk went through time when things weren't easy with
> Grandstream phones, I don't know what the current state of affairs are
> and I guess it is all great now if it working now via your Zap.
> 
> www.voip-info.org or using google to search the archives of this list
> might also help, especially if you search perhaps for the name of your
> firewall or router and asterisk or something like one way audio?  This is
> what I did when I started.
> 
> Also if you have available other SIP clients to try on your network and
> some patience I'm certain this can be tracked down and sorted out.
> 
> It might even be something as simple as 'nat=yes' 'host=dynamic'. There
> are lots of sample configs on www.voip-info.org as well as those supplied
> by Asterisk to work through.  Slowly change options from the sample config
> and with patience you get the hang of things :-)
> 
> Hope that helps a little.  Just trying to put something back for all those
> that helped me.
> 
> Good luck.
> 
> Chris
> 
> On Sat, 17 Apr 2004, Vlok Stone wrote:
> 
> > On Sat, 2004-04-17 at 14:01, Chris Orme wrote:
> > > Hi Vlok,
> > > 
> > > When a call connects is the audio one way ?  Can the remote person hear
> > > you but you can't hear them ?
> > yes. 
> > > 
> > > Which way is the audio or is it silent in both directions ?
> > > The echo test?  Is this FWDs echo test or the one running on your
> > > asterisk box (as that is not outside you LAN is it) ?
> > ouside
> > > 
> > > I'm thinking this could be a NAT or firewall issue ?
> > me too. what would i look for. 
> > > 
> > > Maybe you could give more info or a diagram of the set up you have there
> > > so I can have a think about it?
> > > 
> > 
> > > Chris
> > > 
> > > On Sat, 17 Apr 2004, Vlok Stone wrote:
> > > 
> > > > I'm having a sound issue. I'm using BT100 (102). When I dial the echo
> > > > test ( or anything for that matter) outside of my LAN there's no sound
> > > > when it answers although I hear the ringing tones. Is this an RTP or
> > > > codec issue. When I dial through Zap everything is fine. Thanx.
> > > > 
> > > > ___
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[Asterisk-Users] asterisk database support

[gaillac harry]

Hello,

Is it possible to use a database for provisionning sip clients?

CVS provides sip-friends.sql in order to create tables (not database)
what may i do with that tables?

Regards

Harry

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Re: [Asterisk-Users] no sound when connected

[Vlok Stone]

On Sat, 2004-04-17 at 14:43, Chris Orme wrote:
> Very much sounds like a firewall issue not allowing voice packets back in 
> to you (for the received audio) or them not finding you somehow.
> 
> Think about how do you connect to the internet.  Perhaps 'it' (whatver
> device it is doing firewalling/NAT) is configurable through its bios or a
> web interface or by telnet or ssh.  Depends what 'it' is, but 'it' is
> likely to be involved in the problem.
> 
> You didn't send info of your configuration as to which protocol IAX/SIP
> you are using and how you are trying to connect so I can't give a
> specific answer on how to help you.  Or I didn't read closely enough.
SIP is the protocol. 
> 
> I guess it might be:
> 
> BT 102 -SIP-> Asterisk on local LAN w/PSTN access/zap cards -SIP??->
> firewall/router -adsl?-> internet -SIP-> Asterisk (2)
yes that's the basic layout. firewall is linux w/ 2 nics iptables are
down. I am able to ping out from the asterisk server. So, it is
forwarding. 
iptables -L
Chain INPUT (policy ACCEPT)
target prot opt source   destination

Chain FORWARD (policy ACCEPT)
target prot opt source   destination

Chain OUTPUT (policy ACCEPT)
target prot opt source   destination

still no sound returns from FWD. 
I'm sure it's the firewall, but can't figure out what's getting denied. 

> 
> But I don't know as you didn't say :-(
> 
> I know Asterisk went through time when things weren't easy with
> Grandstream phones, I don't know what the current state of affairs are
> and I guess it is all great now if it working now via your Zap.
> 
> www.voip-info.org or using google to search the archives of this list
> might also help, especially if you search perhaps for the name of your
> firewall or router and asterisk or something like one way audio?  This is
> what I did when I started.
> 
> Also if you have available other SIP clients to try on your network and
> some patience I'm certain this can be tracked down and sorted out.
> 
> It might even be something as simple as 'nat=yes' 'host=dynamic'. There
> are lots of sample configs on www.voip-info.org as well as those supplied
> by Asterisk to work through.  Slowly change options from the sample config
> and with patience you get the hang of things :-)
I have nat=yes and host=dynamic
> 
> Hope that helps a little.  Just trying to put something back for all those
> that helped me.
Thank You. You're help is very much appreciated. I hope I may also be of
assistance soon to others. 

> Good luck.
> 
> Chris
> 
> On Sat, 17 Apr 2004, Vlok Stone wrote:
> 
> > On Sat, 2004-04-17 at 14:01, Chris Orme wrote:
> > > Hi Vlok,
> > > 
> > > When a call connects is the audio one way ?  Can the remote person hear
> > > you but you can't hear them ?
> > yes. 
> > > 
> > > Which way is the audio or is it silent in both directions ?
> > > The echo test?  Is this FWDs echo test or the one running on your
> > > asterisk box (as that is not outside you LAN is it) ?
> > ouside
> > > 
> > > I'm thinking this could be a NAT or firewall issue ?
> > me too. what would i look for. 
> > > 
> > > Maybe you could give more info or a diagram of the set up you have there
> > > so I can have a think about it?
> > > 
> > 
> > > Chris
> > > 
> > > On Sat, 17 Apr 2004, Vlok Stone wrote:
> > > 
> > > > I'm having a sound issue. I'm using BT100 (102). When I dial the echo
> > > > test ( or anything for that matter) outside of my LAN there's no sound
> > > > when it answers although I hear the ringing tones. Is this an RTP or
> > > > codec issue. When I dial through Zap everything is fine. Thanx.
> > > > 
> > > > ___
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> > > > 
> > > 
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Re: [Asterisk-Users] no sound when connected

[Chris Orme]

Very much sounds like a firewall issue not allowing voice packets back in 
to you (for the received audio) or them not finding you somehow.

Think about how do you connect to the internet.  Perhaps 'it' (whatver
device it is doing firewalling/NAT) is configurable through its bios or a
web interface or by telnet or ssh.  Depends what 'it' is, but 'it' is
likely to be involved in the problem.

You didn't send info of your configuration as to which protocol IAX/SIP
you are using and how you are trying to connect so I can't give a
specific answer on how to help you.  Or I didn't read closely enough.

I guess it might be:

BT 102 -SIP-> Asterisk on local LAN w/PSTN access/zap cards -SIP??->
firewall/router -adsl?-> internet -SIP-> Asterisk (2)

But I don't know as you didn't say :-(

I know Asterisk went through time when things weren't easy with
Grandstream phones, I don't know what the current state of affairs are
and I guess it is all great now if it working now via your Zap.

www.voip-info.org or using google to search the archives of this list
might also help, especially if you search perhaps for the name of your
firewall or router and asterisk or something like one way audio?  This is
what I did when I started.

Also if you have available other SIP clients to try on your network and
some patience I'm certain this can be tracked down and sorted out.

It might even be something as simple as 'nat=yes' 'host=dynamic'. There
are lots of sample configs on www.voip-info.org as well as those supplied
by Asterisk to work through.  Slowly change options from the sample config
and with patience you get the hang of things :-)

Hope that helps a little.  Just trying to put something back for all those
that helped me.

Good luck.

Chris

On Sat, 17 Apr 2004, Vlok Stone wrote:

> On Sat, 2004-04-17 at 14:01, Chris Orme wrote:
> > Hi Vlok,
> > 
> > When a call connects is the audio one way ?  Can the remote person hear
> > you but you can't hear them ?
> yes. 
> > 
> > Which way is the audio or is it silent in both directions ?
> > The echo test?  Is this FWDs echo test or the one running on your
> > asterisk box (as that is not outside you LAN is it) ?
> ouside
> > 
> > I'm thinking this could be a NAT or firewall issue ?
> me too. what would i look for. 
> > 
> > Maybe you could give more info or a diagram of the set up you have there
> > so I can have a think about it?
> > 
> 
> > Chris
> > 
> > On Sat, 17 Apr 2004, Vlok Stone wrote:
> > 
> > > I'm having a sound issue. I'm using BT100 (102). When I dial the echo
> > > test ( or anything for that matter) outside of my LAN there's no sound
> > > when it answers although I hear the ringing tones. Is this an RTP or
> > > codec issue. When I dial through Zap everything is fine. Thanx.
> > > 
> > > ___
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> > > [EMAIL PROTECTED]
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> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > > 
> > 
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Re: [Asterisk-Users] Snom 200 Admin Password

[Pertti Pikkarainen]

There is a way.
Right after reboot, and when you see the first text,  hit any key
and you will get a 'boot menu'.  Give the phone an ip-address and define 
a tftp-server.
The bootfile must be named snom200.bin ( e.g rename the latest snom sw ).

After you have succesfully got it to download the code,
the phone is also resetted to factory defaults.   You will see erasing 
flash etc.
If the download fails the phone will use the sw it has got and there 
will be no change
in the config either.

--Pertti

Chris Orme wrote:

Hi.

Did you buy the phone or get it second hand ?   If second hand do you have
any paperwork from the person you bought it from and did they buy it
through official distribution?
If you got it through distribution I would am fairly sure your vendor
might be able to help ?
I have a rough idea of how it would be possible but I would think you'll
probably have to prove ownership as this password is how carriers lock
their phones.  If you got it from a carrier I imagine you might possibly
have to pay them an unlock charge so you can change carriers.
Or did you accidently set the admin password?

Chris

On Sat, 17 Apr 2004, WipeOut wrote:

 

Hi,

I have a Snom 200 that has had admin mode switched off and I have no 
idea when the admin password has been set to.. Does anyone know of a way 
to reset the phone to factory defaults??

Later..
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Re: [Asterisk-Users] no sound when connected

[Vlok Stone]

On Sat, 2004-04-17 at 14:01, Chris Orme wrote:
> Hi Vlok,
> 
> When a call connects is the audio one way ?  Can the remote person hear
> you but you can't hear them ?
yes. 
> 
> Which way is the audio or is it silent in both directions ?
> The echo test?  Is this FWDs echo test or the one running on your
> asterisk box (as that is not outside you LAN is it) ?
ouside
> 
> I'm thinking this could be a NAT or firewall issue ?
me too. what would i look for. 
> 
> Maybe you could give more info or a diagram of the set up you have there
> so I can have a think about it?
> 

> Chris
> 
> On Sat, 17 Apr 2004, Vlok Stone wrote:
> 
> > I'm having a sound issue. I'm using BT100 (102). When I dial the echo
> > test ( or anything for that matter) outside of my LAN there's no sound
> > when it answers although I hear the ringing tones. Is this an RTP or
> > codec issue. When I dial through Zap everything is fine. Thanx.
> > 
> > ___
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[Asterisk-Users] [Asterisk-Users]: Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)

[Chris Orme]

Same here.. aliases file looks botched.  Not just the article
potentially broken perhaps ?

Chris

   - The following addresses had permanent fatal errors -
<[EMAIL PROTECTED]>
(reason: 550 5.1.1 <[EMAIL PROTECTED]>... User unknown)

   - Transcript of session follows -
... while talking to keystone.cmp.com.:
>>> DATA
<<< 550 5.1.1 <[EMAIL PROTECTED]>... User unknown
550 5.1.1 <[EMAIL PROTECTED]>... User unknown
<<< 503 5.0.0 Need RCPT (recipient)
451 4.0.0 Cannot open hash database /etc/mail/aliases.db: Invalid
argument

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Re: [Asterisk-Users] Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)

[Jeremy McNamara]

JR Richardson wrote:

Send your comments to [EMAIL PROTECTED]

 



I would but

<[EMAIL PROTECTED]>:
192.155.65.27 does not like recipient.
Remote host said: 550 5.1.1 <[EMAIL PROTECTED]>... User unknown
Giving up on 192.155.65.27.


Jeremy McNamara

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Re: [Asterisk-Users] no sound when connected

[Chris Orme]

Hi Vlok,

When a call connects is the audio one way ?  Can the remote person hear
you but you can't hear them ?

Which way is the audio or is it silent in both directions ?
The echo test?  Is this FWDs echo test or the one running on your
asterisk box (as that is not outside you LAN is it) ?

I'm thinking this could be a NAT or firewall issue ?

Maybe you could give more info or a diagram of the set up you have there
so I can have a think about it?

Chris

On Sat, 17 Apr 2004, Vlok Stone wrote:

> I'm having a sound issue. I'm using BT100 (102). When I dial the echo
> test ( or anything for that matter) outside of my LAN there's no sound
> when it answers although I hear the ringing tones. Is this an RTP or
> codec issue. When I dial through Zap everything is fine. Thanx.
> 
> ___
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[Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan

[Chris Orme]

Hi!

I am having difficultly in having users of various SIP devices obtain the
correct behaviour when they call a busy number ie. only hearing the
Congestion/Busy tone.  I assume this might be because the SIP device
itself generates the 'ring' tone?

With my current setup in the dialplan extract (below) the user of the SIP
device hears one 'ring' and then the busy tone if a number is busy.

I have tried using 'Congestion' (instead of Answer+Hangup) but then their
SIP phone rings indefinitely (or until the 45 secs timeout) and they never
know the number they called was busy and they wait needlessly for 45 secs.

I'm hoping in earnest that someone might be able to post back a quick
change to my dialplan to let me know if I can improve on this behaviour by
changing a few lines in the dialplan or a defining a macro or something or
just if this what happens to everyone with Asterisk perhaps you could let
me know so I can stop searching for a fix ?

I wrote this dialplan last October when I was really new to Asterisk and
just accepted the behaviour until now when I'm wondering if it can be
refined - I've tried and failed and read all that I could.

My dialplan is for the outgoing SIP call is:

exten => _00.,1,AbsoluteTimeout(3600)
exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
exten => _00.,3,Answer
exten => _00.,4,Hangup
exten => _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r)
exten => _00.,104,Answer
exten => _00.,105,Hangup

(if call can go through on TRUNK1 send it out, if TRUNK1 is out of
capacity and therefore busy then try trunk 2 before giving up) if that is
busy (therefore it is likely the number really is busy then grab the
caller and hang them up (and they then hear 'busy').  

(using Congestion instead of Hangup gives the same behaviour here too but
the person calling out still gets the one 'ring' before the busy tone -
also removing the ,r makes no difference either)

Also, if TRUNK2 were busy would it be possible to go to TRUNK3 by
defining context 204 or not ?

Hoping someone has the patience to laugh at what I did and suggest how I
could fix it if it's fixable ?? maybe ? - please ?

Thanks for listening.  Have a great rest of weekend.

Chris

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RE: [Asterisk-Users] no sound when connected

[Todd Lieberman]

Sounds more like a firewall issue.  TL

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Vlok Stone
Sent: Saturday, April 17, 2004 6:03 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] no sound when connected


I'm having a sound issue. I'm using BT100 (102). When I dial the echo
test ( or anything for that matter) outside of my LAN there's no sound
when it answers although I hear the ringing tones. Is this an RTP or
codec issue. When I dial through Zap everything is fine. Thanx.

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[Asterisk-Users] no sound when connected

[Vlok Stone]

I'm having a sound issue. I'm using BT100 (102). When I dial the echo
test ( or anything for that matter) outside of my LAN there's no sound
when it answers although I hear the ringing tones. Is this an RTP or
codec issue. When I dial through Zap everything is fine. Thanx.

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Re: [Asterisk-Users] Snom 200 Admin Password

[Chris Orme]

Hi.

Did you buy the phone or get it second hand ?   If second hand do you have
any paperwork from the person you bought it from and did they buy it
through official distribution?

If you got it through distribution I would am fairly sure your vendor
might be able to help ?

I have a rough idea of how it would be possible but I would think you'll
probably have to prove ownership as this password is how carriers lock
their phones.  If you got it from a carrier I imagine you might possibly
have to pay them an unlock charge so you can change carriers.

Or did you accidently set the admin password?


Chris

On Sat, 17 Apr 2004, WipeOut wrote:

> Hi,
> 
> I have a Snom 200 that has had admin mode switched off and I have no 
> idea when the admin password has been set to.. Does anyone know of a way 
> to reset the phone to factory defaults??
> 
> Later..
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[Asterisk-Users] Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)

[JR Richardson]

* Brethren,

It's a sad day in our community.  Please join me in a moment of silence for
the death of responsible journalism.  Silence.good
enough.

This article goes on to tell about Pingtel's announcement of forming the
"first open source community aimed at creating SIP based servers".

http://www.networkmagazine.com/shared/article/showArticle.jhtml?articleId=18
900066&classroom=

Yipeee!

I am embarrassed and appalled with the lack of recognition or mention of all
the tremendous work in this "already existing community".  I'm writing a
letter to the editor and encourage all of you to do the same.

Send your comments to [EMAIL PROTECTED]

Mark Spencer,

I caught your presentation at the Linux-Kongress
http://graphics.cs.uni-sb.de/VCORE/recordings.html .  I want to personally
thank you for validating all that I have been doing to promote Asterisk to
anyone and everyone who will take the time to listen.  When I talk about it
and show off working systems in action, people get excited and are quite
impressed.  Having your 30 min presentation to go along with my demo
increases my credibility 10 fold.

Thank you all for contributing to this great community.

JR



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[Asterisk-Users] Snom 200 Admin Password

[WipeOut]

Hi,

I have a Snom 200 that has had admin mode switched off and I have no 
idea when the admin password has been set to.. Does anyone know of a way 
to reset the phone to factory defaults??

Later..
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Re: [Asterisk-Users] Problem with x-ten lite

[Alex Brett]

The problem could be to do with the silence surpression feature in X-ten 
Lite.  If you go into Advanced System Settings, Audio Settings, Silence 
Settings, you should have Transmit Silence set to Yes as Asterisk has a 
compatibility issue with the way it tries to surpress transmitting the 
silence (or at least it did when I first used it and I'm assuming it 
still does, if I'm wrong please feel free to correct me...).

Shad Mortazavi wrote:
I prefer the look and feel of the x-ten lite. However, when ever I use 
my x-ten lite I get a lot of breakup in my communication.

 

E.g. I will play some hold music, and every 5-6 seconds I drop some 
packets. I don’t have the same issue with SJPhone.

 

I’m sure this is a configuration issues, but I can work out where?

 

Can someone point me in the right directions?

 

Thanks and Regards

 

Shad Mortazavi

---

**Nexus Technical Manager**

n|m Nexus Management Inc

Neutral Bay

Sydney

 

Hope this helps,
Alex Brett
[EMAIL PROTECTED]
+44 (0)870 744 2170
http://www.loho.co.uk/
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[Asterisk-Users] Problem with x-ten lite

[Shad Mortazavi]

Dear Group,

 

At the moment I use SJPhone as my soft phone with Asterisk. 



I prefer the look and feel of the x-ten lite. However, when
ever I use my x-ten lite I get a lot of breakup in my communication. 

 

E.g. I will play some hold music, and every 5-6 seconds I
drop some packets. I don’t have the same issue with SJPhone.

 

I’m sure this is a configuration issues, but I can
work out where?

 

Can someone point me in the right directions?

 

Thanks and Regards

 

Shad Mortazavi

---

Nexus Technical Manager

n|m Nexus Management Inc 

Neutral Bay

Sydney

 

[Asterisk-Users] asterisk database support

[gaillac harry]

Hello,

Is it possible to use a database for provisionning sip clients?

CVS provides sip-friends.sql in order to create tables (not database)
what may i do with that tables?

Regards

Harry

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