Aditya Kumar | 1 May 02:56 2010
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Re: No change in payload. (SDP)

Thanks a lot Kevin for the reply


From: Kevin P. Fleming <kpfleming <at> digium.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users <at> lists.digium.com>
Sent: Thu, April 29, 2010 5:43:15 AM
Subjec t: Re: [asterisk-users] No change in payload. (SDP)

Aditya Kumar wrote:
> re-posting the question.
> -----------
> use case:
> when some one in my pbx calls 100.200, I have translations well defined,
> Media also (media via asterisk)  --Works.
> when some one calls bob, or for any names I am adding Domain and call is
> been sent to the other party  -- Works, no media...
>
> For the cases when it is talking to the external work,
> I want Astersik not to do anything with the SDP/payload.
> I want it to send as it is to the external proxy.
>
> How can I achieve this? so that the SDP/payload will not be modified for
> users talking to the external world.
> I want media for those external devices to come Directly  to the users
> in m y pbx. (with out going t asterisk)
>
> 2) also related question is can I have the xml payload in the originator
> and call is routed via PBX to the Target.
> The xml payload also must be carried to the target.
> is it possible....
>
> This will really help me as I was held up with this :(

Neither of these are possible; Asterisk is a B2BUA, not a proxy, and as
such the outgoing INVITE is a *different* session from the incoming one.
That means that Asterisk has to be able to understand the SDP content
that arrives so it can forward media between the two sessions.

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming <at> digium.com
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Rudi Oosthuizen | 1 May 09:38 2010
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B400P card crashes conncection

Had a similar problem with a B410p BRI card. Had to enable (or disable) the 100ohms termination jumper on the card, because the telco provider uses different exchanges ( Maybe someone can give more clarity on this?)

Maybe it’s the same problem, enabling resolved my issue.

Grep chan_dahdi /var/log/asterisk/full | grep –i “pri got event” should also give indication off errors on the line from the telco side, if cleared should be no errors.

 

Rudi Oosthuizen

 

 

Ø  Hi,

 

Ø  I have a B400P BRI card with point-to-point connection (signalling:

 

Ø  bri_cpe) with this dmesg: http://pastebin.com/sXrRt1yM

 

Ø  When i restart asterisk server, the card cannot connect to the telco, the control led flashes red. If I unplug the cable between the ISDN nt and the card and wait 40 sec, the card can connect and works properly.

 

Ø  The telco says the asterisk crashes the connection with the telco, when I let the NT reconnect, the card connects properly.

 

 

Ø  Do you have any idea how to solve this problem? Thanks for any help in advance.

 

 

Ø  Best regards,

Ø  Peter Gelencser

 

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--[ UxBoD ]-- | 1 May 16:49 2010
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Re: Being attacked by an Amazon EC2 ...

----- Original Message -----
> Randy-
> 
> > On Wed, Apr 21, 2010 at 5:33 PM, Steve Murphy <murf <at> parsetree.com>
> > wrote:
> >> Assuming that every such spamming/hacking/attack site is funded on
> >> a stolen identity/CC number, it will soon sink into Amazon that
> >> they are
> >> getting a bad rep, and losing money on such problems, as all such
> >> charges are reversed when the identity theft is discovered... How
> >> they overcome
> >> the problem, should be a tribute to the marvelous power of human
> >> ingenuity.
> >
> > Interesting point about the stolen CC numbers. If that is true, then
> > they will be forced to investigate for their own internal damage
> > control.
> 
> You are nothing if not persistent, an excellent quality in a case like
> this. By now I'm sure Amazon execs are
> wondering who is this Randulo guy, hehe.
> 

Slammed again last night by a A-WS server; see if anything comes back from their abuse department!

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Randy R | 1 May 19:48 2010
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Re: Being attacked by an Amazon EC2 ...

On Sat, May 1, 2010 at 4:49 PM, --[ UxBoD ]-- <uxbod <at> splatnix.net> wrote:
> Slammed again last night by a A-WS server; see if anything comes back from their abuse > department!

FWIW, I chose another provider for our most recent customer who needed
cloud hosting, only because of the EC2 flood Attacks and Amazon's weak
defense and lack of cooperation. All they have done so far is PR spin.
We need them to actually do something. In the meantime, they've lost
my business and I hope others are voting with their feet.

I also had an interesting discussion with one of the people behind
http://projecthoneypot.org who said they'd be interested in working
with us on devising a lookup scheme like the one they've been using
for comment spammers, etc. I can tell you from first hand experience
that their DNSBL has saved me hours and avoids 95% of the comment spam
we were getting before I wrote a simple function to access PHP's
database.

As soon as I return from China, I will get back in touch with them and
we should set up a meeting with everyone who is concerned by this EC2
abuse thing. I think we can do some good work together;

An interesting sidenote to Projet HoneyPot is that the site is down
because of a disk failure. But the interesting note is that it is a
SSD! So much for no moving parts being more reliable!

/r

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James Lamanna | 1 May 20:59 2010
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Re: ATA shootout: PAP2T versus Grandstream Handytone 286

>> It seems that the PAP2T does support TFTP and an XML-based config for
>> deployments...
>>

I've used both the Grandstream 286 and the Linksys PAP2T.
I have been able to get some limited faxing to work using T30 with a PAP2T.
Configuration and provisioning of the Linksys is very easy through
either the web GUI
or XML configuration files, which can be transferred through TFTP or HTTP.

I can only hope that Cisco will update the firmware of the PAP2T to
support T38 one day...

-- James

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SIP | 2 May 04:46 2010

Re: AGI <==> DeadAGI

On 4/30/2010 6:03 PM, Luki wrote:
>> It is irrelevant who hangs up, you want to just use DeadAGI in the h
>> extension
>>      
> I wish that would be the case, but at least on 1.4 I see:
>
> [Apr 30 14:59:38]     -- Executing [h <at> master-route:1] DeadAGI(...) in new stack
> [Apr 30 14:59:38] WARNING[27845]: res_agi.c:2160 deadagi_exec: Running
> DeadAGI on a live channel will cause problems, please use AGI
>
> The good news is, we run tens of thousands of calls every day through
> this box and about half of them spit out this warning, but it never
> caused any problems for over a year. Thus this warning is probably
> safe to ignore.
>
> Luki
>
>    
Agreed.  We run DeadAGI for a considerable number of calls since it has 
the ability to run post-hangup cleanup no matter which side hangs up 
(unlike AGI). We see this warning constantly, and ignore it... constantly.

N.

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Jonas Kellens | 2 May 10:54 2010
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Re: Portech MV-374 does not register behind NAT

Jared,

the Portech SIMbox is registering to a DNS name. The firewall is off and the NAT is a Zyxel NBG-419 router.

No mather what port I set, It is not working :

<--- SIP read from my_public_ip:5070 --->
REGISTER sip:sip.sipserver.tld SIP/2.0
Via: SIP/2.0/UDP 192.168.1.25:5070;branch=z9hG4bK4465928ede
From: "SIM 1-1" <sip:simsim1 <at> sip.sipserver.tld>;tag=6672a4c3
To: "SIM 1-1" <sip:simsim1 <at> sip.sipserver.tld>
Call-ID: 277210db31b0ae184675c3245d5e949f <at> 192.168.1.25
Contact: <sip:simsim1 <at> 192.168.1.25:5070>
CSeq: 56 REGISTER
Max-Forwards: 70
Expires: 60
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
User-Agent: Mv-37x (904290)
Content-Length: 0


<------------->
[May  2 10:38:51] --- (12 headers 0 lines) ---
[May  2 10:38:51] Using latest REGISTER request as basis request
[May  2 10:38:51] Sending to 192.168.1.25 : 5070 (no NAT)
[May  2 10:38:51]
<--- Transmitting (NAT) to my_public_ip:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.25:5070;branch=z9hG4bK4465928ede;received=my_public_ip
From: "SIM 1-1" <sip:simsim1 <at> sip.sipserver.tld>;tag=6672a4c3
To: "SIM 1-1" <sip:simsim1 <at> sip.sipserver.tld>
Call-ID: 277210db31b0ae184675c3245d5e949f <at> 192.168.1.25
CSeq: 56 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
[May  2 10:38:51]
<--- Transmitting (NAT) to my_public_ip:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.25:5070;branch=z9hG4bK4465928ede;received=my_public_ip
From: "SIM 1-1" <sip:simsim1 <at> sip.sipserver.tld>;tag=6672a4c3
To: "SIM 1-1" <sip:simsim1 <at> sip.sipserver.tld>;tag=as12309a5f
Call-ID: 277210db31b0ae184675c3245d5e949f <at> 192.168.1.25
CSeq: 56 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="103001vc", nonce="7c2508cc"
Content-Length: 0

Is it normal that there is a Via-header sent from the SIMbox with its local IP-address in it ??
Is it normal that "SIP read from my_public_ip:5070" has the same port number as the SIP-account (simsim1) 192.168.1.25:5070 ??
Could it be that NAT is not working correctly in my router ??


Jonas.


On 04/22/2010 05:53 PM, Jared Smith wrote:
Is the device registering to an IP address, or do a DNS name? What type of NAT firewall are you using? This reminds me of a problem I had years ago with a Cisco PIX firewall, where it would rewrite IP addresses in the SIP Request URI, causing the authentication to fail. One solution was to have it register to a fully-qualified domain name instead of an IP address, so that the Request URI wouldn't get overwritten. It's certainly worth a shot... -- Jared Smith Digium, Inc.
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Dan Journo | 2 May 14:34 2010

Re: Calls Dropping

> my advise check your internet connection on the remote location and keep a ping from that network to your server running all the time to check for time outs.

 

How can i log a continuous ping test to a file and include the date and time of each ping?

I’ve found this bash code but it only logs once the tests have all finished. If I set it to continuous and then kill the task when I want to view the pings, it doesn’t record the data.

 

#!/bin/sh

NOW=$(date +"%T %m/%d/%Y")

PING=$(ping -qc 5 example.com | grep '5 packets')

echo $NOW: $PING >> /home/matt/ping.log

exit 0

 

Thanks

Dan

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Bob Smither | 2 May 15:02 2010

Re: Calls Dropping


On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote:

<snip>

> How can i log a continuous ping test to a file and include the date
> and time of each ping?

Try this:

#!/bin/sh
for (( ; ; ))
do
  NOW=$(date +"%T %m/%d/%Y")
  PING=$(ping -qc 1 example.com)
  echo $NOW: $PING >> pinger.log
done
exit 0

You can then monitor the log file using:

$ tail -f pinger.log

You will need to use ^C to kill the script.

Hope this helps.

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Steve Underwood | 2 May 15:19 2010

Re: ATA shootout: PAP2T versus Grandstream Handytone 286

On 05/02/2010 02:59 AM, James Lamanna wrote:
>>> It seems that the PAP2T does support TFTP and an XML-based config for
>>> deployments...
>>>
>>>        
> I've used both the Grandstream 286 and the Linksys PAP2T.
> I have been able to get some limited faxing to work using T30 with a PAP2T.
> Configuration and provisioning of the Linksys is very easy through
> either the web GUI
> or XML configuration files, which can be transferred through TFTP or HTTP.
>
> I can only hope that Cisco will update the firmware of the PAP2T to
> support T38 one day...
>    
They've made it pretty clear that isn't going to happen. I suspect its a 
lack of resources - memory and processing power - rather than being 
bloody minded and trying to maintain some differentiation for the 
SPA2102. They can only do one port of G.729, and I assume that is more 
than just saving on per port patent licence fees.

Steve

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Gmane