Re: [asterisk-users] Block Specific Number on Inbound

2011-12-29 Thread Stuart Sheldon
Check out the X Boy/Girl friend feature.

http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf

Around the middle of the page.

Stu


-Original Message-
From: Kevin Oravits korav...@rcolegal.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Subject: [asterisk-users] Block Specific Number on Inbound
Date: Fri, 30 Dec 2011 01:39:46 +

Greetings,

 

Is there a way to block a specific inbound number? I’ve found code
online for blocking all nocallerid and all 800, etc. but nothing for a
specific number. My company is wanting me to block a specific number. Is
this possible in Asterisk 1.4 and 1.6 or do I need to go through my
Service Provider?

 

Thanks,

 

Kevin Oravits 

Phone Sys Admin

 


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Re: [asterisk-users] FW: Under heavy attack

2010-11-01 Thread Stuart Sheldon
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Here's my take on the attack... Sigh...


http://www.stuartsheldon.org/blog/2010/11/sip-brute-force-attacks-escalate-over-halloween-weekend/

Stu

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Re: [asterisk-users] Under heavy attack

2010-10-30 Thread Stuart Sheldon
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We are also seeing an increase in attacks. And yes, there is a benefit
to blocking them. They tend to go away if you have them restricted,
where if you let them go at it, they will sit on your host for sometimes
hours.

Stu


On 10/30/2010 12:43 PM, Joel Maslak wrote:
 Is there really any benefit to blocking these, if you use good passwords?
 
 On Sat, Oct 30, 2010 at 1:20 PM, Warren Selby wcse...@selbytech.com
 mailto:wcse...@selbytech.com wrote:
 
 I'm experiencing this on one of my clients servers. The attack is
 ongoing. 
 
 Thanks,
 --Warren Selby
 
 On Oct 30, 2010, at 2:28 PM, Zeeshan Zakaria zisha...@gmail.com
 mailto:zisha...@gmail.com wrote:
 
 My main asterisk server is under unusual heavy attack, and so far
 Fail2Ban has blocked about 30 IPs, from various different
 countries. At this time it is blocking about 1 IP address every
 few minutes.

 Just wondering if anybody else is also experiencing unusually
 increased hack attempts today?

 Zeeshan A Zakaria

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Re: [asterisk-users] Under heavy attack

2010-10-30 Thread Stuart Sheldon
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On 10/30/2010 08:25 PM, Warren Selby wrote:
 To me it seems the real question is What is going on today?. I
 normally get eight to ten asterisk-related fail2ban alerts a day
 between a few client sites - today I've received at least 10 times
 that many attacks on just one site. These are all coming in from
 different ip addresses, a new one every few minutes. These addresses
 are located all across the globe. This seems like some kind of
 coordinated assault - maybe someone is activating a 'bot-net' for sip
 attacks?

We are seeing the same thing... It could be a bot-net, but it is a very
poorly organized attack. If is was a single bot-net, you would assume
that the systems would each pick a group of addresses, not all attack
the same addresses.

It could be an attempt to get a large number of systems blacklisted. If
someone was to spoof 1000s of addresses that cause operators to
black-list those addresses, they could knock quite a few systems off the
map. This could cause legitimate operators to get blocked, or, discredit
the current method used to detect and block SIP brute force attacks.

Just my two cents...

Stuart Sheldon
ACT USA


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Re: [asterisk-users] Being attacked by an Amazon EC2 ...

2010-04-21 Thread Stuart Sheldon
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Randy R wrote:
 On Wed, Apr 21, 2010 at 2:55 PM, Fred Posner f...@teamforrest.com
 wrote:
 On Apr 21, 2010, at 4:50 AM, Gordon Henderson wrote:
 
 On Tue, 20 Apr 2010, Frank Bulk wrote:
 
 Please take note of their posting: 
 https://aws.amazon.com/security/ which discusses the issue and
 what they're doing to improve response.
 And is anyone on the list worthy of being considered a
 significant SIP provider to be honoured with the privilege of
 working with them?
 
 Gordon
 
 None of the carriers I deal with have been contacted. Of course,
 them only contacting significant providers... does that mean it's
 ok if the attacks happen to non-significant providers or
 end-points?
 
 ---fred http://qxork.com
 
 If it got to their BS/PR page/blog it means they're hearing about 
 complaints on the net as well as people like you submitting. Everyone
  please keep posting where you can and sooner or later, someone big 
 will pick up the story.
 
 Funny, I'd think the most worthy people to comment on this issue
 are on this list. That's the feedback they should be looking for and 
 working on at Amazon EC2.
 
 /r
 

We might me reading their PR wrong... Maybe there were large SIP
providers that were compromised due to this attack... Maybe they are
keeping that quiet at the request of those providers... It could also be
that the aliens in hiding in Colorado are behind the whole thing! ... Oh
no! I've said too much!!! LOL...

It could actually be the case that this whole issue went beyond what we
are seeing, and they are trying to protect one of their Whale customers...

Needless to say, what about the SSH brute force attacks that originate
from their network? What about the SPAM that flows like a fountain from
their net blocks?

This was nothing more then PR hype...

Stu


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Re: [asterisk-users] Amazon EC2 SIP floods - you can help

2010-04-18 Thread Stuart Sheldon
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Randy R wrote:
 On Sun, Apr 18, 2010 at 1:56 PM, Tzafrir Cohen tzafrir.co...@xorcom.com 
 wrote:
 It seems that at least Slashdot is responsive:

 http://it.slashdot.org/story/10/04/17/2059256/SIP-Attacks-From-Amazon-EC2-Going-Unaddressed
 
 Yes, there's a lot of talk here, some of it sympathetic, some less so,
 but at least there's discussion. I expect a response from Amazon at
 some point, but not until the visibility level becomes painful. One
 other person in the Slashdot posts threatens to stop buying things
 from Amazon. More of those will likely surface.
 
 I a related question, if the IP addresses were spoofed, how could a
 response be directed back? Don't the register attempts, because they
 need a response necessarily carry the correct source IP?
 
 /r
 

Yes,

If the IP addresses were spoofed, it would be simply a DoS attack.

Stu

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Re: [asterisk-users] Amazon EC2 SIP floods - you can help

2010-04-18 Thread Stuart Sheldon
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Randy R wrote:
 On Sun, Apr 18, 2010 at 5:38 PM, Stuart Sheldon s...@actusa.net wrote:
 I a related question, if the IP addresses were spoofed, how could a
 response be directed back? Don't the register attempts, because they
 
 If the IP addresses were spoofed, it would be simply a DoS attack.
 
 This is what I thought, so when people say yeah, but they could be
 spoofed this isn't a valid argument.
 
 A huge number of requests going to your server with an originating EC2
 IP needs to be shut down first, questions asked after.
 
 Only Amazon can fix this. They have not only the IP info but also full
 customer data, including banking info.
 
 What possible excuse can they provide? Is this why they are silent?
 There's no good excuse other than, it would cut into our profits.
 
 Maybe we could get GigaOm interested or some other high-visibility blog.
 
 /r
 

For what it's worth, here is my Blog Article from the incident...

http://www.stuartsheldon.org/blog/2010/04/sip-brute-force-attack-originating-from-amazon-ec2-hosts/

Stu

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Re: [asterisk-users] Being attacked by an Amazon EC2 ...

2010-04-11 Thread Stuart Sheldon
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We reported abuse Saturday morning... As of yet, no change in traffic.

I have sent requests upstream to filter all UDP/5060 traffic from EC-2
range to stop the DDOS that we are under, but have only gotten 2 of our
4 providers to comply.

At this point, I guess well all just ride it out...

Stu


Tom Stordy-Allison wrote:
 Hi,
 
 This is exactly what I've just joined this mailing list about.
 
 Has anyone has any luck getting Amazon to stop the instances? I'm stuck with 
 around 700Kbps of my 2.5Mbps inbound in use as my firewall blocks the 
 requests as below. 
 
 Cheers,
 
 Tom
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norbert Zawodsky
 Sent: 11 April 2010 20:57
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Being attacked by an Amazon EC2 ...
 
 Am 11.04.2010 17:05, schrieb Mark Smith:
 Same this end from 184.73.17.150.
 Use this little piece of iptables magic to block the whole of Amazon's EC2 
 ip-
 range.

 iptables -F
 iptables -A INPUT -m iprange --src-range 216.182.224.0-216.182.239.255 -j 
 DROP
 iptables -A INPUT -m iprange --src-range 72.44.32.0-72.44.63.255 -j DROP
 iptables -A INPUT -m iprange --src-range 67.202.0.0-67.202.63.255 -j DROP
 iptables -A INPUT -m iprange --src-range 75.101.128.0-75.101.255.255 -j DROP
 iptables -A INPUT -m iprange --src-range 174.129.0.0-174.129.255.255 -j DROP
 iptables -A INPUT -m iprange --src-range 204.236.192.0-204.236.255.255 -j 
 DROP
 iptables -A INPUT -m iprange --src-range 184.73.0.0-184.73.255.255 -j DROP
 iptables -A INPUT -m iprange --src-range 216.236.128.0-216.236.191.255 -j 
 DROP
 iptables -A INPUT -m iprange --src-range 184.72.0.0-184.72.63.255 -j DROP
 iptables -A INPUT -m iprange --src-range 79.125.0.0-79.125.127.255 -j DROP
 service iptables save

 This sorts it out in the short-term until Amazon realise their service is 
 being utilised by arseholes.




   
 Hi Mark!
 
 your little iptables magic is a very good idea! Implementation took  1
 minute :-)
 I'll use it until a better idea comes up ... (which I don't expect
 within a short term)
 
 Thank you!
 
 Norbert
 

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Re: [asterisk-users] Enter Value and continue dialplan

2008-10-30 Thread Stuart Sheldon
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Take a look at Read()

Stu


David Klaverstyn wrote:
 Hi,
 
  
 
 What function or application do I use to get people to type digits into
 the phone and store the value into a variable?  The application
 WaitExten is not what I want that will jump to the new extension.
 
  
 
 I want users to enter a number into the phone and store it as a variable
 so I can use it later in the dial plan.
 
  
 
 
 
 
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[asterisk-users] Bad Link on Website...

2008-01-03 Thread Stuart Sheldon
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Hash: SHA1

Not sure where to report this...

http://www.asterisk.org/downloads

Right hand download box, Asterisk 1.4.17 points to 1.4.1

Just a heads up.

Stu

- --
And all I can do is keep on telling you, I want you, I need you,
But there aint no way Im ever gonna love you, Now dont be sad
cause two out of three aint bad...
   -- Meatloaf - Two out of Three aint bad - Lyrics
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[asterisk-users] Help on strange problem...

2007-11-16 Thread Stuart Sheldon
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Hash: SHA1

Hey all,

I'm having problems with calls dropping after 15 - 20 seconds from a
particular provider. The are using a NexTone gateway. Here are the details:

Successful call:
INVITE  cseq 1  From NexTone
100 Trying  cseq 1  From Asterisk
100 Trying  cseq 1  From Asterisk
200 OK (G711U)  cseq 1  From Asterisk
ACK cseq 1  From NexTone
INVITE (G711U)  cseq 2  From NexTone
100 Trying  cseq 2  From Asterisk
200 OK  cseq 2  From Asterisk
ACK cseq 2  From NexTone
200 OK (711U)   cseq 1  From Asterisk
ACK cseq 1  From NexTone

Call continues until one side hangs up...


Failed Call:
Call audio is fine and all seems well but after 15 to 20 sec the call
drops...

INVITE  cseq 1  From NexTone
100 Trying  cseq 1  From Asterisk
100 Trying  cseq 1  From Asterisk
200 OK (G711U)  cseq 1  From Asterisk
INVITE (G711U)  cseq 2  From NexTone
100 Trying  cseq 2  From Asterisk
491 Request Pending cseq 2  From Asterisk
ACK cseq 1  From NexTone
ACK cseq 2  From NexTone
200 OK (G711U)  cseq 1  From Asterisk
ACK cseq 1  From NexTone
200 OK (G711U)  cseq 1  From Asterisk
ACK cseq 1  From NexTone
200 OK (G711U)  cseq 1  From Asterisk
ACK cseq 1  From NexTone
200 OK (G711U)  cseq 1  From Asterisk
ACK cseq 1  From NexTone
200 OK (G711U)  cseq 1  From Asterisk
ACK cseq 1  From NexTone
200 OK (G711U)  cseq 1  From Asterisk
ACK cseq 1  From NexTone
BYE cseq 1  From NexTone
200 OK  cseq 1  From Asterisk

I see this in the console after the call disconnects:

WARNING[24417]: chan_sip.c:1938 retrans_pkt: Maximum retries exceeded on
transmission [EMAIL PROTECTED] for seqno
1 (Critical Response)

This fails much more often then it is successful...

Anyone have a clue on this???


Stu Sheldon
Tech Committee Chairperson
S.C.A.L.E.



- --
Open up the window Let some air into this room I think I'm almost
chokin' From the smell of stale perfume And that cigarette you're
smoking 'Bout scared me half to death Open up the window, sucker
Let me catch my breath
  -- Three Dog Night - Mama Told Me Not to Come - Lyrics
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[asterisk-users] Asterisk 1.4.2 Requires Zaptel from 1.4 svn branch for zap_chan?

2007-03-21 Thread Stuart Sheldon
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Looks like there is a requirement for a later Zaptel the the 1.4.0
release for the zap_chan driver to build...

Stu

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They spell it da Vinci and pronounce it da Vinchy.  Foreigners
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-- Mark Twain
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Re: [asterisk-users] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped

2007-03-20 Thread Stuart Sheldon
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Are you using Answer() before VoiceMailMain()?

Stu


Timothy McKee wrote:
 I've been running the 8/1/2004 Head release up until a little over a
 week ago.  I was forced to due to a card failure to upgrade to 1.2.16
 without any advance preparation or testing (most of my connections are
 via satellite to all corners of the globe with high latency).
 
 Up until the upgrade I was running with very few issues.  Since the
 upgrade I have been experiencing strange issues with my Polycom SP-601
 phones.  My customers attempt to get their voicemail and Asterisk drops
 their connection ~15 seconds after they dial VM.  I have captured a SIP
 debug and included it (somewhat sanitized).  I'm not a SIP guru, but I
 can see the 15 second timer being set and I see repeated INVITEs being
 sent without any acks.  OPTIONs are being sent and acked.  The remote
 SIP phone is 'eden-1000a' and the voicemail extension is 9990.  *This
 worked just fine up until the upgrade.*
 
 Does this ring a bell with anyone out there???
 
 Tim McKee
 tmckee at sdnglobal dot com
 SDN Global
 
 ==
 
 pbx*CLI sip debug peer eden-1000a
 SIP Debugging Enabled for IP: 10.253.4.50:5060
 pbx*CLI
 -- SIP read from 10.253.4.50:5060:
 INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF
 From: eden-1000a
 sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3
 To: sip:[EMAIL PROTECTED];user=phone
 CSeq: 1 INVITE
 Call-ID: [EMAIL PROTECTED]
 Contact: sip:[EMAIL PROTECTED]
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
 NOTIFY, PRACK, UPDATE, REFER
 User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
 Supported: 100rel,replaces
 Allow-Events: talk,hold,conference
 Max-Forwards: 70
 Content-Type: application/sdp
 Content-Length: 245
 
 v=0
 o=- 978307756 978307756 IN IP4 10.253.4.50
 s=Polycom IP Phone
 c=IN IP4 10.253.4.50
 t=0 0
 m=audio 2228 RTP/AVP 0 18 8 101
 a=sendrecv
 a=rtpmap:0 PCMU/8000
 a=rtpmap:18 G729/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 
 --- (14 headers 11 lines) ---
 Using INVITE request as basis request -
 [EMAIL PROTECTED]
 Sending to 10.253.4.50 : 5060 (NAT)
 Reliably Transmitting (no NAT) to 10.253.4.50:5060:
 SIP/2.0 407 Proxy Authentication Required
 Via: SIP/2.0/UDP
 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF;received=10.253.4.50
 From: eden-1000a
 sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3
 To: sip:[EMAIL PROTECTED];user=phone;tag=as7f808f0f
 Call-ID: [EMAIL PROTECTED]
 CSeq: 1 INVITE
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
 Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
 nonce=2584558d
 Content-Length: 0
 
 
 ---
 Scheduling destruction of call '[EMAIL PROTECTED]'
 in 15000 ms
 Found user 'eden-1000a'
 pbx*CLI
 -- SIP read from 10.253.4.50:5060:
 INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF
 From: eden-1000a
 sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3
 To: sip:[EMAIL PROTECTED];user=phone
 CSeq: 1 INVITE
 Call-ID: [EMAIL PROTECTED]
 Contact: sip:[EMAIL PROTECTED]
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
 NOTIFY, PRACK, UPDATE, REFER
 User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
 Supported: 100rel,replaces
 Allow-Events: talk,hold,conference
 Max-Forwards: 70
 Content-Type: application/sdp
 Content-Length: 245
 
 v=0
 o=- 978307756 978307756 IN IP4 10.253.4.50
 s=Polycom IP Phone
 c=IN IP4 10.253.4.50
 t=0 0
 m=audio 2228 RTP/AVP 0 18 8 101
 a=sendrecv
 a=rtpmap:0 PCMU/8000
 a=rtpmap:18 G729/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 
 --- (14 headers 11 lines) ---
 Ignoring this INVITE request
 pbx*CLI
 -- SIP read from 10.253.4.50:5060:
 ACK sip:[EMAIL PROTECTED] SIP/2.0
 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF
 From: eden-1000a
 sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3
 To: sip:[EMAIL PROTECTED];user=phone;tag=as7f808f0f
 CSeq: 1 ACK
 Call-ID: [EMAIL PROTECTED]
 Contact: sip:[EMAIL PROTECTED]
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
 NOTIFY, PRACK, UPDATE, REFER
 User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
 Max-Forwards: 70
 Content-Length: 0
 
 
 --- (11 headers 0 lines) ---
 pbx*CLI
 -- SIP read from 10.253.4.50:5060:
 INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA
 From: eden-1000a
 sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3
 To: sip:[EMAIL PROTECTED];user=phone
 CSeq: 2 INVITE
 Call-ID: [EMAIL PROTECTED]
 Contact: sip:[EMAIL PROTECTED]
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
 NOTIFY, PRACK, UPDATE, REFER
 User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
 Supported: 100rel,replaces
 Allow-Events: talk,hold,conference
 Proxy-Authorization: Digest username=eden-1000a, realm=asterisk,
 nonce=2584558d, 

[asterisk-users] Setting Sip Headers From Dial App?

2007-03-05 Thread Stuart Sheldon
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

This might sound strange, but is there anyway for Asterisk to set extra
sip headers based on a sip phone returning a 302 in a dialplan?

Example:

PSTN = Asterisk = SIP-Phone, SIP-Phone returns 302 Redirect, Asterisk
sets X-Something: Some_Value  X-Somethingelse: Some_Other_Value, then
sends the new invite with added headers.

Stu Sheldon
ACT USA

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Randomly Generated Fortune Tag:
Q:  How do you catch a unique rabbit?
A:  Unique up on it!

Q:  How do you catch a tame rabbit?
A:  The tame way!
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[asterisk-users] CDR Help...

2006-10-11 Thread Stuart Sheldon
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Hash: SHA1

I'm having a hard time tracing calls that go through the parking lot. I
see the call placed from the original extension, and then it looks as if
it is hung up, then I see another extension grabbing a call from the
parking lot, but there are no fields in the parking lot call to tell me
the origin of the call... I see the same problem with a call transfers...

What am I missing here?

Any help would be great!

Stuart Sheldon



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A:  That's the Law of Spline Demand.
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Re: [asterisk-users] Asterisk Server : IDE HDD frequent crash

2006-10-06 Thread Stuart Sheldon
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I would look at ventilation if I were you. Drive failures at the rate
you are talking about can usually be traced back to thermal failures.

Just a thought

Stu


Dushyanth wrote:
 Hey guys,
 
 Iam having a peculiar problem with my asterisk installation. The specs 
 are..
 
 [EMAIL PROTECTED] ~]# asterisk -V
 Asterisk 1.2.7.1
 
 Wildcard: Digium Wildcard TE110P T1/E1
 Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 2 FXO, 2 FXS)
 Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 1 FXO, 3 FXS)
 Wildcard TDM: Wildcard TDM2400P Prototype (24 modules) (12 FXO's - rest 
 empty)
 
 Total 15 FX0's, 5 FXS out of which 5 to 6 FXO/FXS are being used. We have 
 about 300 active SIP accounts. 
 
 Queues, SIP extensions, Agents are in MySQL database using asterisk 
 realtime static.
 
 CPU : Intel(R) Xeon(TM) CPU 3.06GHz with Hyper threading
 RAM : 1G
 Mobo : Intel SE7501HG2
 
 The system is stable, however, the IDE disk crashes every 3/4 months. There 
 are DMA timeout errors for the IDE disk before it fails completely. The 
 same issue occured for the past three disks and I was doubting the 
 recommended hdparm setting 
 
 'hdparm -d 1 -X udma2 -c 3 /dev/IDE Device'
 
 So, I removed this setting after the last crash and the system workd fine 
 for another 3 months. Yes'day, the disk failed again with same symptoms. 
 All the disks were seagate baraccuda IDE drives.
 
 zttool doesnt show any IRQ misses even without the above hdparm setting and
 there is no noticeable problem in asterisk with the PRI line etc. Below is 
 my /proc/interrupts as well as /dev/hda settings.
 
 [EMAIL PROTECTED] ~]# cat /proc/interrupts
CPU0   CPU1
   0:   24771857   24719039IO-APIC-edge  timer
   1:102 62IO-APIC-edge  i8042
   8:  1  0IO-APIC-edge  rtc
   9:  0  0   IO-APIC-level  acpi
  14: 134159 135915IO-APIC-edge  ide0
 185:   32988610   16463264   IO-APIC-level  wctdm
 193:   22173177   27275710   IO-APIC-level  wctdm
 201:   21737611   27711650   IO-APIC-level  wctdm24xxp
 209:   22038077   27401613   IO-APIC-level  wcte11xp
 225:   18992311  0   IO-APIC-level  eth1
 233:1171166879   IO-APIC-level  eth0
 NMI:  0  0
 LOC:   49493157   49493156
 ERR:  0
 MIS:  0
 
 [EMAIL PROTECTED] ~]# hdparm -i /dev/hda
 
 /dev/hda:
 
  Model=ST340014A, FwRev=3.06, SerialNo=5JX96VFV
  Config={ HardSect NotMFM HdSw15uSec Fixed DTR10Mbs RotSpdTol.5% }
  RawCHS=16383/16/63, TrkSize=0, SectSize=0, ECCbytes=4
  BuffType=unknown, BuffSize=2048kB, MaxMultSect=16, MultSect=16
  CurCHS=16383/16/63, CurSects=16514064, LBA=yes, LBAsects=78165360
  IORDY=on/off, tPIO={min:240,w/IORDY:120}, tDMA={min:120,rec:120}
  PIO modes:  pio0 pio1 pio2 pio3 pio4
  DMA modes:  mdma0 mdma1 mdma2
  UDMA modes: udma0 udma1 udma2 udma3 udma4 *udma5
  AdvancedPM=no WriteCache=enabled
  Drive conforms to: ATA/ATAPI-6 T13 1410D revision 2:
 
  * signifies the current active mode
 
 I looked at the mailing lists and couldnt any such issues reported. 
 
 Please advice. Should i be using SCSI disks on RAID 1 or something ? Will 
 that help ?
 
 Also, should i be looking at any other mobo then Intel SE7501HG2 ? Iam 
 planning to put in a another asterisk server as a failover and would 
 appreciate inputs abt the kind of hardware i should be using for the system 
 with the specs i mentioned.
 
 Thanks
 Dushyanth
 
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Re: [asterisk-users] Zaptel-1.2.9 compile error

2006-09-08 Thread Stuart Sheldon
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I can confirm the same problem, it looks like the oct612x directory tree
is missing from the tarball

Stu


Bill Maidment wrote:
 Hi
 I've just tried to compile the zaptel-1.2.9 release and I get the
 following error:
 
 HOSTCC  /usr/local/src/zaptel-1.2.9/wct4xxp/fw2h
 /usr/local/src/zaptel-1.2.9/wct4xxp/fw2h
 /usr/local/src/zaptel-1.2.9/wct4xxp/OCT6114-128D.ima
 /usr/local/src/zaptel-1.2.9/wct4xxp/vpm450m_fw.h
 make[3]: *** No rule to make target
 `/usr/local/src/zaptel-1.2.9/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h',
 
 needed by `/usr/local/src/zaptel-1.2.9/wct4xxp/vpm450m.o'.  Stop.
 make[2]: *** [/usr/local/src/zaptel-1.2.9/wct4xxp] Error 2
 make[1]: *** [_module_/usr/local/src/zaptel-1.2.9] Error 2
 make[1]: Leaving directory `/usr/src/kernels/2.6.17-1.2174_FC5-i686'
 make: *** [linux26] Error 2
 
 zaptel-1.2.8 compiled OK. So what has changed? Did I do something wrong?
 Cheers
 Bill
 

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[asterisk-users] IAX2 Connection fails over time...

2006-07-27 Thread Stuart Sheldon
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hey all,

I have a x86 Pentium D asterisk system with two Digium 400's in it. I am
establishing a IAX2 Connection to another Asterisk system running on a
Solaris server.

When a call is placed between the two systems, everything seems fine for
a variable period of time, then for some reason beyond what my
diagnostics has found, the call begins to lag, and both asterisk's
servers report LAG. Network wise, the systems have a 4-10 msec ping
time. Once the call is terminated, everything returns to normal, and the
call can be reconnected. Until the call is terminated, no other calls
can be setup with that host.

Both systems are running 1.2.x. We are using the GSM codec for the calls.

Any ideas on what we should check?

Stu

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Re: [asterisk-users] Merlin Legend PRI not quite there yet

2006-07-27 Thread Stuart Sheldon
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Hash: SHA1

Maybe I missed something, but which side provides the clock? Have you
tried reversing the clock source? Sure looks like timing errors to me.

Stu Sheldon


Sterling Moses wrote:
 Greetings list,
 
 I have been beating around a problem for a while now and decided after
 weeks of searching I would try the list once again.
 
 The jist is this:
 
 Merlin Legend connected to verizon via a 100D (PRI) card. Works great.
 
 Asterisk connected via a crossover'd TE110P to a second 100D (PRI) in
 the legend.
 
 The green light on the TE110P comes on, but we have a red alarm on the
 100D card.
 
 Asterisk can see RED ALARMS on the PRI when we busy out the 100D card on
 the legend system. Asterisk also shows the alarms disappear when we
 restore the 100D card on the legend.  We reset the card and run a
 loopback. After the loopback the red alarm is gone for roughly 4 to 10
 minutes then turns back on. I believe this to be a threshold of errors
 being reached which triggers the red alarm.
 
 Checking the status of the 100D card shows it has a RED alarm and error
 events:
 Checking the error events shows
 
 MIS (Misframes) 250
 Slips: 57
 Errored Seconds: 255
 Bursty Seconds: 017
 Severely Errored Seconds: 255
 Frame Slips: 255
 
 We have replaced the crossover cable. (Using standard 1-5, 2-4, 4-2, 5-1
 pinout).
 
 We have replaced the 100D with a known good unit
 
 We have replaced the previous T100P with this new TE110P
 
 We replaced the server with a supermicro P4 server.
 
 All the results are the same.
 
 QUESTION is where can I go from here? What can we do to correct this
 problem? Could this be a timing setting on the legend switch?
 
 Anything would be appreciated, even a snicker or two about our still
 using merlin legends will be accepted. :-)
 
 Sterling.
 
 
 
 Asterisk intense debug on span 1:
 -
 
 --- SNIP 
 
 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
 extended) ]
 0 bytes of data
 asterisk1*CLI
  [ 00 03 7f ]
 
  Unnumbered frame:
  SAPI: 00  C/R: 0 EA: 0
   TEI: 001EA: 1
M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
 extended) ]
  0 bytes of data
 Sending Set Asynchronous Balanced Mode Extended
 
 [ 02 01 7f ]
 
 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
 extended) ]
 0 bytes of data
 asterisk1*CLI
  [ 00 03 7f ]
 
  Unnumbered frame:
  SAPI: 00  C/R: 0 EA: 0
   TEI: 001EA: 1
M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
 extended) ]
  0 bytes of data
 Sending Set Asynchronous Balanced Mode Extended
 
 [ 02 01 7f ]
 
 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
 extended) ]
 0 bytes of data
 asterisk1*CLI
  [ 00 03 7f ]
 
  Unnumbered frame:
  SAPI: 00  C/R: 0 EA: 0
   TEI: 001EA: 1
M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
 extended) ]
  0 bytes of data
 Sending Set Asynchronous Balanced Mode Extended
 
 [ 02 01 7f ]
 
 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
 extended) ]
 0 bytes of data
 
 -- snip ---
 
 
 ztcfg -v shows:
 
 Zaptel Configuration
 ==
 
 SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
 
 Channel map:
 
 Channel 01: Clear channel (Default) (Slaves: 01)
 Channel 02: Clear channel (Default) (Slaves: 02)
 Channel 03: Clear channel (Default) (Slaves: 03)
 Channel 04: Clear channel (Default) (Slaves: 04)
 Channel 05: Clear channel (Default) (Slaves: 05)
 Channel 06: Clear channel (Default) (Slaves: 06)
 Channel 07: Clear channel (Default) (Slaves: 07)
 Channel 08: Clear channel (Default) (Slaves: 08)
 Channel 09: Clear channel (Default) (Slaves: 09)
 Channel 10: Clear channel (Default) (Slaves: 10)
 Channel 11: Clear channel (Default) (Slaves: 11)
 Channel 12: Clear channel (Default) (Slaves: 12)
 Channel 13: Clear channel (Default) (Slaves: 13)
 Channel 14: Clear channel (Default) (Slaves: 14)
 Channel 15: Clear channel (Default) (Slaves: 15)
 Channel 16: Clear channel (Default) (Slaves: 16)
 Channel 17: Clear channel (Default) (Slaves: 17)
 Channel 18: Clear channel (Default) (Slaves: 18)
 Channel 19: Clear channel (Default) (Slaves: 19)
 Channel 20: Clear channel (Default) (Slaves: 20)
 Channel 21: Clear channel (Default) (Slaves: 21)
 Channel 22: Clear channel (Default) (Slaves: 22)
 Channel 23: Clear channel (Default) (Slaves: 23)
 Channel 24: D-channel (Default) (Slaves: 24)
 
 24 channels configured.
 
 ztclock shows:
 -
 ztclock - clock source accuracy test (3 passes)
 
 Flushing input buffer...
 Flush Complete.
 
 Test is approximately 3