date:20110722

I see, thank you for explaning. The reason for my concern is, we are
sometimes having DTMF issues on outbound calls. It seems when the user
(Polycom) enters digits, they are not being recognized by the other end.

On Thu, Jul 21, 2011 at 5:17 PM, Kevin P. Fleming wrote:

> On 07/21/2011 03:54 PM, vip killa wrote:
>
>> What if asterisk sends telephony events that are not in range of 0-15
>> though?
>>
>
> You are misunderstanding how SDP works; when an SDP offer or answer is
> sent, that indicates what the sender is willing to *receive*, not what it is
> going to send.
>
> If the Sonus device sent "fmtp:101 0-15" in its SDP, then Asterisk should
> not send 'event 16' events to it. If it does, that's a bug, although
> standard programming practices would mean that it wouldn't be harmful, it
> would just be ignored by the Sonus device.
>
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
> --
> __**__**_
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>  
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Re: [asterisk-users] asterisk's SDP

I have a call trace of one of these calls...and this seems strange:
asterisk sends on INVITE
a=fmtp:101 0-16
then 183 Session progress is sent back with:
a=fmtp:101 0-16
then asterisk sends 183 Session progress with:
a=fmtp:127 0-16
OK is sent back with:
a=fmtp:101 0-16
then asterisk sends OK with:
a=fmtp:127 0-16

Would the above cause DTMF not to be read on remote end?


On Fri, Jul 22, 2011 at 8:12 AM, vip killa  wrote:

> I see, thank you for explaning. The reason for my concern is, we are
> sometimes having DTMF issues on outbound calls. It seems when the user
> (Polycom) enters digits, they are not being recognized by the other end.
>
>
> On Thu, Jul 21, 2011 at 5:17 PM, Kevin P. Fleming wrote:
>
>> On 07/21/2011 03:54 PM, vip killa wrote:
>>
>>> What if asterisk sends telephony events that are not in range of 0-15
>>> though?
>>>
>>
>> You are misunderstanding how SDP works; when an SDP offer or answer is
>> sent, that indicates what the sender is willing to *receive*, not what it is
>> going to send.
>>
>> If the Sonus device sent "fmtp:101 0-15" in its SDP, then Asterisk should
>> not send 'event 16' events to it. If it does, that's a bug, although
>> standard programming practices would mean that it wouldn't be harmful, it
>> would just be ignored by the Sonus device.
>>
>>
>> --
>> Kevin P. Fleming
>> Digium, Inc. | Director of Software Technologies
>> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype:
>> kpfleming
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>> Check us out at www.digium.com & www.asterisk.org
>>
>> --
>> __**__**_
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>  http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>  
>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>
>
>
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Re: [asterisk-users] asterisk's SDP

2011-07-22 Thread Jesie Paluca

Most likely if DTMF is not recognized on the far end, it would be an
incompatibility setting of DTMF support or bug on either UAC and UAS.

Wireshark trace at both end will help you understand the issue.


On Fri, Jul 22, 2011 at 8:12 PM, vip killa  wrote:

> I see, thank you for explaning. The reason for my concern is, we are
> sometimes having DTMF issues on outbound calls. It seems when the user
> (Polycom) enters digits, they are not being recognized by the other end.
>
> On Thu, Jul 21, 2011 at 5:17 PM, Kevin P. Fleming wrote:
>
>> On 07/21/2011 03:54 PM, vip killa wrote:
>>
>>> What if asterisk sends telephony events that are not in range of 0-15
>>> though?
>>>
>>
>> You are misunderstanding how SDP works; when an SDP offer or answer is
>> sent, that indicates what the sender is willing to *receive*, not what it is
>> going to send.
>>
>> If the Sonus device sent "fmtp:101 0-15" in its SDP, then Asterisk should
>> not send 'event 16' events to it. If it does, that's a bug, although
>> standard programming practices would mean that it wouldn't be harmful, it
>> would just be ignored by the Sonus device.
>>
>>
>> --
>> Kevin P. Fleming
>> Digium, Inc. | Director of Software Technologies
>> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype:
>> kpfleming
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>> Check us out at www.digium.com & www.asterisk.org
>>
>> --
>> __**__**_
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>  http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>  
>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>
>
>
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> _
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Re: [asterisk-users] asterisk's SDP

How can we wireshark a trace on the remote end? It is a peer such as Level3
or Dash

On Fri, Jul 22, 2011 at 9:15 AM, Jesie Paluca wrote:

> Most likely if DTMF is not recognized on the far end, it would be an
> incompatibility setting of DTMF support or bug on either UAC and UAS.
>
> Wireshark trace at both end will help you understand the issue.
>
>
> On Fri, Jul 22, 2011 at 8:12 PM, vip killa  wrote:
>
>> I see, thank you for explaning. The reason for my concern is, we are
>> sometimes having DTMF issues on outbound calls. It seems when the user
>> (Polycom) enters digits, they are not being recognized by the other end.
>>
>> On Thu, Jul 21, 2011 at 5:17 PM, Kevin P. Fleming 
>> wrote:
>>
>>> On 07/21/2011 03:54 PM, vip killa wrote:
>>>
 What if asterisk sends telephony events that are not in range of 0-15
 though?

>>>
>>> You are misunderstanding how SDP works; when an SDP offer or answer is
>>> sent, that indicates what the sender is willing to *receive*, not what it is
>>> going to send.
>>>
>>> If the Sonus device sent "fmtp:101 0-15" in its SDP, then Asterisk should
>>> not send 'event 16' events to it. If it does, that's a bug, although
>>> standard programming practices would mean that it wouldn't be harmful, it
>>> would just be ignored by the Sonus device.
>>>
>>>
>>> --
>>> Kevin P. Fleming
>>> Digium, Inc. | Director of Software Technologies
>>> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype:
>>> kpfleming
>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>>> Check us out at www.digium.com & www.asterisk.org
>>>
>>> --
>>> __**__**
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>  http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>  
>>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>>
>>
>>
>> --
>> _
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>>
>
>
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Re: [asterisk-users] Strange network issue

2011-07-22 Thread Mark Deneen

On Thu, Jul 21, 2011 at 7:13 PM, Mike Diehl  wrote:

> Hi all,
>
> I've got a strange problem with a customer's phones.
>
> They've got a bunch of Grandstreams that seem to be rock solid... until
> 7:00pm.  At 7:00, some of the phones become unavailable, and stay down.
>  Call
> quality is solid almost all the time.  But right at 7:00, things go bad.
>  Only
> some of the phone lines go down and they stay down until the phone is
> rebooted.
>
> I'm not even sure what to look for when I go to the site.  Any ideas?
>
>
Many years ago, in my college days, the network in one building would fail
around a certain time every day.  The sun would hit the network closet
around the same time every day in the summer, causing the equipment to
overheat and temporarily fail.

I would go there and observe everything which happens at 7:00.  Maybe it's
something that a cleaning service inadvertently does, like faulty wiring + a
vacuum cleaner.

-M
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[asterisk-users] Question about codec re-negotiation in asterisk 1.4.X

2011-07-22 Thread Matteo Campana

Hi all,
I have a major issue with a codec renegotiation in an asterisk 1.4.33.1
setup, which leads me to ask a general question about asterisk 1.4.X codec
negotiation: asterisk can support a re-negotiation of a codec "on the fly"
through a re-Invite? If my SIP provider sends me a re-invite changing codec
from g729 to g711, asterisk properly handle the matter?
I see in the trace that asterisk responds 200 OK to the provider, but *does
not send the re-invite to the UAC, and stops to send rtp to the UAC*.

In mantis/jira I see this issue:
https://issues.asterisk.org/jira/browse/ASTERISK-17261?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel#issue-tabs
which
is similar, and Matthew Nicholson wrote this comment: "This is not the way
fax passthrough works in asterisk. If you would like to do passthrough in
this manner, set up the channel for ulaw or alaw from the start, or use T.38
passthrough. T.38 passthrough is the only fax passthrough configuration that
we officially support".

So asterisk does not support a codec change "on the fly" with a re-invite,
unless it is a t38 re-invite?
This behaviour is also present in new asterisk versions (eg asterisk
1.4.42)?

Thanks in advance,
Matteo
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Re: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X

2011-07-22 Thread Eric Wieling

Asterisk does not support changing codecs on the fly.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana
Sent: Friday, July 22, 2011 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X

Hi all,
I have a major issue with a codec renegotiation in an asterisk 1.4.33.1 setup, 
which leads me to ask a general question about asterisk 1.4.X codec 
negotiation: asterisk can support a re-negotiation of a codec "on the fly" 
through a re-Invite? If my SIP provider sends me a re-invite changing codec 
from g729 to g711, asterisk properly handle the matter?
I see in the trace that asterisk responds 200 OK to the provider, but does not 
send the re-invite to the UAC, and stops to send rtp to the UAC.

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Re: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X

2011-07-22 Thread Matteo Campana

On Fri, Jul 22, 2011 at 4:46 PM, Eric Wieling  wrote:

> ** **
>
> Asterisk does not support changing codecs on the fly.
>


And why asterisk sends 200 OK to the provider, if does not support its
re-invite?

M.




> 
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Matteo Campana
> *Sent:* Friday, July 22, 2011 10:39 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Question about codec re-negotiation in
> asterisk 1.4.X
>
> ** **
>
> Hi all,
>
> I have a major issue with a codec renegotiation in an asterisk 1.4.33.1
> setup, which leads me to ask a general question about asterisk 1.4.X codec
> negotiation: asterisk can support a re-negotiation of a codec "on the fly"
> through a re-Invite? If my SIP provider sends me a re-invite changing codec
> from g729 to g711, asterisk properly handle the matter?
>
> I see in the trace that asterisk responds 200 OK to the provider, but *does
> not send the re-invite to the UAC, and stops to send rtp to the UAC*.
>
> ** **
>
> ** **
>
> -
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Re: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X

 From: "Eric Wieling" 
Sent: Friday, July 22, 2011 10:46 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] Question about codec re-negotiation in 
asterisk 1.4.X

Asterisk does not support changing codecs on the fly.   From: 
asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo 
Campana
Sent: Friday, July 22, 2011 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question about codec re-negotiation in asterisk 
1.4.X   Hi all,  I have a major issue with a codec renegotiation in an 
asterisk 1.4.33.1 setup, which leads me to ask a general question about 
asterisk 1.4.X codec negotiation: asterisk can support a re-negotiation of 
a codec "on the fly" through a re-Invite? If my SIP provider sends me a 
re-invite changing codec from g729 to g711, asterisk properly handle the 
matter?   I see in the trace that asterisk responds 200 OK to the provider, 
but does not send the re-invite to the UAC, and stops to send rtp to the 
UAC.   
I know 1.8.x can do this but. I don't think 1.4.x had the option. Check it 
out.

exten => Process,1,Set(SIP_CODEC=ulaw)  Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003 

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Re: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X

2011-07-22 Thread Eric Wieling

Asterisk supports reinvites (if reinvites are enabled in sip.conf), just not 
changing codecs in the middle of the call.  If anyone has managed to get it 
to work, I'd love to hear about it.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana
Sent: Friday, July 22, 2011 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about codec re-negotiation in asterisk 
1.4.X

On Fri, Jul 22, 2011 at 4:46 PM, Eric Wieling 
mailto:ewiel...@nyigc.com>> wrote:

Asterisk does not support changing codecs on the fly.

And why asterisk sends 200 OK to the provider, if does not support its 
re-invite?

M.

From: 
asterisk-users-boun...@lists.digium.com

[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Matteo Campana
Sent: Friday, July 22, 2011 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X

Hi all,
I have a major issue with a codec renegotiation in an asterisk 1.4.33.1 setup, 
which leads me to ask a general question about asterisk 1.4.X codec 
negotiation: asterisk can support a re-negotiation of a codec "on the fly" 
through a re-Invite? If my SIP provider sends me a re-invite changing codec 
from g729 to g711, asterisk properly handle the matter?
I see in the trace that asterisk responds 200 OK to the provider, but does not 
send the re-invite to the UAC, and stops to send rtp to the UAC.

-
--
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Re: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X

Eric

With 1.8.x I use.

exten => Process,1,Set(SIP_CODEC=ulaw)

And the system kicks the call over to ulaw. Now this is just prior to the 
answer so I don't know if it meets your criteria. But it works great to enforce 
inline T.30 audio faxes. I also use the f/F option T.38 or T.30 on recevie fax. 
This option was added as part of a patch in 1.8 and is in the 1.10/2.0 branch.  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003


 From: "Eric Wieling" 
Sent: Friday, July 22, 2011 11:06 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] Question about codec re-negotiation in asterisk 
1.4.X

  Asterisk supports reinvites (if reinvites are enabled in sip.conf), just not 
changing codecs in the middle of the call.  If anyone has managed to get it 
to work, I'd love to hear about it. From: 
asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana
Sent: Friday, July 22, 2011 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about codec re-negotiation in asterisk 
1.4.XOn Fri, Jul 22, 2011 at 4:46 PM, Eric Wieling  
wrote: Asterisk does not support changing codecs on the fly.And 
why asterisk sends 200 OK to the provider, if does not support its re-invite?   
M.   From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana
Sent: Friday, July 22, 2011 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X 
   Hi all,  I have a major issue with a codec renegotiation in an asterisk 
1.4.33.1 setup, which leads me to ask a general question about asterisk 1.4.X 
codec negotiation: asterisk can support a re-negotiation of a codec "on the 
fly" through a re-Invite? If my SIP provider sends me a re-invite changing 
codec from g729 to g711, asterisk properly handle the matter?   I see in the 
trace that asterisk responds 200 OK to the provider, but does not send the 
re-invite to the UAC, and stops to send rtp to the UAC. -

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Re: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X

2011-07-22 Thread Eric Wieling

Ah, we do not use 1.8 yet.I've been unable to get 1.8 to transcode between 
g722 and ulaw.   I assume it is a config issue.

Does your (pre-answer) example change the codec for BOTH legs of the call or 
just the incoming leg or outgoing leg?  When I was referring to a "call" I 
meant both legs of the call.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman
Sent: Friday, July 22, 2011 11:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about codec re-negotiation in asterisk 
1.4.X

Eric

With 1.8.x I use.

exten => Process,1,Set(SIP_CODEC=ulaw)

And the system kicks the call over to ulaw. Now this is just prior to the 
answer so I don't know if it meets your criteria. But it works great to enforce 
inline T.30 audio faxes. I also use the f/F option T.38 or T.30 on recevie fax. 
This option was added as part of a patch in 1.8 and is in the 1.10/2.0 branch.
Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003


Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003


From: "Eric Wieling" 
Sent: Friday, July 22, 2011 11:06 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] Question about codec re-negotiation in asterisk 
1.4.X
Asterisk supports reinvites (if reinvites are enabled in sip.conf), just not 
changing codecs in the middle of the call.  If anyone has managed to get it 
to work, I'd love to hear about it.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana
Sent: Friday, July 22, 2011 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about codec re-negotiation in asterisk 
1.4.X


On Fri, Jul 22, 2011 at 4:46 PM, Eric Wieling 
mailto:ewiel...@nyigc.com>> wrote:

Asterisk does not support changing codecs on the fly.


And why asterisk sends 200 OK to the provider, if does not support its 
re-invite?

M.




From: 
asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Matteo Campana
Sent: Friday, July 22, 2011 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X

Hi all,
I have a major issue with a codec renegotiation in an asterisk 1.4.33.1 setup, 
which leads me to ask a general question about asterisk 1.4.X codec 
negotiation: asterisk can support a re-negotiation of a codec "on the fly" 
through a re-Invite? If my SIP provider sends me a re-invite changing codec 
from g729 to g711, asterisk properly handle the matter?
I see in the trace that asterisk responds 200 OK to the provider, but does not 
send the re-invite to the UAC, and stops to send rtp to the UAC.


-

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Re: [asterisk-users] Phase out macro command.

Hey all

I am looking at pulling out the macro command from all of my dial plans. 
This is due to the long term phase out of macros as per the upgrade 
documentation.   I would like to get feed back from others on how they 
might go about this. Based on what I see it looks like Gosub / Return is 
the way to do this I have already started this with new code written for 
1.8.x.. From what I have found this requires variables to be scoped to the 
entire call plan verses the older method of passing macro level variables. 


I would love to hear others take on this topic as well before I pull out 
all of my macros.

Thanks

Bryant Zimmerman (ZK Tech Inc.)
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Re: [asterisk-users] Phase out macro command.

 From: "Bryant Zimmerman" 
Sent: Friday, July 22, 2011 11:21 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Phase out macro command.

Hey all

I am looking at pulling out the macro command from all of my dial plans. 
This is due to the long term phase out of macros as per the upgrade 
documentation.   I would like to get feed back from others on how they 
might go about this. Based on what I see it looks like Gosub / Return is 
the way to do this I have already started this with new code written for 
1.8.x.. From what I have found this requires variables to be scoped to the 
entire call plan verses the older method of passing macro level variables. 

I would love to hear others take on this topic as well before I pull out 
all of my macros.

Thanks

Bryant Zimmerman (ZK Tech Inc.)

Sorry guys I went back and read the wiki again and see that I can pass 
arguments in the gosub. I have been out there multiple times and for some 
reason it just had not got it until I stopped and read word for word again. 
It now make more sense.  I guess some days it just takes asking the 
question to wake up and see the answer has been there staring me in the 
face. Too many hours too little sleep some days. 

Thanks
Bryant

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Re: [asterisk-users] a=sendonly Music On Hold ignored

2011-07-22 Thread Matthew J. Roth

Michael,

It looks like your problem is caused by a phone with a non-standard
SDP session version implementation.  The phone is sending an INVITE
with SDP that contains an "a=sendonly" line.  Asterisk should respond
with an OK that contains an "a=recvonly" line, but it responds with
"a=sendrecv" instead.

Asterisk is probably ignoring the SDP in the INVITE, because the phone
never updates the SDP session version.  It remains at 0 throughout the
entire dialog:

 o=  usernamesess-id sess-version  nettype  addrtype  addr
 --  --    ---    -
 o=  500 109600  0 IN   IP4   192.168.1.109
 o=  500 109601  0 IN   IP4   192.168.1.109
 o=  500 109600  0 IN   IP4   192.168.1.109

Luckily, there is an option to force Asterisk to ignore the SDP
session version number and treat all SDP data as new data.  Try adding
"ignoresdpversion=yes" to the phone's configuration in sip.conf.

For the sake of future list readers, please respond to this post with
"[RESOLVED]" appended to the subject line if this fixes your problem.

> That's what I get: 
> [root@pbx ~]# asterisk -rx 'module show' | egrep 
> 'format_g729.so|format_pcm.so' 
> format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 
> format_g729.so Raw G729 data 0 
> 
> What does the 5 (or in my case 0) stand for? 

It's the "Use Count", but I'm not certain how to interpret it.  On one
of my servers chan_sip.so has a use count of 110451 and rising.  That
seems like it should indicate some sort of leak, but Asterisk has been
running continuously for over half a year without any problems.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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[asterisk-users] Connecting to a Taqua switch

2011-07-22 Thread Philip Prindeville

Anyone have any configuration experience connecting Asterisk 1.8 to the PSTN 
via SIP on a Taqua 7000 switch?

My local carrier recently upgraded software and changed their configs so that 
signalling and media are on different cards (and hence different IP addresses), 
and it's causing issues.

I suspect there are other factors at play... it may or may not be behind a 
properly configured SBC.

Thanks,

-Philip

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Re: [asterisk-users] Connecting to a Taqua switch

2011-07-22 Thread Jim Dickenson

My provider has always sent the SIP control info from one IP and the media 
packets from another. As long as your firewall passes the data there should be 
no problem. I did not have to do anything special in my configuration. This is 
using ABE which is based on 1.4.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jul 22, 2011, at 10:09 AM, Philip Prindeville wrote:

> Anyone have any configuration experience connecting Asterisk 1.8 to the PSTN 
> via SIP on a Taqua 7000 switch?
> 
> My local carrier recently upgraded software and changed their configs so that 
> signalling and media are on different cards (and hence different IP 
> addresses), and it's causing issues.
> 
> I suspect there are other factors at play... it may or may not be behind a 
> properly configured SBC.
> 
> Thanks,
> 
> -Philip
> 
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Re: [asterisk-users] asterisk rpm build problem

2011-07-22 Thread Barry Miller

On Fri, Jul 22, 2011 at 10:10:01AM +0200, marek cervenka wrote:
> hi,
> 
> i'm trying build asterisk rpm
> normal compilation is ok but rpm building always fail
> 
> centos6/asterisk 1.8.5.0
> 
> any ideas?
> 
> 
> gcc -o recno/rec_utils.o -c recno/rec_utils.c -MD -MT recno/rec_utils.o 
> -MF .recno_rec_utils.o.d -MP -pthread -Wall -D__DBINTERFACE_PRIVATE -I. 
> -I.. -Iinclude -Ihash -Ibtree -Irecno 
> -I/root/rpmbuild/BUILD/asterisk-1.8.5.0/include -O2 -g -march=i386 
 [snip]
> /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: 
> undefined reference to `__sync_fetch_and_add_4'
> /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: 
> undefined reference to `__sync_fetch_and_add_4'
> ccss.o: In function `ast_atomic_fetchadd_int':
> /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: 
> undefined reference to `__sync_fetch_and_add_4'
> /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: 
> undefined reference to `__sync_fetch_and_add_4'
> /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: 
> undefined reference to `__sync_fetch_and_add_4'
> cdr.o:/root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: 
> more undefined references to `__sync_fetch_and_add_4' follow
> utils.o: In function `ast_atomic_dec_and_test':
> /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:646: 
> undefined reference to `__sync_sub_and_fetch_4'
> utils.o: In function `ast_atomic_fetchadd_int':
> /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: 
> undefined reference to `__sync_fetch_and_add_4'

I do not use centos (or build rpms, for that matter), but at least on
OpenBSD the "-march=i386" causes the same undefineds you are seeing.

I have to force i686 when configuring asterisk:
./configure --prefix=/usr/local CFLAGS=-march=i686 
for the build to succeed.

I'm not sure if this applies in your case, but maybe it will help.

-- 
Barry

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Re: [asterisk-users] Strange network issue

2011-07-22 Thread Dave Platt

> They've got a bunch of Grandstreams that seem to be rock solid... until 
> 7:00pm.  At 7:00, some of the phones become unavailable, and stay down.  Call 
> quality is solid almost all the time.  But right at 7:00, things go bad.  
> Only 
> some of the phone lines go down and they stay down until the phone is 
> rebooted.
> 
> I'm not even sure what to look for when I go to the site.  Any ideas?

I'd look to see if there are any electrical circuits (lights,
fans, etc.) which are on a timer of some sort, and are automatically
powered off at 7 PM.

If somebody mistakenly plugged a piece of network kit into such a
circuit, it would lose power at that time, and your network might
end up being partitioned, or routing (switch or IP-level) might
change abruptly.

If (for example) the phones were being DHCP-provisioned with
network numbers and a "here is your default gateway" configuration,
and that gateway were to lose power, the phones would lose connectivity,
and might not recover until they discarded their DHCP credentials
and routing information, and broadcast for a new configuration...
which would happen if they were power-cycled, or (if not then)
many hours later.

Similar things could happen if (for example) a janitor were to
plug a floor polisher into a power circuit shared with servers
or network equipment... the turn-on / turn-off power sags and
spikes might knock networking gear off-line.  [This is not
a hypothetical example... numerous cases of this sort of thing
have been reported over the years.]

If these phones are being DHCP-provisioned, you might want to
check each phone and see what configuration has been acquired...
i.e. if it got its information from the "real" DHCP server,
or from some other source.  I've had network problems in the past
result from people plugging some sort of "all-in-one" appliance
or server into an existing net... the appliance starts trying to
provide DHCP service and routing on its own.  This can seriously
disrupt the network... either immediately (if the appliance's
configuration is incompatible with the network) or at a later time
(if e.g. the appliance is acting as a router, and is then powered
off).



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[asterisk-users] 10.0.0 better than 2.0.0?

2011-07-22 Thread Tony Mountifield

I read Kevin's piece in asterisk-announce about the new numbering scheme,
and saw in svn-commits some tagging of 10.0.0-beta1.

Perhaps I'm thick (I hope not!), but I really can't see why calling the
next version 10.0.0 is any better than calling it 2.0.0!

I'm surprised not to have seen ANY talk in asterisk-users or aserisk-dev
about it, since the announcement.

Cheers
Tony
-- 
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] 10.0.0 better than 2.0.0?

I agree, the numbering seems to make no sense. Oh well it's just an
arbitrary measurement of non-progress anyway

On Fri, Jul 22, 2011 at 3:44 PM, Tony Mountifield wrote:

> I read Kevin's piece in asterisk-announce about the new numbering scheme,
> and saw in svn-commits some tagging of 10.0.0-beta1.
>
> Perhaps I'm thick (I hope not!), but I really can't see why calling the
> next version 10.0.0 is any better than calling it 2.0.0!
>
> I'm surprised not to have seen ANY talk in asterisk-users or aserisk-dev
> about it, since the announcement.
>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: t...@softins.co.uk - http://www.softins.co.uk
> Play: t...@mountifield.org - http://tony.mountifield.org
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] 10.0.0 better than 2.0.0?

2011-07-22 Thread Danny Nicholas

I thought it was going to be 1.10.0

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Friday, July 22, 2011 2:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 10.0.0 better than 2.0.0?

I agree, the numbering seems to make no sense. Oh well it's just an
arbitrary measurement of non-progress anyway

On Fri, Jul 22, 2011 at 3:44 PM, Tony Mountifield 
wrote:

I read Kevin's piece in asterisk-announce about the new numbering scheme,
and saw in svn-commits some tagging of 10.0.0-beta1.

Perhaps I'm thick (I hope not!), but I really can't see why calling the
next version 10.0.0 is any better than calling it 2.0.0!

I'm surprised not to have seen ANY talk in asterisk-users or aserisk-dev
about it, since the announcement.

Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] 10.0.0 better than 2.0.0?

2011-07-22 Thread Patrick Lists


Hi Tony,

On 07/22/2011 09:44 PM, Tony Mountifield wrote:

I read Kevin's piece in asterisk-announce about the new numbering scheme,
and saw in svn-commits some tagging of 10.0.0-beta1.


Totally missed that one. Just did a quick browse.


Perhaps I'm thick (I hope not!), but I really can't see why calling the
next version 10.0.0 is any better than calling it 2.0.0!


Don't think you are thick. There's no difference other than that 10 
sounds more mature. So I'd say it's marketing L33tSp34k.



I'm surprised not to have seen ANY talk in asterisk-users or aserisk-dev
about it, since the announcement.


Frankly I could not care less about version numbers. Instead I prefer 
stability, speedy bug fixes and new cool features. But 2.0.0 has my vote.


Regards,
Patrick

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Re: [asterisk-users] 10.0.0 better than 2.0.0?

2011-07-22 Thread Norbert Zawodsky


Maybe just a typo ? Misplaced dots between all those 1's and 0's ...
Maybe we should call it version "12" instead of 1100 ;-)

Am 22.07.2011 21:50, schrieb Danny Nicholas:


I thought it was going to be 1.10.0

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa

*Sent:* Friday, July 22, 2011 2:48 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] 10.0.0 better than 2.0.0?

I agree, the numbering seems to make no sense. Oh well it's just an 
arbitrary measurement of non-progress anyway


On Fri, Jul 22, 2011 at 3:44 PM, Tony Mountifield 
mailto:t...@mountifield.org>> wrote:


I read Kevin's piece in asterisk-announce about the new numbering scheme,
and saw in svn-commits some tagging of 10.0.0-beta1.

Perhaps I'm thick (I hope not!), but I really can't see why calling the
next version 10.0.0 is any better than calling it 2.0.0!

I'm surprised not to have seen ANY talk in asterisk-users or aserisk-dev
about it, since the announcement.

Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk  - 
http://www.softins.co.uk
Play: t...@mountifield.org  - 
http://tony.mountifield.org


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--
I know we've come a long way,
We're changing day to day,
But tell me, where do the children play?

Yusuf Islam, Nov. 1970


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Re: [asterisk-users] 10.0.0 better than 2.0.0?

2011-07-22 Thread C F

Since this change I started measuring temperature in Rankine. Its now
592.67 degrees here (south NJ).

On Fri, Jul 22, 2011 at 3:44 PM, Tony Mountifield  wrote:
> I read Kevin's piece in asterisk-announce about the new numbering scheme,
> and saw in svn-commits some tagging of 10.0.0-beta1.
>
> Perhaps I'm thick (I hope not!), but I really can't see why calling the
> next version 10.0.0 is any better than calling it 2.0.0!
>
> I'm surprised not to have seen ANY talk in asterisk-users or aserisk-dev
> about it, since the announcement.
>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: t...@softins.co.uk - http://www.softins.co.uk
> Play: t...@mountifield.org - http://tony.mountifield.org
>
> --
> _
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>

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Re: [asterisk-users] 10.0.0 better than 2.0.0?

2011-07-22 Thread Warren Selby

On Fri, Jul 22, 2011 at 2:58 PM, Norbert Zawodsky wrote:

> Maybe just a typo ? Misplaced dots between all those 1's and 0's ...
> Maybe we should call it version "12" instead of 1100 ;-)
>
> Am 22.07.2011 21:50, schrieb Danny Nicholas:
>
>>
>> I thought it was going to be 1.10.0
>>
>>
No, they're referring to the new asterisk numbering system announced
yesterday on the asterisk-announce mailing list[1].  Basically, the
consensus was (amongst Digium employees I assume, since I didn't see any
discussion on the topic on this list or the -dev list (although I admit I
don't follow the -dev list as closely)) that there is basically never going
to be a change so drastic to the asterisk core that it would warrant calling
it Asterisk 2.0.  Because of this, the whole concept of having 1.x releases
becomes redundant, since it leads one to believe that eventually there will
be a 2.0, so they're dropping the "1." part of the version numbers, and
starting with what would have been version 1.10, they'll just start calling
it version 10.0.  The next version would be Asterisk 11.0, and then Asterisk
12.0, etc.

Personally, I think this is a horrible idea.  I thought Digium would have
better sense than this, especially after the failed experiment with the
1.6.x numbering change that they reverted on within 18 months.  The
confusion caused by the several different version numbers is significant,
and unnecessary.  Also, the fact that there have been three separate
numbering schemes in the last 3 years give me the impression of instability
and insecurity in the maintainers of the project, because if something as
simple as incrementing a version number can't be figured out, how reliable
is the asterisk project itself?  Big version number changes should indicate
big changes to the core of the system, not just a change in the philosophy
on how you want to number and market your project.

I guess what I'm getting at is this - if you're convinced that there's
enough change to the core to warrant a version number change away from 1.x,
then just make it 2.0, not 10.0.  Jumping from 1.8 to 10.0 is just
confusing.

[1] -
http://lists.digium.com/pipermail/asterisk-announce/2011-July/000331.html

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--Warren Selby, dCAP
http://www.SelbyTech.com 
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Re: [asterisk-users] 10.0.0 better than 2.0.0?

2011-07-22 Thread Danny Nicholas

Maybe they are trying to live down their bad press from 1.6 and 1.8 by
abandoning the 1.X schema.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Friday, July 22, 2011 3:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 10.0.0 better than 2.0.0?

On Fri, Jul 22, 2011 at 2:58 PM, Norbert Zawodsky 
wrote:

Maybe just a typo ? Misplaced dots between all those 1's and 0's ...
Maybe we should call it version "12" instead of 1100 ;-)

Am 22.07.2011 21:50, schrieb Danny Nicholas:

I thought it was going to be 1.10.0

No, they're referring to the new asterisk numbering system announced
yesterday on the asterisk-announce mailing list[1].  Basically, the
consensus was (amongst Digium employees I assume, since I didn't see any
discussion on the topic on this list or the -dev list (although I admit I
don't follow the -dev list as closely)) that there is basically never going
to be a change so drastic to the asterisk core that it would warrant calling
it Asterisk 2.0.  Because of this, the whole concept of having 1.x releases
becomes redundant, since it leads one to believe that eventually there will
be a 2.0, so they're dropping the "1." part of the version numbers, and
starting with what would have been version 1.10, they'll just start calling
it version 10.0.  The next version would be Asterisk 11.0, and then Asterisk
12.0, etc.

Personally, I think this is a horrible idea.  I thought Digium would have
better sense than this, especially after the failed experiment with the
1.6.x numbering change that they reverted on within 18 months.  The
confusion caused by the several different version numbers is significant,
and unnecessary.  Also, the fact that there have been three separate
numbering schemes in the last 3 years give me the impression of instability
and insecurity in the maintainers of the project, because if something as
simple as incrementing a version number can't be figured out, how reliable
is the asterisk project itself?  Big version number changes should indicate
big changes to the core of the system, not just a change in the philosophy
on how you want to number and market your project.

I guess what I'm getting at is this - if you're convinced that there's
enough change to the core to warrant a version number change away from 1.x,
then just make it 2.0, not 10.0.  Jumping from 1.8 to 10.0 is just
confusing.  

[1] -
http://lists.digium.com/pipermail/asterisk-announce/2011-July/000331.html

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--Warren Selby, dCAP
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Re: [asterisk-users] 10.0.0 better than 2.0.0?

2011-07-22 Thread Matthew J. Roth

Kevin P. Fleming: The versions all go to ten. Look, right across the
board, ten, ten, ten and...

Asterisk Users: Oh, I see. And most open source projects upgrade to
two?

Kevin P. Fleming: Exactly.

Asterisk Users: Does that mean it's better? Is it any better?

Kevin P. Fleming: Well, it's eight better, isn't it? It's not two. You
see, most blokes, you know, will be running at two. You're on two
here, all the way up, all the way up, all the way up, you're on two on
your software. Where can you go from there? Where?

Asterisk Users: I don't know.

Kevin P. Fleming: Nowhere. Exactly. What we do is, if we need that
extra push over the cliff, you know what we do?

Asterisk Users: Put it up to ten.

Kevin P. Fleming: Ten. Exactly. Eight better.

Asterisk Users: Why don't you just make two better and make two be the
top number and make that a little better?

Kevin P. Fleming: [pause] Asterisk goes to ten.

--

Sorry, couldn't resist.

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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[asterisk-users] Asterisk 10.0.0 Beta 1 Now Available!

2011-07-22 Thread Asterisk Development Team

The Asterisk Development Team is pleased to announce the first beta release of
Asterisk 10.0.0-beta1. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

With the release of the Asterisk 10 branch, the preceding '1.' has been removed
from the version number per the blog post available at
http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/

All interested users of Asterisk are encouraged to participate in the
Asterisk 10 testing process. Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira. It is also very useful to see
successful test reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing
channel on IRC to work together in testing the many parts of Asterisk.
Additionally users can make use of the RPM and DEB packages now being built for
all Asterisk releases. More information available at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages

Asterisk 10 is the next major release series of Asterisk. It will be a
Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk
versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

A short list of included features includes:

* T.38 gateway functionality has been added to res_fax.
* Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
* New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
* Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
* Support for defining hints has been added to pbx_lua.
* Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
* Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/10/CHANGES?view=checkout

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta1

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] Strange network issue

2011-07-22 Thread Hans Witvliet

On Fri, 2011-07-22 at 10:58 -0700, Dave Platt wrote:
> > They've got a bunch of Grandstreams that seem to be rock solid... until 
> > 7:00pm.  At 7:00, some of the phones become unavailable, and stay down.  
> > Call 
> > quality is solid almost all the time.  But right at 7:00, things go bad.  
> > Only 
> > some of the phone lines go down and they stay down until the phone is 
> > rebooted.
> > 
> > I'm not even sure what to look for when I go to the site.  Any ideas?
> 
> I'd look to see if there are any electrical circuits (lights,
> fans, etc.) which are on a timer of some sort, and are automatically
> powered off at 7 PM.
> 
> If somebody mistakenly plugged a piece of network kit into such a
> circuit, it would lose power at that time, and your network might
> end up being partitioned, or routing (switch or IP-level) might
> change abruptly.
> 

Hi,

Even if there is no equipment you own controlled by a timer, you still
can suffer from it.

Some power companies have different rates for power you use during
daytime or at night.
So even if _you_ don't have equipment on a timer, your neighbours might
have. Something like electrical boilers or so, or other "heavy
equipment". Switching them on/off can cause huge spikes on the
electrical wires.

A couple of neigbours at work have their own micro-power-generators.
About one in ten times, when they start delevering power to the grid,
all of our test-systems go down. Only the systems behind the
re-generated UPS (that removes spikes from the powerlines) are protected
against them.

So nasty litte spikes are harder to detect/tracedown than a full
blackout.

hw

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[asterisk-users] use dahdi for local terminal modem access?

2011-07-22 Thread William Stillwell

I have some terminals that have phone lines.

 

One of my tech had an idea of using IAXmodem or something similar to use
existing PRI/DAHDI Trucks for dial out via the asterisk/Linux console.

 

Anybody ever heard of doing this?

 

I would think maybe would use iaxmodem maybe and a shell terminal app?

 

 

(basically I'm dialing into a remote access device that uses a pots like for
remote administration, and don't want to string a channel bank off my
asterisk box, and a hook to a modem)

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[asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

Hello,

I am wondering if there is a way to drop SIP packets for generic
transactions? For example, only SIP PEERs are allowed to call in and receive
ACK or Declined rather that those inviting a call who are not PEERs at all.

Currently my Asterisk setup sends, "*SIP/2.0 603 Declined" *to any stranger
invites because my dialplan includes Hangup(). Is there any way I can not
send a 603 declined so to mislead the probe runner?

Thanks
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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"


On 07/22/2011 07:32 PM, Bruce B wrote:

Hello,

I am wondering if there is a way to drop SIP packets for generic
transactions? For example, only SIP PEERs are allowed to call in and
receive ACK or Declined rather that those inviting a call who are not
PEERs at all.

Currently my Asterisk setup sends, "*SIP/2.0 603 Declined" *to any
stranger invites because my dialplan includes Hangup(). Is there any
way I can not send a 603 declined so to mislead the probe runner?


There is really no way to accomplish that except with a firewall.


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Re: [asterisk-users] use dahdi for local terminal modem access?

2011-07-22 Thread Lyle Giese


On 07/22/11 18:13, William Stillwell wrote:

I have some terminals that have phone lines.

One of my tech had an idea of using IAXmodem or something similar to use
existing PRI/DAHDI Trucks for dial out via the asterisk/Linux console.

Anybody ever heard of doing this?

I would think maybe would use iaxmodem maybe and a shell terminal app?

(basically I’m dialing into a remote access device that uses a pots like
for remote administration, and don’t want to string a channel bank off
my asterisk box, and a hook to a modem)



--


Depends on your expectation.  Because of compression in the codecs, it 
will be hard to get fast dialup.  If you mean ssh or telnet, it might 
work.  If you mean vnc or RDP over this, you may not get enough usable 
bandwidth to do that.


Given this, I have in an emergency dialed into a RAS server via a VoIP 
line. My laptop connected at 14,400bps.  All I needed to do was telnet 
into an APC masterswitch to toggle power on one outlet.  It worked.


I was surprised at getting a 14,400bps connect.  I was not expecting 
that high and really did not need that high.  300 baud probably would 
have been fast enough to telnet into an APC masterswitch.


Lyle Giese
LCR Computer Services, Inc.

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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

Thanks for the input. I am really surprised. But yes, I want exactly what
firewall does, DROP packet instead of REJECTING it.

So, you are saying that one has to tamper the SIP stack to add the option to
not respond to un-trusted sources?
I really thought Asterisk might have this built in as a feature.


I can't even do a dialplan search for a registered PEER because even if I
find the IP to not be a trusted I still need to Hangup() on the invite which
in turn send 603 Declined.

There isn't really any work-around to this?

Thanks again


On Fri, Jul 22, 2011 at 7:39 PM, Alex Balashov wrote:

> On 07/22/2011 07:32 PM, Bruce B wrote:
>
>> Hello,
>>
>> I am wondering if there is a way to drop SIP packets for generic
>> transactions? For example, only SIP PEERs are allowed to call in and
>> receive ACK or Declined rather that those inviting a call who are not
>> PEERs at all.
>>
>> Currently my Asterisk setup sends, "*SIP/2.0 603 Declined" *to any
>> stranger invites because my dialplan includes Hangup(). Is there any
>> way I can not send a 603 declined so to mislead the probe runner?
>>
>
> There is really no way to accomplish that except with a firewall.
>
>
> --
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> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
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>
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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

Asterisk does not expose low-level control of its SIP stack.  It's something 
intended to be configured and used at the application level.

If you really want to do this without a firewall, put a Kamailio proxy in front 
of your Asterisk install and drop things as you see fit.  But why go through 
the trouble?  What's wrong with iptables?

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Web: http://www.evaristesys.com/

On Jul 22, 2011, at 9:30 PM, Bruce B  wrote:

> Thanks for the input. I am really surprised. But yes, I want exactly what 
> firewall does, DROP packet instead of REJECTING it.
> 
> So, you are saying that one has to tamper the SIP stack to add the option to 
> not respond to un-trusted sources?
> I really thought Asterisk might have this built in as a feature.
> 
> 
> I can't even do a dialplan search for a registered PEER because even if I 
> find the IP to not be a trusted I still need to Hangup() on the invite which 
> in turn send 603 Declined. 
> 
> There isn't really any work-around to this?
> 
> Thanks again
> 
> 
> On Fri, Jul 22, 2011 at 7:39 PM, Alex Balashov  
> wrote:
> On 07/22/2011 07:32 PM, Bruce B wrote:
> Hello,
> 
> I am wondering if there is a way to drop SIP packets for generic
> transactions? For example, only SIP PEERs are allowed to call in and
> receive ACK or Declined rather that those inviting a call who are not
> PEERs at all.
> 
> Currently my Asterisk setup sends, "*SIP/2.0 603 Declined" *to any
> stranger invites because my dialplan includes Hangup(). Is there any
> way I can not send a 603 declined so to mislead the probe runner?
> 
> There is really no way to accomplish that except with a firewall.
> 
> 
> -- 
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
> 
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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

2011-07-22 Thread Paul Belanger


On 11-07-22 07:32 PM, Bruce B wrote:

Hello,

I am wondering if there is a way to drop SIP packets for generic
transactions? For example, only SIP PEERs are allowed to call in and receive
ACK or Declined rather that those inviting a call who are not PEERs at all.

Currently my Asterisk setup sends, "*SIP/2.0 603 Declined" *to any stranger
invites because my dialplan includes Hangup(). Is there any way I can not
send a 603 declined so to mislead the probe runner?


Have you tried disabling guests?

sip.conf
[general]
allowguest=no

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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

Paul,

Won't that just send a 403 Forbidden?

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Web: http://www.evaristesys.com/

On Jul 22, 2011, at 9:48 PM, Paul Belanger  wrote:

> On 11-07-22 07:32 PM, Bruce B wrote:
>> Hello,
>> 
>> I am wondering if there is a way to drop SIP packets for generic
>> transactions? For example, only SIP PEERs are allowed to call in and receive
>> ACK or Declined rather that those inviting a call who are not PEERs at all.
>> 
>> Currently my Asterisk setup sends, "*SIP/2.0 603 Declined" *to any stranger
>> invites because my dialplan includes Hangup(). Is there any way I can not
>> send a 603 declined so to mislead the probe runner?
>> 
> Have you tried disabling guests?
> 
> sip.conf
> [general]
> allowguest=no
> 
> -- 
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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

2011-07-22 Thread Paul Belanger


On 11-07-22 09:51 PM, Alex Balashov wrote:

Paul,

Won't that just send a 403 Forbidden?

I believe so, but I was proposing a different SIP message then 603 
Declined.  As you mentioned, a firewall is the real solution if OP wants 
to drop packets.


Asterisk is a B2BUA, not a firewall.

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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

Yeah, a 403 would still not be accepted. Thanks for the input.

So, what do you do with the firewall? You are suggesting just to allow
trusted parties and not others? Well, that is not possible in my case. Vast
number of scattered users all over the globe. I hate to think there is no
way to not announce ourselves as a SIP server to un-trusted users.

Or is there something else that can be done with the firewall to all
"dynamic" trust IPs and drop packets from unregistered sources?

Thanks again

On Fri, Jul 22, 2011 at 10:04 PM, Paul Belanger wrote:

> On 11-07-22 09:51 PM, Alex Balashov wrote:
>
>> Paul,
>>
>> Won't that just send a 403 Forbidden?
>>
>>  I believe so, but I was proposing a different SIP message then 603
> Declined.  As you mentioned, a firewall is the real solution if OP wants to
> drop packets.
>
> Asterisk is a B2BUA, not a firewall.
>
>
> --
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>
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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"


On 07/22/2011 10:04 PM, Paul Belanger wrote:


On 11-07-22 09:51 PM, Alex Balashov wrote:

Paul,

Won't that just send a 403 Forbidden?


I believe so, but I was proposing a different SIP message then 603
Declined.


:-)  Ah, I see.

Yeah, it seems the OP is looking for a means by which Asterisk can be 
made to not respond with any SIP feedback at all.


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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"


On 07/22/2011 10:11 PM, Bruce B wrote:


Vast number of scattered users all over the globe. I hate to think
there is no way to not announce ourselves as a SIP server to
un-trusted users.


Not easily.  This is a problem all service providers have to deal with, 
and so do you.  You have to have your SIP services open to the world, 
but they don't necessarily need to be easy to DoS or dictionary scan.


Intra-industrially, the solution is usually some form of SBC or other 
administrative border/edge security element.  In the open-source world, 
a lot of the steeling, rate-limiting, etc. can be done with 
OpenSER/Kamailio/OpenSIPS.


(Shameless plug: That's what we do all day commercially.)

A common strategy is to use a non-standard SIP port ('bindport' in 
sip.conf).  No, it doesn't stop all scans, but in our experience, it 
will stop a good 95%+ of them.  When almost everyone does use the 
standard SIP port, and thus there are so many low-hanging targets, it's 
not worth bothering with a full ~65k UDP port scan.  Certainly, the 
average SIPvicious scanner won't bother with anything but 5060.



Or is there something else that can be done with the firewall to all
 "dynamic" trust IPs and drop packets from unregistered sources?


That raises an interesting question:

How do the users register to begin with, if their REGISTER requests 
won't be processed unless their IP is already known to be a registrant? 
 :-)


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Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

Thanks again for the depth of knowledge you are offering.

So, I am taking a pass on the firewall since it won't do what I need but I
understand that it can do country block etc...thought not a full proof
still.

I am really not worried about DoS or more importantly DDoS as I have no hope
those can be prevented anyhowbeen hit by one on a pfSense router and it
was just absorb as much as you can.

I like the different port idea though with the current scattered ATAs and
SIP phones it's unpractical for me to ask them all to change to a random
port.

Quote,* "How do the users register to begin with, if their REGISTER requests
won't be processed unless their IP is already known to be a registrant?
 :-)"*

Well, unfortunately I don't have the luxury of knowing their IP and the
closest I know is their IP range.

But I guess this is what is as I have seen big providers also return back
DECLINED from their gateways if one is not on their authorized list.

So, my final questions:

1- So, you are saying that either of OpenSER/Kamailio/OpenSIPS actually give
me the full capability to the SIP stack to do the sort of thing I was asking
for? And this can run on the same server as Asterisk is running?

Thanks a bunch

On Fri, Jul 22, 2011 at 10:18 PM, Alex Balashov
wrote:

> On 07/22/2011 10:11 PM, Bruce B wrote:
>
>  Vast number of scattered users all over the globe. I hate to think
>> there is no way to not announce ourselves as a SIP server to
>> un-trusted users.
>>
>
> Not easily.  This is a problem all service providers have to deal with, and
> so do you.  You have to have your SIP services open to the world, but they
> don't necessarily need to be easy to DoS or dictionary scan.
>
> Intra-industrially, the solution is usually some form of SBC or other
> administrative border/edge security element.  In the open-source world, a
> lot of the steeling, rate-limiting, etc. can be done with
> OpenSER/Kamailio/OpenSIPS.
>
> (Shameless plug: That's what we do all day commercially.)
>
> A common strategy is to use a non-standard SIP port ('bindport' in
> sip.conf).  No, it doesn't stop all scans, but in our experience, it will
> stop a good 95%+ of them.  When almost everyone does use the standard SIP
> port, and thus there are so many low-hanging targets, it's not worth
> bothering with a full ~65k UDP port scan.  Certainly, the average SIPvicious
> scanner won't bother with anything but 5060.
>
>
>  Or is there something else that can be done with the firewall to all
>>  "dynamic" trust IPs and drop packets from unregistered sources?
>>
>
> That raises an interesting question:
>
> How do the users register to begin with, if their REGISTER requests won't
> be processed unless their IP is already known to be a registrant?  :-)
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>  http://www.asterisk.org/hello
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> To UNSUBSCRIBE or update options visit:
>  
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>
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Re: [asterisk-users] use dahdi for local terminal modem access?

2011-07-22 Thread William Stillwell

Um, no VOIP involved here.

I have an asterisk server with 2 23B+D PRI's

I want to telnet/ssh into the asterisk server, and make an outbound call 
serial based modem/terminal connection (Like the 80/90's BBS Days). 

No TCP/IP or PPP or crazyness

(ie, dialing into a Modem set to AA hooked to a Cisco Console Port)



> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Lyle Giese
> Sent: Friday, July 22, 2011 8:07 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] use dahdi for local terminal modem
> access?
> 
> On 07/22/11 18:13, William Stillwell wrote:
> > I have some terminals that have phone lines.
> >
> > One of my tech had an idea of using IAXmodem or something similar to
> use
> > existing PRI/DAHDI Trucks for dial out via the asterisk/Linux
> console.
> >
> > Anybody ever heard of doing this?
> >
> > I would think maybe would use iaxmodem maybe and a shell terminal
> app?
> >
> > (basically I'm dialing into a remote access device that uses a pots
> like
> > for remote administration, and don't want to string a channel bank
> off
> > my asterisk box, and a hook to a modem)
> >
> >
> >
> > --
> 
> Depends on your expectation.  Because of compression in the codecs, it
> will be hard to get fast dialup.  If you mean ssh or telnet, it might
> work.  If you mean vnc or RDP over this, you may not get enough usable
> bandwidth to do that.
> 
> Given this, I have in an emergency dialed into a RAS server via a VoIP
> line. My laptop connected at 14,400bps.  All I needed to do was telnet
> into an APC masterswitch to toggle power on one outlet.  It worked.
> 
> I was surprised at getting a 14,400bps connect.  I was not expecting
> that high and really did not need that high.  300 baud probably would
> have been fast enough to telnet into an APC masterswitch.
> 
> Lyle Giese
> LCR Computer Services, Inc.
> 
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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

2011-07-22 Thread Steve Edwards


On Fri, 22 Jul 2011, Bruce B wrote:

1- So, you are saying that either of OpenSER/Kamailio/OpenSIPS actually 
give me the full capability to the SIP stack to do the sort of thing I 
was asking for? And this can run on the same server as Asterisk is 
running?


Configure OpenSIPS to listen to 5060 and Asterisk to listen to 5061.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000--
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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

2011-07-22 Thread Robert-iPhone

I like to put mine on 3389

hahaha just kidding.

Personally I'm starting to convert to FreeSwitch - oops I had to say it.

Security can be difficult and there are some good SBCs out there - just begs 
investment in technology - OH and bright staff


Sent from my iPhone

On Jul 23, 2011, at 12:09 AM, Steve Edwards  wrote:

> On Fri, 22 Jul 2011, Bruce B wrote:
> 
>> 1- So, you are saying that either of OpenSER/Kamailio/OpenSIPS actually give 
>> me the full capability to the SIP stack to do the sort of thing I was asking 
>> for? And this can run on the same server as Asterisk is running?
> 
> Configure OpenSIPS to listen to 5060 and Asterisk to listen to 5061.
> 
> -- 
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
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> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"