[asterisk-users] Change RX Signalling Bits in Dahdi drivers

[Optical Phoenix]

Greeting,
I am trying to setup PLAR signalling in asterisk. I have modified the FXSLS
TX bits in dahdi-base.c on line 2580, and I can make calls.

.sig_type = DAHDI_SIG_FXSLS,
.bits[DAHDI_TXSIG_ONHOOK]  = DAHDI_BITS_ABCD, /*changed by  for PLAR*/
.bits[DAHDI_TXSIG_OFFHOOK] = (0), /*changed by  for PLAR*/
.bits[DAHDI_TXSIG_START]   = DAHDI_BITS_ABCD, /*changed by for PLAR*/

When I got to change the rx, its a bit more complex. I have learned from
this list that dahdi_rbsbits() handles the rx bits, but my changes seem to
have no effect. Does anyone have a good understanding of this function? I
would appreciate any help you can provide.


case DAHDI_SIG_FXSLS:
if (!(cursig & DAHDI_BBIT)) {  /*Dennis RINGING  */  /*<- I think
this is checking if the state is different from a set value? needs
clarification*/
/* Check for ringing first */
__dahdi_hooksig_pvt(chan, DAHDI_RXSIG_RING);
break;
}
if ((chan->sig != DAHDI_SIG_FXSLS) && (cursig & DAHDI_ABIT)) {  /*<--
Why is it checking DAHDI_SIG_FXSLS? do I need to modify this to be 
also?*/
/* if went on hook */
__dahdi_hooksig_pvt(chan, DAHDI_RXSIG_ONHOOK);  /*<--I think this
is passing it to a function that reacts to the signal in this case the
onhook signal?*/
} else {
__dahdi_hooksig_pvt(chan, DAHDI_RXSIG_OFFHOOK);  /* <-- same here but
with the off hook? */
}
break;
   case DAHDI_SIG_CAS:
/* send event that something changed */
__qevent(chan, DAHDI_EVENT_BITSCHANGED);
break;

Thanks
Dennis
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[asterisk-users] Redirect incoming call to SIP trunk.

[Luis H. Forchesatto]

Greetings.

I got two asterisk servers, one is connected to another via sip trunk. The
incoming calls are routed to the time period an if matches is transfered to
the designed extension. If don't, is redirected to a second time period.
Then, if the call matches the second time period it need to be transfered
to the trunk that directs to the second server.

How do I do to configure it this way?

The trunk it must be transfered has a outbound route configured too.

Any further details just ask.

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Re: [asterisk-users] What would cause a drop between two asterisk systems?

[Duncan Turnbull]

On 6/03/2013, at 9:06 AM, John Novack  wrote:

> 
> Carlos Alvarez wrote:
>> 
>> 
>> On Tue, Mar 5, 2013 at 2:32 PM, Hose  
>> wrote:
>> We have an asterisk frontend terminating all our SIP phones to, and an
>> asterisk backend with a wildcard PRI card in it connecting to the PTSN.
>> The frontend handles 99% of dialplan logic and just hands off anything
>> outgoing to the backend via IAX2, which dials out on one of the open
>> channels.
>> 
>> IAX is buggy.  We've never seen a reliable system using it.  We've given up 
>> on it.
IAX seems easy to me

We run interoffice from NZ to Australia and many systems in between. 

No issues at all

Cheers Duncan


> I have seen this assertion from time to time, but never any real details
> 
> There is a world wide network of users who communicate using IAX, and many 
> with PSTN service from providers using IAX. with no complaints
> Can someone please provide meaningful details on what "buggy" really means? 
> Rather than such a sweeping condemnation. If it is so buggy, why isn't it 
> either fixed or discontinued?
> 
> It certainly is much less prone to hacking and abuse than SIP. Probably not 
> due to the protocol design as much as it isn't as universal
> 
> John Novack
> 
> 
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Re: [asterisk-users] What would cause a drop between two asterisk systems?

[Hose]

What you say...John Novack (jnov...@stromberg-carlson.org):

> 
> Carlos Alvarez wrote:
> >
> >
> >On Tue, Mar 5, 2013 at 2:32 PM, Hose  >> wrote:
> >
> >We have an asterisk frontend terminating all our SIP phones to, and an
> >asterisk backend with a wildcard PRI card in it connecting to the PTSN.
> >The frontend handles 99% of dialplan logic and just hands off anything
> >outgoing to the backend via IAX2, which dials out on one of the open
> >channels.
> >
> >
> >IAX is buggy.  We've never seen a reliable system using it.  We've given up 
> >on it.
> I have seen this assertion from time to time, but never any real details
> 
> There is a world wide network of users who communicate using IAX, and many 
> with PSTN service from providers using IAX. with no complaints
> Can someone please provide meaningful details on what "buggy" really means? 
> Rather than such a sweeping condemnation. If it is so buggy, why isn't it 
> either fixed or discontinued?
> 
> It certainly is much less prone to hacking and abuse than SIP. Probably not 
> due to the protocol design as much as it isn't as universal
> 
> John Novack

I'll keep the SIP option open - never really considered it actually,
just figured the go-to protocol for connecting two * boxes together
would be IAX2. It'd be good for troubleshooting purposes at least to
narrow down issues and compare results.

If anyone else has suggestions, I'm all ears.

hose

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Re: [asterisk-users] fail2ban filter issue

[John Novack]

eherr wrote:


Not sure if this has been answered but I cannot find a solution.

I am running Asterisk 1.4.26.3


Very VERY old. If you need to continue in 1.4, you should really up to the last 
1.4.44
Many MANY changes and broken changes between 1.4.26 and 1.4.44

John Novack


I am seeing the following lines in my log files:

A: [2013-03-05 13:54:27] NOTICE[6928] chan_sip.c: Failed to authenticate user 
;tag=DmVIjOlfYiiL

B: [2013-03-05 12:20:00] NOTICE[6928] chan_sip.c: Failed to authenticate user 
101;tag=eec630f1

C: [2013-03-03 05:15:02] NOTICE[31158] chan_sip.c: Registration from 
'"101381"' failed for '85.25.23.129' - No 
matching peer found

Now Two Part Question:

Part 1:

I understand that line C is from some soft phone like Xlite, IP phone, or 
program trying to register extension 101381 to my server and the user exten 
does not exist. I don't understand the method for A or B. I don't understand 
what generates that error message. Can someone explain?

Part 2:

Fail2ban blocks line C as per the regex in filters/asterisk.conf. What I don't 
understand is why doesn't lines A or B have a built in regex line? This goes 
back to not knowing the method that generates the error message in part 1.

Also, can I update the regex in asterisk.conf,

From: NOTICE.* .*: Failed to authenticate user .*@.* 


To:  NOTICE.* .*: Failed to authenticate user .*

It should ban both A and B, along with the original Regex line that I modified.

Question is, would this present a problem under normal circumstances? I know 
when the line comes up with my.asterisk.server.ip it will get ignored because I 
am in the ignoreip list but I want to make sure it will be OK to adjust.

Thanks community!

-E



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Re: [asterisk-users] What would cause a drop between two asterisk systems?

[John Novack]

Carlos Alvarez wrote:



On Tue, Mar 5, 2013 at 2:32 PM, Hose mailto:hose+aster...@bluemaggottowel.com>> wrote:

We have an asterisk frontend terminating all our SIP phones to, and an
asterisk backend with a wildcard PRI card in it connecting to the PTSN.
The frontend handles 99% of dialplan logic and just hands off anything
outgoing to the backend via IAX2, which dials out on one of the open
channels.


IAX is buggy.  We've never seen a reliable system using it.  We've given up on 
it.

I have seen this assertion from time to time, but never any real details

There is a world wide network of users who communicate using IAX, and many with 
PSTN service from providers using IAX. with no complaints
Can someone please provide meaningful details on what "buggy" really means? 
Rather than such a sweeping condemnation. If it is so buggy, why isn't it either fixed or 
discontinued?

It certainly is much less prone to hacking and abuse than SIP. Probably not due 
to the protocol design as much as it isn't as universal

John Novack


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[asterisk-users] fail2ban filter issue

[eherr]

Not sure if this has been answered but I cannot find a solution.

 

I am running Asterisk 1.4.26.3

 

I am seeing the following lines in my log files:

 

A: [2013-03-05 13:54:27] NOTICE[6928] chan_sip.c: Failed to authenticate user 
;tag=DmVIjOlfYiiL

 

B: [2013-03-05 12:20:00] NOTICE[6928] chan_sip.c: Failed to authenticate user 
101;tag=eec630f1

 

C: [2013-03-03 05:15:02] NOTICE[31158] chan_sip.c: Registration from 
'"101381"' failed for
'85.25.23.129' - No matching peer found

 

Now Two Part Question:

 

Part 1: 

 

I understand that line C is from some soft phone like Xlite, IP phone, or 
program trying to register extension 101381 to my server
and the user exten does not exist. I don't understand the method for A or B. I 
don't understand what generates that error message.
Can someone explain?

 

Part 2:

 

Fail2ban blocks line C as per the regex in filters/asterisk.conf. What I don't 
understand is why doesn't lines A or B have a built
in regex line? This goes back to not knowing the method that generates the 
error message in part 1. 

 

Also, can I update the regex in asterisk.conf,  

 

From: NOTICE.* .*: Failed to authenticate user .*@  
.* 

To:  NOTICE.* .*: Failed to authenticate user .*

 

It should ban both A and B, along with the original Regex line that I modified.

 

Question is, would this present a problem under normal circumstances? I know 
when the line comes up with my.asterisk.server.ip it
will get ignored because I am in the ignoreip list but I want to make sure it 
will be OK to adjust.

 

Thanks community!

 

-E

 

 

 

 

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Re: [asterisk-users] What would cause a drop between two asterisk systems?

[Carlos Alvarez]

On Tue, Mar 5, 2013 at 2:32 PM, Hose wrote:

> We have an asterisk frontend terminating all our SIP phones to, and an
> asterisk backend with a wildcard PRI card in it connecting to the PTSN.
> The frontend handles 99% of dialplan logic and just hands off anything
> outgoing to the backend via IAX2, which dials out on one of the open
> channels.
>

IAX is buggy.  We've never seen a reliable system using it.  We've given up
on it.  I'd try SIP.  Easy to do, no real reason not to.

Check all of the networking involved.  Leave a ping test running between
the systems constantly, then see if it dropped packets when you get a
dropped call.


-- 
Carlos Alvarez
TelEvolve
602-889-3003
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[asterisk-users] What would cause a drop between two asterisk systems?

[Hose]

We have an asterisk frontend terminating all our SIP phones to, and an
asterisk backend with a wildcard PRI card in it connecting to the PTSN.
The frontend handles 99% of dialplan logic and just hands off anything
outgoing to the backend via IAX2, which dials out on one of the open
channels.

Lately we've been getting a disconnected calls. Keeping the consoles
running it doesn't seem to be the PRI initiating the hangups, as I'll
when I see hangups intiiated on the backend / PRI side:

  -- Span 2: Channel 0/21 got hangup request, cause 16

Instead, I'm seeing 

 == Spawn extension (outbound, (dialed #), 3) exited non-zero on 
'IAX2/asterisk-frontend2-603'
-- Hungup 'IAX2/asterisk-frontend2-603'

Which indicates the frontend initiated a hangup. But on the frontend I'm
seeing auto fallthroughs to the h extension, which only happens if the
hangup is initiated from the backend:

-- Auto fallthrough, channel 'SIP/phone1-0167' status is 'ANSWER'
(h extension stuff follows)

If that side was initiating the hangup, I'd just see a jump to the h
extension, with no auto fallthrough. So it looks like there may be a
communication interruption between the front and backends.

The problem is this happens intermittently, so I can't reproduce it
reliably. I've held open a call for 30+ minutes and not run into the
problem, while someone's been on a call for 7 minutes and this happens.
It doesn't seem feasible to constantly run IAX2 debugs from the console
on any open call - does anyone have suggestions on how to troubleshoot
this? Weirdly enough, this only seems to happen when users dial into
conference bridges (not local) such as WebEx and GoToMeeting, but that
might just be because of the length of those calls. 

Will tweaking things like the IAX2 jitter buffer help? The two systems 
are barely four hops apart with an average of .2 ms ping times between
them on a very resilient network (two of those hops are through core
transports). I've never seen ping loss between them, even when running
ping tests for hours during heavy call volume periods. The loads on the
machines are minimal - never seen the load go above .10 during normal
operation. But it does seem like something between them is making them
drop calls. 

hose

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Re: [asterisk-users] red alarm on span - do channels in the group automatically get skipped over?

[Hose]

What you say...Yves A. (yves...@gmx.de):

> yes, this is the way, asterisk / the channeldriver handles it.
> you can simulate the failure of one span by just pulling out the
> cable and see what happens..
> on top, you can influence the order, the channels are used by using
> dahdi/g1 or dahdi/G1...
> regards,
> yves
> 
> Am 05.03.2013 07:31, schrieb Hose:
> >Hello,
> >
> >If I put two spans' worth of channels, say 1-23 from span 1 and 25-47 in
> >span 2, in one group, but only span 2 was showing OK and the other was
> >down / showing a RED alarm, would asterisk automatically skip over
> >trying to use channels 1-23 when doing outbound calls? e.g.,
> >dial(dahdi/g1/(number) would just jump to channel 25?
> >
> >Testing seems to bear this out, but I'm not positive about it.
> >
> >hose

Thanks gentlemen!

hose

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Re: [asterisk-users] Error to install Asterisk

[termo termosel]

Hi, Ok, tomorrow I will send the output when I will be in the office! Thanks!
 > From: asterisk_l...@earthshod.co.uk
> To: asterisk-users@lists.digium.com
> Date: Tue, 5 Mar 2013 16:11:01 +
> Subject: Re: [asterisk-users] Error to install Asterisk
> 
> On Tuesday 05 March 2013, termo termosel wrote:
> > Hi,
> > when I try to install Asterisk 11.2.1 the console return error which it
> > tells: /usr/bin/ld: final link failed: No space left on device
> > and the process exits installation.
> > How can I solve this problem? Tmp folder is empty.
> > Thanks,Jordi
> 
> Try entering this command:
> # df -h
> and paste the complete output in a message.
> 
> This will show the amount of space used and remaining on all filesystems, in 
> human-readable notation  (i.e. it will automatically select the units: bytes, 
> kilo, mega, giga or terabytes, so as to get a sensible figure).
> 
> You'll almost certainly have to move some files out of the way.  Have you 
> got, 
> or can you get, a USB external HDD; which either already has a Linux ext4 
> file 
> system on it, or contains only sacrificial data?  
> 
> -- 
> AJS
> 
> Answers come *after* questions.
> 
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Re: [asterisk-users] Error to install Asterisk

[A J Stiles]

On Tuesday 05 March 2013, termo termosel wrote:
> Hi,
> when I try to install Asterisk 11.2.1 the console return error which it
> tells: /usr/bin/ld: final link failed: No space left on device
> and the process exits installation.
> How can I solve this problem? Tmp folder is empty.
> Thanks,Jordi

Try entering this command:
# df -h
and paste the complete output in a message.

This will show the amount of space used and remaining on all filesystems, in 
human-readable notation  (i.e. it will automatically select the units: bytes, 
kilo, mega, giga or terabytes, so as to get a sensible figure).

You'll almost certainly have to move some files out of the way.  Have you got, 
or can you get, a USB external HDD; which either already has a Linux ext4 file 
system on it, or contains only sacrificial data?  

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Error to install Asterisk

[Gertjan Baarda]

I stand corrected..

On Tue, Mar 5, 2013 at 4:58 PM, Hans Witvliet  wrote:

> -Original Message-
> From: termo termosel 
> Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> To: asterisk-users@lists.digium.com 
> Subject: Re: [asterisk-users] Error to install Asterisk
> Date: Tue, 5 Mar 2013 14:30:05 +
>
> Hi,
>
>
> if I write du -sh the response is 271M. I don't know that it means.
>
>
> Thanks,
> Jordi
>
>
>
> -Original Message-
>
> Hi Jordi,
>
> The "du" utility will show you the Disk Utilisation (hence the
> abbriviation du)
>
> What might be more relevant, is how much space is free.
> That you can examine with: df -h
>
>
> hw
>
>
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Re: [asterisk-users] Error to install Asterisk

[Hans Witvliet]

-Original Message-
From: termo termosel 
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion

To: asterisk-users@lists.digium.com 
Subject: Re: [asterisk-users] Error to install Asterisk
Date: Tue, 5 Mar 2013 14:30:05 +

Hi,


if I write du -sh the response is 271M. I don't know that it means.


Thanks,
Jordi



-Original Message-

Hi Jordi,

The "du" utility will show you the Disk Utilisation (hence the
abbriviation du)

What might be more relevant, is how much space is free.
That you can examine with: df -h


hw


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Re: [asterisk-users] Error to install Asterisk

[Gertjan Baarda]

It means 271 megabyte. Can you post the complete output?

Sent from my iPhone

On 5 mrt. 2013, at 15:31, termo termosel  wrote:

 Hi,

if I write du -sh the response is 271M. I don't know that it means.

Thanks,
Jordi

--
From: gertjan.baa...@gmail.com
Date: Tue, 5 Mar 2013 14:29:37 +0100
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Error to install Asterisk

Is there enough space on the device left?
Check this with: du -sh
--Gertjan

On Tue, Mar 5, 2013 at 2:20 PM, termo termosel wrote:

Hi,

when I try to install Asterisk 11.2.1 the console return error which it
tells:

/usr/bin/ld: final link failed: No space left on device

and the process exits installation.

How can I solve this problem? Tmp folder is empty.

Thanks,
Jordi

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Re: [asterisk-users] Error to install Asterisk

[termo termosel]

Hi,
if I write du -sh the response is 271M. I don't know that it means.
Thanks,Jordi

From: gertjan.baa...@gmail.com
Date: Tue, 5 Mar 2013 14:29:37 +0100
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Error to install Asterisk

Is there enough space on the device left? Check this with: du -sh--Gertjan

On Tue, Mar 5, 2013 at 2:20 PM, termo termosel  wrote:





Hi,
when I try to install Asterisk 11.2.1 the console return error which it tells:
/usr/bin/ld: final link failed: No space left on device


and the process exits installation.
How can I solve this problem? Tmp folder is empty.
Thanks,Jordi  

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Re: [asterisk-users] multiple sipusers tables

[Vieri]

Right, thanks!

--- On Tue, 3/5/13, Gertjan Baarda  wrote:

Maybe you can workaround it by creating a view in SQL?-- Gertjan

On Tue, Mar 5, 2013 at 2:10 PM, Vieri  wrote:


Hi,



I have 2 databases DBNAME1 and DBNAME2. In each database I have a table called 
sipusers (so DBNAME1.sipusers and DBNAME2.sipusers).



Can I use both sipusers tables in Asterisk RealTime?



Something like this:

/etc/asterisk/extconfig.conf:

[settings]

sipusers => odbc,DBNAME1,sipusers

sippeers => odbc,DBNAME1,sipusers

sipusers => odbc,DBNAME2,sipusers

sippeers => odbc,DBNAME2,sipusers



If Asterisk 11 doesn't support this right now, will it in the future?



Thanks,



Vieri





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Re: [asterisk-users] Error to install Asterisk

[Gertjan Baarda]

Is there enough space on the device left?
Check this with: du -sh
--Gertjan

On Tue, Mar 5, 2013 at 2:20 PM, termo termosel wrote:

> Hi,
>
> when I try to install Asterisk 11.2.1 the console return error which it
> tells:
>
> /usr/bin/ld: final link failed: No space left on device
>
> and the process exits installation.
>
> How can I solve this problem? Tmp folder is empty.
>
> Thanks,
> Jordi
>
> --
> _
> -- 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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Re: [asterisk-users] multiple sipusers tables

[Gertjan Baarda]

Maybe you can workaround it by creating a view in SQL?
-- Gertjan

On Tue, Mar 5, 2013 at 2:10 PM, Vieri  wrote:

> Hi,
>
> I have 2 databases DBNAME1 and DBNAME2. In each database I have a table
> called sipusers (so DBNAME1.sipusers and DBNAME2.sipusers).
>
> Can I use both sipusers tables in Asterisk RealTime?
>
> Something like this:
> /etc/asterisk/extconfig.conf:
> [settings]
> sipusers => odbc,DBNAME1,sipusers
> sippeers => odbc,DBNAME1,sipusers
> sipusers => odbc,DBNAME2,sipusers
> sippeers => odbc,DBNAME2,sipusers
>
> If Asterisk 11 doesn't support this right now, will it in the future?
>
> Thanks,
>
> Vieri
>
>
> --
> _
> -- 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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[asterisk-users] Error to install Asterisk

[termo termosel]

Hi,
when I try to install Asterisk 11.2.1 the console return error which it tells:
/usr/bin/ld: final link failed: No space left on device
and the process exits installation.
How can I solve this problem? Tmp folder is empty.
Thanks,Jordi  --
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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   http://www.asterisk.org/hello

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[asterisk-users] multiple sipusers tables

[Vieri]

Hi,

I have 2 databases DBNAME1 and DBNAME2. In each database I have a table called 
sipusers (so DBNAME1.sipusers and DBNAME2.sipusers).

Can I use both sipusers tables in Asterisk RealTime?

Something like this:
/etc/asterisk/extconfig.conf:
[settings]
sipusers => odbc,DBNAME1,sipusers
sippeers => odbc,DBNAME1,sipusers
sipusers => odbc,DBNAME2,sipusers
sippeers => odbc,DBNAME2,sipusers

If Asterisk 11 doesn't support this right now, will it in the future?

Thanks,

Vieri


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