Re: [Asterisk-Users] Feedback request: AGI GET DATA change termination digits

[Freddi Hansen]

Hi, this is my 1.st response to this list, i hope this will work.

I tend to agree with Steven since just allowing other termination digits probaly wont 
solve your upcoming the issues anyway. I use a wrapper around the 'get digit' which 
allows me to specify that the * digit repeats the menu but maxium 3 times and if the * 
star digit is used twice in sequence (without other digits inbetween ) then it means 
'go to the menulevel above current level'.
This was todays 'noise' from me.
Freddi
Subject: Re: [Asterisk-Users] Feedback request: AGI GET DATA change -
termination digits
From: Steven Critchfield <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Date: Sat, 18 Oct 2003 19:14:14 -0500
Reply-To: [EMAIL PROTECTED]
While that change is fine, you could also just write the same
functionality with get digit and deal with it inside the AGI app.
On Sat, 2003-10-18 at 16:50, Paul Crick wrote:

** REPOST: A week later and no feedback - am I the only one
** who'd find this functionality useful? No other AGI stuff
** out there needing something similar?
I'd like some feedback on potentially submitting a request (and probably a
patch too) to change the way the AGI command GET DATA works.
Right now, # terminates the entry, which is then returned with the #
stripped off the end. What I'd like is to allow user configurable
termination digits, which are not stripped off the end.
Reasoning: Some entries you'd like to terminate with #. Right now it's fine,
you can tell if # was pressed or not by looking for the lack of a (timeout)
entry in the returned result. You may want to allow * to cancel an entry.
This is not possible right now. Systems I've coded previously allow # to
terminate and complete a digit entry, * to correct an incorrect entry
(playing the prompt again and restarting digit collection). Pressing  * with
no prior digit entry cancels the step and returns to the previous menu.
I guess there's a compatibility issue with stuff that's out there already
but if it was an optional 4th parameter this would be backwards compatible.
Proposed new syntax:
  GET DATA
If terminator is specified (and it may be multicharacter, like "*#" to give
me the functionality above), return the digit string collected so far,
including the terminating digit. The calling app can strip the trailing
character if needed.
Thoughts?

Cheers
Paul
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-- Steven Critchfield <[EMAIL PROTECTED]>

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Subject: Re: [Asterisk-Users] Where can i get the g.723 codec?

[Freddi Hansen]

I hope this doesn't show up twice (posted from wrong mail adr.)

Hi Eric,
You can actually get boards like this already from companies like 
Mapletree.
Its a hardware pci carrier card where you add the number of DSP modules 
that you need.
This hardware may be a bit 'high end' for most users on this list but 
several people seems to address this issue pretty often. One card may be 
equipped with dsp's to handle
488 simultanoeus sessions in any mix of supported codecs(incl. G.723.1 
and G729..).
I whish that I had the time to make the 'glue logic' thats needed to 
connect the
* codes api with Mapletree busmastering codecs channels.
P.S. I am NOT a Mapletree salesperson.

-The makers of hardphones prolly get their G72x licensing by using a DSP
-that already has a license.  The DSP can't be that expensive.  I wish
-someone would make a PCI card with something like 8 of these chips on it
-and sell it cheap.  Should be pretty easy to build a codec for Asterisk
-that uses the DSP card.
-On Mon, 2003-11-03 at 09:39, Gavin Hamill wrote:


On Mon, 2003-11-03 at 15:14, Eric Wieling wrote:



> Licensing info for the G723.1 codec, direct from the holding company
> that licenses the codec.
> > http://www.dspg.com/technology/LicensePricing.html
  

 >From what I remember when I looked into this about a year ago, this
 isn't even the end of it, since whilst DSPG represent /most/ of the IP
 holders on the codec, there are still others, and if you want to be
 completely sure of being legally in the clear, then you must reach
 seperate licensing arrangements with them
 If only some of the hard-phones would use Speex or similar, then all
 these problems would Go Away, and the production costs for the phones
 could drop, giving the manufr. the same amount of margin, but at a
  lower

 market cost.

 Cheers,
 Gavin.




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[Asterisk-Users] SoftFax question

[Freddi Hansen]

Hi,
I am looking at using the softfax that Steve Underwood has developed.
It's very straight forward when you assign an extension for the fax.
A function that several pbx's has is that they listen for the 'faxtone' 
for 5 seconds
after 'answer' in the menu where you can enter your local extension number,
it's normally done in parallel with the DTMF detection.  I think that 
the logical solution
would be if the DTMF mask given to the DTMFdetector could  had a digit  
for fax or
if there was a 'background' function that  we could check on with  
IfFaxGoTo(xxx).
I haven't been able to google any function in '*' that would help us 
with this so that's
why I try the list in case I (hopefully) have overlooked something.

The above function would be nice since you could share the same access 
number for phone
and fax (like the old autofax switches). Secondly when people mistakenly 
queues a fax for
you main access number it would just be dropped into the 'faxbox' 
instaedt of  calling you
10 times over the next 20 minutes.

Freddi

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[Asterisk-Users] Re: SoftFax question

[Freddi Hansen]

Thanks Steve and you other guys for your help.
I hope the quality of the other stuff I did yesterday is better
than my search for 'fax extension' (a bit embarrassed)
Freddi
On Wed, 12 Nov 2003, Freddi Hansen wrote:

Hi,
I am looking at using the softfax that Steve Underwood has developed.
It's very straight forward when you assign an extension for the fax.
A function that several pbx's has is that they listen for the 'faxtone'
for 5 seconds
after 'answer' in the menu where you can enter your local extension 
number,
it's normally done in parallel with the DTMF detection.  I think that
 



You want a fax extension:

exten=>fax,1,Blah()

A google for 'fax extension' turns up the announcement of this feature
here:
http://lists.digium.com/pipermail/asterisk-users/2002-October/005414.html
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Re: [Asterisk-Users] I hate to do this but..

[Freddi Hansen]

Date: Thu, 13 Nov 2003 07:19:59 -0500
From: Jeremy McNamara <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] I hate to do this but..
Reply-To: [EMAIL PROTECTED]
Josh Roberson wrote:


On Tue, 2002-11-26 at 14:52, Jamin W. Collins wrote:
   

   

Is there any current/planned support for Aculab hardware?

  http://www.aculab.com

Looks like they have Linux drivers and an SDK.
 

 

Has any advancement taken place in this?  Has someone developed a
working channel driver for this product?  I have one, and would like to
see if it would be a possibility to get working...
 

Flebay it and use the proceeds to support Digium by purchasing Zaptel 
hardware.  http://www.digium.com/



Jeremy McNamara

 

Hi ,
I hope that you aren't to quick to ditch  aculab project.  I dont see  
Aculab as an competitor to
Digium at all. Their cards does cost about 3 times as much as the TE410 
that I use now.
Why would anyone even consider using these cards then ?. I have myself 
been using them for
10+ years so I have a bit of experience with their cards. If it was 
possible to use an Aculab card
and a TE410 in the same server then we would be able to dump the last of 
our old servers and replace
them with '*' servers. The reason that we use the Aculab cards is 
because of their C7/SS7 ISUP support
which is a 'must' in several of our installations. In addition to this 
they do support a large number of legacy
R2 protocol and they do carry PTT approvals for using these protocols in 
almost any country.

I think that the card they have with  4 E1's / H323 and ethernet could 
be considered just as a MG seen from the '*'
side and that might be the most easy way to go if you dont want go all 
the way and make a channel driver.
My 1c on that issue.
Freddi







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[Asterisk-Users] Re: I hate to do this but..

[Freddi Hansen]

I think that the card they have with  4 E1's / H323 and ethernet could
be considered just as a MG seen from the '*'
side and that might be the most easy way to go if you dont want go all
the way and make a channel driver.
My 1c on that issue.
Freddi
The Aculab VoIP card actually only supports 2E1. Also, it is possible to use
an API to drive it to handle the RTP only, leaving SIP/h.323 whatever to the
application. However, what you can't do with the Aculab side of things (and
I have though about this myself) is use it as a replacement for the Digium
cards as there is no way to get the voice onto the PCI bus (even with
Prosody).
I'm afraid I would consider using these cards instead of the Digium ones
otherwise for two reasons:
1) Worldwide certification and approval
2) Worldwide protocol support.
However, they don't work and thats all there is to it!

Linus

I think that a simple/light integration should be possible. The Aculab 
acting as an endpoint should be able to register
at the '*' (allthough I haven't actually tried it). The voice will not 
be using either the S.100 or PCI bus,
it would be coming via the ethernet. You can think of it just like a MG 
that speaks SS7 against the PTT and
H323 again '*'.  The program that uses the Aculab API would just be a 
'dumb' bridging program.
The other benefit you get is that all the G723 and G729 to ulaw can be 
done on the embedded DSP on the board.

Freddi







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[Asterisk-Users] Re: I hate to do this but..

[Freddi Hansen]

Freddi Hansen wrote:

I'm afraid I would consider using these cards instead of the Digium ones
otherwise for two reasons:
1) Worldwide certification and approval
2) Worldwide protocol support.



October 02, 2003

Extra...Extra...Read all about it. Digium wants the whole world to know 
that the quad span T1/E1/PRI card, the TE410P, has passed Telecom FCC, 
Euro, and Australian certifications. This milestone in our hardware 
development gives the TE410P a significant boost for deploying 
applications worldwide and allows it to compete with similar cards in 
the market. We passed the rigorous certifications last week and the 
paperwork must be completed to make it official, but the hard work is 
done and we are excited to announce this great news to our customers. 
EMC testing is next.



Jeremy McNamara

Hi Jeremy,
I can see that name and comments doesn't match after your cut of the 
posting.
We are using the TE410P but what I was writing is that we are NOT 
considering to replace the TE410P
with boards from Aculab, we are replacing Aculab boards with Digium 
boards BUT we would need more
Digium boards IF we could use both Digium and Aculab cards in the same 
server. The reason being that
TE410P doesn't support SS7-ISUP so we continue using only Aculab cards 
in the servers that must support
SS7/ISUP.
Freddi.



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[Asterisk-Users] RE: Aculab SS7/ISUP (new subject)

[Freddi Hansen]

Freddi Hansen wrote:
with boards from Aculab, we are replacing Aculab boards with Digium 
boards BUT we would need more
Digium boards IF we could use both Digium and Aculab cards in the same 
server. The reason being that
TE410P doesn't support SS7-ISUP so we continue using only Aculab cards 
in the servers that must support
SS7/ISUP.
Do you use the Aculab SS7/ISUP together with Asterisk somehow?

/Olle
Not yet, I am still porting applications from our old properitary box to
the Asterisk base. There's a lot of stuff that work diffrent and I have to
add quite a bit of code to run a site that consist of an array of '*' servers.
Issues are like database replication, locating which server the SIP user is registered on
a.s.o. The old systems are build around Aculab cards and some of theses systems use
the SS7 but not in a mix with '*' yet. We did some analasys of the SS7 issue before we 
started on the '*' road. We decided that we need different approaches depending upon
the size of the '*' site. 

Big site (upto 16 servers) would use the Milborne box from Datakinetec in UK. It's a
signalling converter that does all the heavy SS7 stuff. Each server would still carry
2 TE410P cards. The channel driver would be rather simple since you get the signalling
over tcp (Q931) in the format like 'incoming call in port 5 timeslot 10'.
For smaller systems the Aculab card with 2 E1 lines and 2 ethernet interfaces could be a solution.
The last release from them that I used did allow you to define a CIC code map that
would span 2 extra TE410P cards and still let the Aculab handle the signalling.
This does ofcourse still require a new '*' channel driver but it wouldn't need to deal 
with the SS7 stack (only Q931).

I hope that I soon will get some time to play with this stuff again.
Freddi






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[Asterisk-Users] RE: Aculab SS7/ISUP

[Freddi Hansen]

On Thu, 2003-11-13 at 16:50, Freddi Hansen wrote:
 

>Freddi Hansen wrote:
 

>> with boards from Aculab, we are replacing Aculab boards with Digium 
>> boards BUT we would need more
>> Digium boards IF we could use both Digium and Aculab cards in the same 
>> server. The reason being that
>> TE410P doesn't support SS7-ISUP so we continue using only Aculab cards 
>> in the servers that must support
>> SS7/ISUP.
   

>
>Do you use the Aculab SS7/ISUP together with Asterisk somehow?
>
>/Olle
 

Not yet, I am still porting applications from our old properitary box to
the Asterisk base. There's a lot of stuff that work diffrent and I have to
add quite a bit of code to run a site that consist of an array of '*' servers.
Issues are like database replication, locating which server the SIP user is registered on
a.s.o. The old systems are build around Aculab cards and some of theses systems use
the SS7 but not in a mix with '*' yet. We did some analasys of the SS7 issue before we 
started on the '*' road. We decided that we need different approaches depending upon
the size of the '*' site. 

Big site (upto 16 servers) would use the Milborne box from Datakinetec in UK. It's a
signalling converter that does all the heavy SS7 stuff. Each server would still carry
2 TE410P cards. The channel driver would be rather simple since you get the signalling
over tcp (Q931) in the format like 'incoming call in port 5 timeslot 10'.
For smaller systems the Aculab card with 2 E1 lines and 2 ethernet interfaces could be a solution.
The last release from them that I used did allow you to define a CIC code map that
would span 2 extra TE410P cards and still let the Aculab handle the signalling.
This does ofcourse still require a new '*' channel driver but it wouldn't need to deal 
with the SS7 stack (only Q931).

I hope that I soon will get some time to play with this stuff again.
Freddi
   

Instead of bothering the channel driver, sounds like you just need to
modify libpri then to handle the Q931 coming from a source other than a
zap channel. Would be much simpler I hope to handle it this way than to
write a special driver.
-- Steven Critchfield
I realized by re-reading my mail that it can mislead a bit. The Q931 
statemachine is embedded on the Aculab card and '*'
interface will have to be through the standard Aculab API  which  more 
or less sits on top of a Q931. I am still a newbie
to '*'  so I am not ready to take a headdive into that project yet.
b.r.
Freddi





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RE: [Asterisk-Users] Port density: DS3 cards?

[Freddi Hansen]

I have no reason to disbelieve this report, but I will offer some 
minor scepticism at this reply.  A well-equipped PC can currently 
handle 8 T1 channels, and it seems that only the IRQ issue is causing 
more channels to not be viable in the current TE410P environment.  It 
would seem reasonable to think that a very well equipped PC (4-way, 
8-way?) would be able to handle the "processing power" requirements 
of a DS3, whatever was meant by that statement.  Of course, there may 
be other underlying issues specific to ImageStream that make this 
impossible; I don't know.

JT

Try to look this building block that should allow you to do T3 to TDMoE 
at wirespeed.
The chip can move data between T3 and Ethernet without touching the PCI bus
but you can still keep full control via the PCI bus if you want to.

http://products.zarlink.com/product_profiles/ZL50111.htm

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RE : [Asterisk-Users] 2nd call leg status?

[Freddi Hansen]

Hi Folks,

Wonder whether this question found an answer ?

I too have a similar question that I can't find an answer so far.

Let me first share my dial plan;

exten => _011.,1,Authenticate(/etc/asterisk/auth.txt |a)
exten => _011.,2,Playback(Pls-wait-while-I-connect)
exten => _011.,3,Absolutetimeout(3600)
exten => _011.,4,Dial(H323/${EXTEN:[EMAIL PROTECTED],70)
exten => _011.,5,NoCDR()  <=if no answer cdr is not written
exten => _011.,6,Busy
exten => _011.,105,NoCDR()  < if called party is busy cdr not wirtten
exten => _011.,106,Busy
exten => i,1,NoCDR() <== if authentication failed cdr not written
exten => i,2,Hangup
Here are my observations

(a) Since Authenticate function is present in my dial paln,  disposition
fieled in cdr always show Answered, so with that I can't figure out 
whether
H323 leg is successfully answered or not.
(b) If the H323 g/w sends the busy signal then CDR is not written, If the
g/w rings and timed out then again CDR is not written (as expected we 
have
priorities set for extensions)
(c) Now if the called party is ringing and originating party just 
hang-up, A
CDR is written. I have no way to differentiate that with a very short 
answer
call.

I think this behavior is a incorrect.

If * answered a call and show up disposition as Answered, then that 
call is
a completed call. So there should be one record in the CDR.

Then for the second leg there should be another record, as now * is
originating a call again and expecting other side(in this case the 
h323 g/w)
to respond.

If the both legs are considered as a single call, then cdr should show 
the
disposition of the final end point.

Please show me if there is a way that I could generate two records in CDR
for this kind of a call, or any other solution for this problem.
Cheers

SW


Hi,
I did have the same problem. I simply issue a ResetCDR(w) as the last 
thing before using
the Dial application. This will reset the 'answered' flag and the CDR 
you get from the Dial
will contain the correct value. I do later do a small of backend 
processing to get the correct
A-line time related to the call.
There may be smarter way of doing this but this quick hack works fine 
for me
Freddi





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re: [Asterisk-Users] help a poor newbie out with SIP choppy one-way problem

[Freddi Hansen]

From: "yair hakak" <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Date: Thu, 19 Feb 2004 07:54:00 +
Subject: [Asterisk-Users] help a poor newbie out with SIP choppy one-way problem
Reply-To: [EMAIL PROTECTED]
Hello all,
i have a one-way choppy sound problem that i can't fix...
here are the relevant points
1. i am running 18.2.04 CVS on rhl 9.0 on a hosted server with a wide pipe 
up/down with no hardware, just SIP connections and voicepulse for outgoing 
IAX calls.
2. conecting to * with SJPhone (SIP) on a windows box that gets 1.5MB down 
and about 100K upload in speed tests (ADSL), so i'm pretty sure client 
bandwidth is not a problem either. the client can ping the server at 
180-200ms as well.  I've also tried x-lite and gotten the same issues.

sip clients register fine, and i can hear incoming audio fine, but on the 
other end it is completely garbled. It is not an IAX problem; if i leave 
voicemail from the SIP client on * and try to pick it up it is garbled, but 
the voicemail prompts are crystal clear.

there was a thread about this at the beginning of january - the only 
solution that came up was to sweep the windows box for worms - which i did, 
and i have no worms.  if anyone who had the problem then has answers, or 
anyone else, i would be most grateful.

thanks,
yair
Try to set the following in your x-lite config.
I had one-way choppy sound and this was the cure.
AdvancedSystemSetting->AudioSettings->SilenceSettings->TransmitSilence:yes

Freddi

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[Asterisk-Users] RE: Asterisk crashed so often

[Freddi Hansen]

From: "" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Date: Thu, 4 Mar 2004 16:33:12 -0800
Subject: [Asterisk-Users] Asterisk crashed so often
Reply-To: [EMAIL PROTECTED]
This is a multi-part message in MIME format.

--=_NextPart_000_0250_01C40206.62A69480
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
FYI that I have experience the same problem many times.  The system is =
running RH 9.0 with Asterisk CVS-02/21/04.  Here is the output from the =
console:
I am using RH 9.0 too.
If I forget to do the:
export LD_ASSUME_KERNEL=2.4.1
before asterisk is started then it will handle less than 10 calls before it crashes.
Otherwise it seems rockstable in our enviroment with SIP and zaptel TP410P's when it's started 
Freddi
 



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Re: [asterisk-users] Asterisk on multi-homed systems

[Freddi Hansen]

We are using Ethernet Bonding with no problems at all. Each server has 2 
build-in NIC's and a quad NIC. They are divided into 3 networks with 2 
NIC's in each. Links are up on all 6 connections and you don't even hear 
a click if I unplug the 'live' ethernet. 3 different networks on 3 or 
more NIC's works fine.
The thing that doen't work is when you assigned multiple IP aliases onto 
a single NIC, Asterisk always choses the lowest/first IP address 
assigned to the card. You can set the 'fromdomain' in the peer that you 
want to force onto the second IP on the NIC but the source address will 
still be the first IP on that NIC.
I think that what we are missing here is a peer parameter where we can 
force the source address.

Freddi

> If I was wanted to multi-home on the same subnet I would use Ethernet 
> Bonding (similar to Windows Teaming) in a failover configuration.  
> This will make one of the links on the LAN active and the second one 
> as a failover in case the first one goes down.  It takes a couple 
> seconds for the 2nd link to come up.  I am not using this in Asterisk 
> at the moment, but I am using it on other servers and it works great.  
> I don't know if this would drop a call during failover, but it's 
> something to explore.
>  
>   Shlomo
>
>  
> On 12/1/07, *Steven* <[EMAIL PROTECTED] 
> > wrote: I have zero issues with 
> multihomed asterisks.
>
> One potential issue is that some people are multihoming onto the same 
> subnet.
> This will cause issues with many applications as normal routing 
> usually sends data OUT the lower IP address if there are two on the
> same subnet.
>
> Multihoming, as a rule should be on separate network.
>
> My company's implementation is one three networks.
> One inside, One to ISP A and one to ISP B.
>
> Like I said, I have had zero issues.
>
>
>
> --
> --
> Steven
>
> http://www.connectech.org/
>
>
>
> "Chris Bagnall" < [EMAIL PROTECTED] > wrote 
> in message news:[EMAIL PROTECTED]
> > Greetings list,
> >
> > I remember a discussion many months ago which ISTR concluded that 
> asterisk didn't play nicely at all in multi-homed setups ( e.g.
> > SIP packets not being sent out through the same interface they were 
> received on, etc.).
> >
> > Is this still the case, or are there versions which have resolved 
> the issue? Even if it's still the case, is this only a problem
> > for SIP, or does it affect asterisk in general?
> >
> > I have a number of servers with dual NICs, each with an independent 
> net connection. After a few recent failures with one provider,
> > it'd be very useful to be able to use the other connection 
> simultaneously, but only if it's not going to cause problems with the
> > rest of the setup.
> >
> > Any suggestions gratefully appreciated.
> >
> > Regards,
> >
> > Chris
> > --
> > C.M. Bagnall, Director, Minotaur I.T. Limited
> > For full contact details visit http://www.minotaur.it
> > This email is made from 100% recycled electrons
> >
> >
> >
> >
> >
> > ___
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> >
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> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >


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[asterisk-users] Zaptel compile error after make update.

[Freddi Hansen]

Hi,
I started to get some Zaptel compile errors after a 'make update'

I did a clean zaptel install with:

svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel

I am still getting the error, is this someelse seeing this ?.

CC [M]  /usr/src/zaptel/kernel/zaptel-base.o
/usr/src/zaptel/kernel/zaptel-base.c: In function 'zt_reallocbufs':
/usr/src/zaptel/kernel/zaptel-base.c:889: error: 'struct zt_chan' has no 
member named 'rebufpolicy'
make[3]: *** [/usr/src/zaptel/kernel/zaptel-base.o] Error 1
make[2]: *** [_module_/usr/src/zaptel/kernel] Error 2
make[2]: Leaving directory `/usr/src/kernels/2.6.23.15-80.fc7-i686'
make[1]: *** [modules] Error 2
make[1]: Leaving directory `/usr/src/zaptel'
make: *** [all] Error 2

It's a FC7 and the Zaptel cards is a TE410P

Freddi


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Re: [asterisk-users] [Asterisk-users] asterisk +heartbeat (Wilton Helm)

[Freddi Hansen]

>
>> having two NICs on the same subnet
>
> I'm trying to wrap my brain around that in the larger network 
> picture.  Two
> NICs in the same subnet (presumably on the same computer) would have 
> access
> to the same other devices.  This could potentially increase bandwidth
> (maybe?) and offer redundancy (if NICS, wiring or switches were the 
> biggest
> source of failure).  I'm not sure how the OS would decide which one to 
> use
> for a given packet, or if an application (such as Asterisk) could 
> determine
> which one to use.  I can see potential problems with addressing, as other
> devices could send to one, and would definitely not know what to do 
> with a
> reply from the other, etc.  I'm not sure this would be an Asterisk bug.
> Without some concept of what I am missing here, I would consider it a
> cockpit error on system setup.  The only reason I can think of for having
> two NICs in a computer would be using it as a router--in which case they
> wouldn't be on the same subnet.  (OK I've done it before for redundant
> paths, but again, the paths should be on different subnets, otherwise how
> does one tell the OS which path was intended?)

Try reading:
http://www.linuxfoundation.org/en/Net:Bonding
We have 3 networks on each of our servers. Each network (and IP) is 
served by 2 nics. (yes 6 nics per server)
Works well with Asterisk, you can disconnect cables or take power from 
one of the core switches without as much as a click  in audio in ongoing 
connections.
 



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Re: [asterisk-users] 1.6 Production ready??

[Freddi Hansen]

Not as impressive as matthew's ref but just to add to the picture.

System uptime: 17 weeks, 7 hours, 30 minutes, 51 seconds
342277 calls processed

Asterisk SVN-branch-1.6.0-r117951 built by root @ localhost.localdomain 
on a i686 running Linux on 2008-05-22 21:13:46 UTC

using and old Dell 1750 with a quad E1 (TE410)  on a Fedora7

Mainly SS7<=>SIP ,AGI.

Freddi.
> Steve Totaro wrote:
>   
>> > 
>> > 
>> > On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson <[EMAIL PROTECTED] 
>> > > wrote:
>> > 
>> > Sebastian Gutierrez wrote:
>> >  > Anyone is using 1.6 in production??
>> >  >
>> >  > Is it ready?
>> > 
>> > I have a number of people using 1.6 in production doing SS7<->SIP,
>> > SS7<->IAX, and SS7<->ISDN gatewaying.
>> > 
>> > One example (doing SS7<->IAX):
>> > 
>> > System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds
>> > 
>> > 8617029 calls processed
>> > 
>> > ---
>> > Matthew Fredrickson
>> > Digium, Inc.
>> > 
>> > 


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Re: [asterisk-users] CDR Desgin

[Freddi Hansen]

>
> To me the obvious answer is to provide a CDR for every call leg so for
> > A calling B via Asterisk there would be two CDRs produced. It's far
> > far easier to disregard the unwanted CDRs than it is to try and
> > generate the missing ones and in some cases it's virtually impossible.
> > If it's weighed up I think people would vote to have accurate CDRs
> > ahead of anything else and if single legs are the best way to do that
> > then it's the way it should be done.
> > 
> > In addition with single leg CDRs it will solve the dilemna about
> > acommodating every possible call scenario that I know has caused you a
> > lot of consternation over the last 18 months.
> > 
> > Sure it's a change from the current situation so maybe needs to use
> > the standard apporach of a configuration setting to opt in initially
> > before becoming the default in the subsequent major release.
>   
>
>
> OK, Greyman, your logic is solid. If we provide a CDR implementation
> that just generates the individual call legs, and ties them together via
> the linkedid
> (see team/group/newcdr), then both camps should be able to derive the
> info
> they need for billing, via hopefully not-overly-complex SQL queries to a
> backend db.
>
> I'll modify my RFC to reflect this line of thinking. Yes, it is a bit of
> shift.
> And, yes, the implementation will make this new approach optional, and
> not
> default. But, pardon if I make it available via the CEL domain come
> implementation time.
>
>
> It should take me a week to rehash my document; perhaps longer (I'm in
> bugfix mode, and this borderline development work); in the meantime,
> those with decided CDR needs might make their wishes known, if they do
> not think this approach will work. Speak now, or forever hold your
> peace; or forever complain... or whatever.
> If this is particularly distressing to you, perhaps your fears might be
> slightly assuaged when you read the details...
>   
I was part of a team that did design a multiservice billing system about 
15 years ago and the solution people seems to agree on here (and me to) 
looks pretty much the same i.e one call may consist of several calls 
legs. In addition to the linkedid it would be nice to have an indication 
in the cdr that tells us that 'this is the lastone on this  linked id'.
Our experience was that  we shouldn't  for load reasons work with cdr's 
in the immidiate multileg format in the DB. So we did collect cdr's in a 
tmp DB until we got the the record with end marker set. We would then 
produce our final cdr for the actual service, store it in billing col. 
and delete it from the multileg col. When a new service is created we 
only have to make a the new customized cdr, we don't have to touch the 
generation of the multileg format.  

Freddi




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Re: [asterisk-users] CDR Design

[Freddi Hansen]

> On Dec 1, 2008, at 9:07 AM, JD wrote:
>
>> Steve Murphy wrote:
>>>
>>> Freddi--
>>>
>>> Very interesting. Brian Degenhardt had some code we just gave some
>>> thought
>>> to, wherein we determine if the last channel involved in a linkedID set
>>> has been closed. If so, then the entire set is finished. We can use 
>>> this
>>> facility to get you a closing attribute, that could be added to the 
>>> last
>>> CDR emmitted for that set; OR, we could just emit another CDR with type
>>> CLOSE or FINAL or something, that signals the end of the chain.
>>>
>>> murf
>>>
>> Just thinking out loud: how about a feature wherein, after the FINAL is
>> sent, asterisk can
>>  1. create a temp text file with just those entries, and
>>  2. launch a user-made script.
>>
>> cdr_manager.conf
>>  [general]
>>  legparsecmd=/usr/local/bin/my_parser.pl
>>
>> wherein the linkedID is passed as the first parameter and the text file
>> name&path as the second
>>
>> Ignore this suggestion if it horribly complicates things.
>
> Hmm.. While I normally like having this kind of "instant 
> notification", I could see this as a very big problem for larger 
> installations.  Most OS's are not so great at launching new tasks, and 
> on a heavily loaded system that could easily be a number of tasks 
> launched every second, each doing a lot of database queries.  Perhaps 
> a different approach would be to have a field that can be set to show 
> that the record(s) have been parsed into whatever standard CDR format 
> you want.  This may or may not make more sense as a separate table 
> with just a list of linkedid's that have been parsed.
>
> Daniel
I think that e.g. a socket would be preferable.  You do not have the 
load of launching new a task for each cdr and the process listening on 
the socket would normally run at lower priority than Astrisk itself. If 
gives the freedom to send unprocessed 'leg-information' to a cdr backend 
process on another server if  heavy loaded enviroment or just process it 
local if it's a small installation.
To me it looks new cdr spec from murf would be a big step forward for 
Asterisk.

Freddi
   



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Re: [asterisk-users] CDR Design

[Freddi Hansen]

I agree with  [EMAIL PROTECTED] we need the events to create the final CDR.
I will not waste list space on a long but just show you 2 reallife 
examples that can't be handled both within the same 'fixed' way of 
generating CDR's as we do now.The new system that 's proposed would 
handle both  just with trimming  of the config file.
A:
Asterisk used on a shared premium number where we pay the companies 
that's chosen in the menu.
The Company chosen can make  internal transfers, the point is that  
A-line time  for the whole session
is 'billable'.
B.
Voip service provider that allows transfers.
B-line time is billable - A pays for leg A to B, when a transfer occurs 
then the leg from B to C must be paid by B

It's difficult to impossible to maintain that kind of logic in the 
Asterisk core.
Asking for 'one' CDR is essentially the same as asking for hardcoded 
cdr-event logic in the core -
I would certainly prefer the way that Murf is proposing.  

If we use Dbus,socket or spawn external program (like with agi) that's 
just an implementation detail but the architecture should be right.

Freddi


> Billing and logging should not be confused theoretically - I agree. 
> But in practice,
> the logging of the calls (not other events of the system) IS used for 
> billing purposes.
> The start and finish time is not enough for many (I not that it is not 
> enough for me).
>
> The accountcode is not enough for me either. From my CDRs I have to 
> extract all the
> information about which provider tried-and-failed or 
> tried-and-succeeded to terminate
> the call. So I need the terminator's info in the CDRs. This is the 
> only way that I can monitor
> what my providers charge me (and believe me, never take for granted 
> that your provider charge
> you with pre-agreed rates, mistakes happen :)). Also, having the 
> terminator's data in the CDR is
> the only way that I can calculate metrics such as ASR, ACD, mean PDD etc.
> And I can't imagine taking all this info from a logging module that 
> mixes CDR log events with
> other ones (hardware events, user agent registrations, etc.)
>
> Since there is no agreement on WHAT to log and since we have the 
> option to put a lot of info
> in the CDRs I think the right way to do it is provide the capability 
> of every single detail that COULD
> be logged and let the end user choose WHAT to log through the 
> configuration. I cannot understand
> tha benefit of a minimal/fixed/non-flexible CDR logging capability 
> when can have the flexibility to
> go from minimal to complex depending on a configuration entry in a 
> proper configuration file.
>
> P.S. Sometimes I wonder if I am the only one in the VoIP world that 
> finds terminator information in the
> CDRs useful (including failed calls).
>
> P.S. Sometime we use the term "billing" only for customer billing 
> processes which nowadays is incorrect
> or insufficient. "Billing" in today's demanding VoIP business means :
>
> 1. Customer Billing : we all know what that is
>
> 2. Provider CDRs cross-check : as I said above, you have to know what 
> your provider charges you in order
> to catch mistakes and in order to able to produce profit/loss reports.
>
> 3. QoS metrics : ASR, ACD, PDD to name a few. These cannot be 
> calculated without proper termination info
> from the CDRs. I see LCR modules being introduced now and then in the 
> asterisk community but they all seem
> a little useless if the above metrics cannot be extracted from the 
> CDRs. What is the benefit of having a low cost provider
> in your LCR if its ASR equals to 0.0001 %? and how can you measure its 
> ASR if the terminator's info (both failed and successful)
> is not in the CDRs?
>
>
> Andrew Thomas wrote:
> It seems to me that we are confusing billing and logging here.  Call
> billing only really needs the start and finish (like we get now) - but
> proper call logging requires all steps.
>
> Do we leave CDR's as they are (for billing purposes) and have a separate
> 'event' driven log for call logging?  Or do we change the CDR structure
> to accommodate logging as well?
>
> Personally, a separate 'event' log seems preferable to me as this keeps
> existing billing platforms useable.  It just means the logging programs
> will need to be re-written to look at a new database for events.
>
> I know we have the AMI - but that puts out a lot more information than
> is needed for simple logging (and requires something to prune and store
> the events in a database of some sort).
>
> Any thoughts? 
> 
> Andrew Thomas
> Technical Services Manager
> DataVox Ltd
> Saddleworth Business Centre
> Huddersfield Road
> Delph, Oldham
> OL3 5DF
>

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Re: [asterisk-users] CDR Design

[Freddi Hansen]

>   I agree with the fact that the base is broken and needs to be fixed 
> first.
>
> -- 
We wouldn't have this discussion if we had a close to perfect CDR design 
that just needed some 'fixing'.
The processes of just adding another couple of patches has been ongoing 
for more than year now.
I think that phase 1 should be creation of the new CDR's according  to 
Steve's spec.
A phase 2 could be an addon to CDR module or external script that would 
create a CDR record exactly as the old CDR record so we maintain 
backward compatibility with peoples existing billing systems that run on 
CDR's.
Imagine that the existing CDR module collect  the events  as the are 
generated and then when it would create the CDR as it does now it runs 
the config controlled interpreter that convert the eventlist to the old 
CDR records. For simple Asterisk usage it would stil work 
'out-of-the-box' with existing callingcard billing a.s.o.
So for those that 'just' want simple CDR's this change wouldn't change 
anything as long as they don't lift the hood.

The benefit would be that all event generation would be decoupled from 
the business logic thats in place for CDR generation and users may have 
control over that business logic.
Using these events for 'realtime' stuff is anther spinoff but not the 
primary reason

my 2 cent.
Freddi 


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Re: [asterisk-users] Latest AstManProxy

[Freddi Hansen]

> Hi,
>
> I unsuccessfully tried to download AstManProxy, clicking over download 
> button in http://github.com/davetroy/astmanproxy/tree/master .
> It failed with "XML error".

I guess I have to insert here not to get caught by top or bottom post  
filters.

You might want to use the version at:

http://github.com/davies147/astmanproxy/tree/master

it's updated and an error that can cause segfault when client 
disconnects has been fixed.

Freddi.

>
> How can you download AstManProxy ?
> Has the project moved to somewhere else ?
> Have its features been deprecated and replaced by something embedded 
> in Asterisk code or elsewhere ?

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[asterisk-users] zaptel errors update

[Freddi Hansen]

2 of our ISDN lines attached to a te410 card (only 2 in use on that 
card) both started to give 'reject frame errors'  after an zaptel 1.4 
update.

prev. i was using a version about 2 month old - 'make update' updated it 
to rev '4301'.

I contacted our telco and they say line is clean, not a single glitch 
within last 48 hours.

When I get the: ERROR[2834]: chan_zap.c:8248 zt_pri_error: !! Got reject 
for frame 14, retransmitting frame 14 now, updating n_r!

It is always when incoming call on ISDN hangs up. (calls going ISDN -> IAX).

Anyone else seing this ?

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[asterisk-users] Error after upgrading from 1.2.18 to 1.4.20

[Freddi Hansen]

scenario is incoming calls on ZAP (TE410p  euro isdn) to SIP (or any 
other channel) and call is answered.

When I hangup on the ISDN side on the 1.2 then the SIP hangs up to 
immidiatly so everything is fine (se short pri debug below).

When I do the same on 1.4.20 then it take more than 30 seconds to 
disconnect when the ISDN calling party hangs up

I have tried to use the use the 1.4 zaptel with the 1.2 installation, 
that works perfect to so to me it looks like a libpri or chan_zap problem.
I do hope though that some has a hint to where I scre... up.

Freddi


OK trace from 1.2

< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 7603/0x1DB3) (Originator)
< Message type: DISCONNECT (69)
< [08 02 80 90]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: User (0)
<  Ext: 1  Cause: Normal Clearing (16), class = Normal 
Event (1) ]
< [1e 02 82 88]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 
0: 0   Location: Public network serving the local user (2)
<   Ext: 1  Progress Description: Inband 
information or appropriate pattern now available. (8) ]
-- Processing IE 8 (cs0, Cause)
-- Processing IE 30 (cs0, Progress Indicator)
-- Channel 0/14, span 1 got hangup request
  == Spawn extension (Pstn-incoming-fwd, 77348855, 2) exited non-zero on 
'Zap/14-1'
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, 
peerstate Disconnect Request
 > Protocol Discriminator: Q.931 (8)  len=9
 > Call Ref: len= 2 (reference 7603/0x1DB3) (Terminator)
 > Message type: RELEASE (77)
 > [08 02 81 90]
 > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: Private network serving the local user (1)
 >  Ext: 1  Cause: Normal Clearing (16), class = Normal 
Event (1) ]
-- Hungup 'Zap/14-1'
< Protocol Discriminator: Q.931 (8)  len=5
< Call Ref: len= 2 (reference 7603/0x1DB3) (Originator)
< Message type: RELEASE COMPLETE (90)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
fonet2*CLI

BAD TRACE from 1.4  below:

fonet3*CLI>
< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 7582/0x1D9E) (Originator)
< Message type: DISCONNECT (69)
< [08 02 80 90]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  
Location: User (0)
<  Ext: 1  Cause: Normal Clearing (16), class = Normal 
Event (1) ]
< [1e 02 82 88]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard 
(0)  0: 0  Location: Public network serving the local user (2)
<   Ext: 1  Progress Description: Inband 
information or appropriate pattern now available. (8) ]
-- Processing IE 8 (cs0, Cause)
-- Processing IE 30 (cs0, Progress Indicator)
q931.c:3779 q931_receive: call 7582 on channel 13 enters state 12 
(Disconnect Indication)
fonet3*CLI>

~ 30 SECOND DELAY HERE.

fonet3*CLI>
fonet3*CLI>
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 7582/0x1D9E) (Originator)
< Message type: RELEASE (77)
< [08 02 80 90]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  
Location: User (0)
<  Ext: 1  Cause: Normal Clearing (16), class = Normal 
Event (1) ]
-- Processing IE 8 (cs0, Cause)
q931.c:3754 q931_receive: call 7582 on channel 13 enters state 0 (Null)
-- Channel 0/13, span 2 got hangup, cause 16
  == Spawn extension (Pstn-incoming-fwd, 77348855, 2) exited non-zero on 
'Zap/44-1'
-- Executing [EMAIL PROTECTED]:1] Hangup("Zap/44-1", "") in new stack
  == Spawn extension (Pstn-incoming-fwd, h, 1) exited non-zero on 'Zap/44-1'
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release 
Request
 > Protocol Discriminator: Q.931 (8)  len=9
 > Call Ref: len= 2 (reference 7582/0x1D9E) (Terminator)
 > Message type: RELEASE COMPLETE (90)
 > [08 02 81 90]
 > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  
Location: Private network serving the local user (1)
 >  Ext: 1  Cause: Normal Clearing (16), class = Normal 
Event (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/44-1'fonet3*CLI>






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Re: [asterisk-users] Error after upgrading from 1.2.18 to 1.4.20

[Freddi Hansen]

Hi I better answer my own post.

I went to the code and the issue is in q931.c

/* wait for a RELEASE so that sufficient time has passed
for the inband audio to be heard */
   
  if (c->progressmask & PRI_PROG_INBAND_AVAILABLE)
break;
  
Changing this line to a comment makes the 1.4 work exactly as 1.2 for 
this issue.
I think that this line should only be executed on 'outbound' pri calls, 
not on inbound.

Freddi
 


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[asterisk-users] Error after svn co of lastest zaptel 1.4

[Freddi Hansen]

Hi,
I got some errors about not being able to create subdir [already 
existing] on a 'make update' in  my zaptel 1.4.
I removed the directory and  did a new  svn co  of zaptel 1.4
[ svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel ]

now I get:

/usr/bin/install -c -D -m 644 tonezone.h /usr/include/zaptel/tonezone.h
make -C firmware hotplug-install DESTDIR= HOTPLUG_FIRMWARE=yes
make[1]: Entering directory `/usr/src/zaptel/firmware'
make[1]: *** No rule to make target 
`zaptel-fw-oct6114-064-1.05.01.tar.gz', needed by `hotplug-install'.  Stop.
make[1]: Leaving directory `/usr/src/zaptel/firmware'
make: *** [install-firmware] Error 2

OS:  FC7
zaptel card: TE405P

Freddi
 

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Re: [asterisk-users] Error after svn co of lastest zaptel 1.4

[Freddi Hansen]

>
> Hi,
> I got some errors about not being able to create subdir [already 
> existing] on a 'make update' in  my zaptel 1.4.
> I removed the directory and  did a new  svn co  of zaptel 1.4
> [ svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel ]
>
> now I get:
> 
> /usr/bin/install -c -D -m 644 tonezone.h /usr/include/zaptel/tonezone.h
> make -C firmware hotplug-install DESTDIR= HOTPLUG_FIRMWARE=yes
> make[1]: Entering directory `/usr/src/zaptel/firmware'
> make[1]: *** No rule to make target 
> `zaptel-fw-oct6114-064-1.05.01.tar.gz', needed by `hotplug-install'.  Stop.
> make[1]: Leaving directory `/usr/src/zaptel/firmware'
> make: *** [install-firmware] Error 2
>
> OS:  FC7
> zaptel card: TE405P
>
> Freddi
I copied the firmwarew files from my old zaptel dir and it works now.
It looks though  as if they are missing after  a new/clean checkout.

Freddi

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Re: [asterisk-users] Reliable wireless SIP phones (Tzafrir Cohen)

[Freddi Hansen]

  The Siemens DECT line are open source. Not broadly available in the US
> though.
  
  
What does this mean? Could you please provide a link for more
information?
  


here's a link to the Siemens open source dect/wifi phones if that's
what you are looking for.

http://gigaset.siemens.com/shc/0,1935,hq_en_0_121782_rArNrNrNrN,00.html

Freddi



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[Asterisk-Users] RE: H.323 - NO AUDIO IN BOTH DIRECTIONS

[Freddi Hansen]

Same problem here.
We have 2 identical servers on same site using chan_323.
CVS was from medio mar. on both servers and audio was fine.
I did upgrade to cvs head yesterday on one server and lost h323 audio on 
that server.
We do make outbound calls on the chan_h323
Freddi

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[Asterisk-Users] RE: NO AUDIO IN BOTH DIRECTIONS

[Freddi Hansen]

lets see if we help Jeremy (and ourselves) to narrow down the timeframe 
when this problem startet.
I have the following release running with the recommended pw/openh323 libs.
Audio is working fine and I use faststart (must).
"Asterisk CVS-04/13/04-22:41:25"
Does anyone have a newer release running that works with audio?.
I had one myself but lost due to HD crash I think it was from May-20 but 
I dont have the exact date.
Freddi

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[Asterisk-Users] RE: H.323 Audio problem UPDATE

[Freddi Hansen]

I have (as I have mentioned before) 2 identical servers connected to to 
same cisco gatekeeper.
Server 1 works fine with no audio problems, server 2 is using cvs head 
and there is no audio when connected.
using same configs on both servers (RH9).  Disabling faststart didn't 
help me.
I have spent some time plugging in exstra debug statements and comparing 
the 2 servers. Here is one thing I find
a bit strange about the the non working server and its easy to 
reproduce.  I think that my no-working server would be
working if my gatekeeper was supporting GSM which it doesn't so I cannot 
verify my claim here.

In h323.conf:
disallow all
allow alaw.
start  '*'
h.323 show codecs
Allowed Codecs:
Table:
  G.711-ALaw-64k{sw} <1>
Set:
  0:
0:
  G.711-ALaw-64k{sw} <1>
which is ok afaik
make 1 call (which passes no audio)
h.323 show codecs
Allowed Codecs:
Table:
   (empty now.)
the  endPoint->GetCapabilities();   returns me an empty string now.
The only codec that 'survives' is for what ever reason the gsm codec.
I will continue to see if I can pinpoint this issue. (I hope that I am 
not of on some wild goose chase).

Freddi




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[Asterisk-Users] RE: Layer 3 VPN Question

[Freddi Hansen]

Hi,
Please keep this discussion on-list. I did search the list 3 weeks ago 
on not much usable did show up.

Here is my scenario, fyi.
I have sip/iax phones registered on my * server.
My ISP can also do A-Z termination and provide local did numbers and 
controls Qos via VLAN/CoS.
If I use the right VLAN tag and CoS then traffic is prioritized over my 
normal internet traffic.
I did buy a small switch/router (Micronet sp1678) which can set the VLAN 
tags and supports CoS per port
but I still feel the solution is a bit clumsy. I would rather use my 
Linux (RH9) to do the same. I am pretty sure it can
but I haven't had any time yet to investigate how, so let the list 
benefit from configuration experiences in this area.

Freddi

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[Asterisk-Users] re: Asterisk CDR & UniqueID

[Freddi Hansen]

Hey All,
We are running a small SIP/IAX termination service at the moment
(planning on growing it) with 2 asterisk machines. One terminates the
SIP/IAX calls from our customers and one is our gateway to our upstream
provider. Both machines are logging CDR data to the same postgres table
using the cdr_psql module.
The problem I am having is I'm trying to work out how to link the CDR
records into a single 'call stream', rather than having separate records
per machine the call passes through.
Reading around on the Wiki and doing a bit of googling, I've worked out
that what I'm trying to do is the "normalization" step of "CDR
mediation", but I have not been able to find out any specifics about how
to go about it.
I would have thought that the originating asterisk machine would
generate the UniqueID for the call (Message-ID in SMTP terms) and pass
that along the call path, but each machine is using the epoch timestamp
of when it sees (or records, not sure which) of the call.
I know I can use the NoCDR app on the first machine in the chain, but
that does not scale if we need to add more machines to our network.
Does:
a) anyone have any idea of what I'm trying to explain, and
b) have any pointers of where I can find more information about doing 
this?

Thanks
Darryl 

Hi,
I had the same problem. One solution is to include the ip-adr or uname 
of the gateway that serves the call, then you can have a true unique-id.
I did patch cdrcsv.c to include an exstra field which is the uname of my 
* server. Another possible way today is to use the 'SetCDRUserField'
to let the 'uname' into your  cdrstream.  For DB scaling issues I would 
recommend that you to use a 'global unique' table to convert your uname
to an integer so you dont get something like a char(32) in your DB key.
Freddi

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[Asterisk-Users] Re: Auto Protocol (depending upon registration....

[Freddi Hansen]

Subject:
[Asterisk-Users] Auto Protocol (depending upon registration
From:
"Gary" <[EMAIL PROTECTED]>
Date:
Tue, 18 Jan 2005 17:06:08 +1000
To:
"asterisk-users@lists.digium.com" 
Hi folks,
I'm sure I had this in a previous life 

Basically the ability to dial with autoselection of either IAX2 or SIP
depending upon the registration of the endpoint.
Ok, I have probably missed it in the wiki as well.
hints ?
Gary
 

Use ChanIsAvail(SIP/mylogin&IAX2/mylogin), and then Dial(${AVAILCHAN})
eventually use a macro.
Freddi
 

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RE: [Asterisk-Users] WI FI IP phones??

[Freddi Hansen]

James Moran wrote:



We need to have about 30 phones on one floor

 



I have seen a couple of test where people claim that wi-fi phone network
should use max. 5 simultanoues calls per accesspoint or your audio will start to  
break up. I would take a look at www.kirk.com. They have a DECT basestation with
H323 interface. You can register all your hand set on a single station and then
use their DECT repeater to get the area coverage. Its expensive for a small system
but with 30 handsets it should be comparable to wifi phones and you get at least
5 times the standby/talktime.
my 2 cents.
Freddi 





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[Asterisk-Users] Call forking/parallel call cdr.

[Freddi Hansen]

Hi,
Consider the following:
exten _X.,1, Dial(Zap/g1/12345&Zap/g1/678912)
we attempt to dial 2 numbers simultaneous and who ever answers get the call.
My issue here is that the cdr only contains like ,Zap/g1-1, which 
doesn't tell if 12345 or 678912 answered the call.
One number could be local and the other international so different rates 
should be applied.
Is there any way to know if it was first or second call that connected?.

I have the same with SIP but here my '[EMAIL PROTECTED]' is logged as 
'[EMAIL PROTECTED]' so a workaround is possible.

I hope that I was just let-down by google today and someone has a solution.
Freddi

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Re: [Asterisk-Users] functional difference: canreinvite=yes, no, or update

[Freddi Hansen]

From:
"Kevin P. Fleming" <[EMAIL PROTECTED]>
Date:
Fri, 18 Feb 2005 09:04:35 -0700
To:
Asterisk Users Mailing List - Non-Commercial Discussion 


Olle E. Johansson wrote:
Actually, we could solve Matthew's problem by checking the IP addresses
against the localnet setting and checking if both phones are on the 
same side. If both are within the localnet, we can reinvite. If both 
are on public side, we can reinvite. But if one is localnet and one 
is public, we could automagitically disable reinvite.

Yes, that is a start. As long as you are comparing the "perceived" 
addresses (which I know you would be, I'm just clarifying for others 
who are reading this thread), that will work, because it won't matter 
what private addresses the remote peers may be using behind their NATs.

It will still break in bizarre routing scenarios, but people who build 
those networks are used to dealing with stuff like that.

This should really be the default behaviour if canreinvite=yes and
localnet is set to something.

Agreed.
Kevin,
what about letting this logic be followed by a dialplan macro that gets 
called to make the decision. May sound weird but we have a 'network-id' 
attached to each sip-user.  That network-id contains the public fixed-ip 
of the users network plus a unique name. If the network id's is matching 
and both users are on their 'home-network' then we do re-invite. This 
allows us to deal with special cases where people are behind 
double/triple nat a.s.o. . In other words it allows the Asterisk user to 
overrule the default behavier.  It's allows us (as an  ITSP) todo a safe 
re-invites on known networks to save on customers (and our) bandwith. 
Our hack is really ugly and it would be a like a dream to have this 
stuff in CVS.
Freddi


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Re: [Asterisk-Users] functional difference: canreinvite=yes, no, or update

[Freddi Hansen]

I don't think it would be logical (or efficient) have this run in a 
dialplan macro at all; that would require creating a channel, copying 
variables into it, etc.

I have been thinking about extending the Asterisk expression evaluator 
to allow it to call out to res modules to do the evaluation (passing 
in the channel variables as entities or something)... this would allow 
you to use res_perl or a future res_python or something to do more 
complex data manipulation.

However, even that is only a small part of the solution, since 
chan_sip treats all this information as static right now, and 
extending it to support a dynamic result would take some work.
Kevin,
You're right about the dialplan macro, being able to use res_perl would 
be a much better solution. I think that one the most important thing 
here is to realize that we can't build (hardcode) a safe logic into 
asterisk that automagically handles all nat/re-invite issues, so there 
should be some way that users under some script control dynamically can 
decide if its safe to re-invite. If you look at this list over the that 
year then there has been an almost countless number of attempts to 
describe a safe scenario to use re-invite (like: if both users behind 
same public ip then allow reinvite), and then someone else is pointing 
out that it won't work in this or that scenario. So I think that it show 
that this logic should be controllable by users who might have 
additional knowledge about their network and therefore being able to 
decisions that might not work in other scenarios. We could have some 
'default rules' which experienced user can modify (without going to the 
c-code).
Freddi


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[Asterisk-Users] Re: Grandstream ATA 286 and ilbc (Anton Krall)

[Freddi Hansen]

  That's what I was starting to think.. Since I've always used ulaw or alaw...
Seems that firmware 1.0.5.23 has ilbc broken. 

|-Original Message-

Hi,
it works for me with that firmware but you must set the ilbc
framerate to 30.
(worked with framerate=20 until the 1.0.5.23 release)
Freddi




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re:[Asterisk-Users] SIP hangup issue

[Freddi Hansen]

hi
if I'm on the phone to somewhere through this "SMART IAD" SIP/FXS 
gateway, and I somehow lose contact with the SIP server (for instance 
the SMART IAD reboots), then the channel will hang until the other 
part hangs up.

is it possible to force a hangup on a channel in which the caller is 
no longer available? this would be the desired functionalityl.

regards
roy 

Use RTP timeout, see sip.conf
Freddi
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re:[Asterisk-Users] Multiple AGI Issues

[Freddi Hansen]

To:
"Asterisk Users Mailing List - Non-Commercial Discussion" 




I've got several issues with AGI/FastAGI

1. When an AGI script sends a command to Asterisk via stdin, why does Asterisk 
block and not return a result until the command is complete? Specifically, the 
dial command. If I send a Dial command to Asterisk, I don't get a return result 
until AFTER the call is HUNG UP. Not when it's ringing, not when the call is 
connected, but when it's DISCONNECTED. Why is that? How are you supposed to use 
commands like CHANNEL STATUS if you have to wait until the call is hung up, to 
check it's status?

2. Why do AGI scripts stay in memory until a call is complete? Is there any way 
to have the script terminate when a call is connected? With this scenario, you 
have a script for every single call in place, and that's really bad from a 
system resource perspective.

3. Seems that no scripting language is up to the task of FastAGI. Perl's 
threads aren't thread-safe with DBI and Python's aren't completely thread safe 
either. Don't know about Ruby, and I ain't no C programmer. What have people 
implemented? I also don't like the threading approach, because if something 
goes wrong with the script/server, you lose the ability to place ANY calls.

Doug

  

Hi,
If you want to have a speedy system that doesn't steal to many system 
resources then you have to use FastAGI. That being said you have to 
program your FastAGI server so it's completely event driven. The way to 
deal with f.ex. the dial command can be to let the FastAGI set a 
dialplan variable and then send the control back to the dialplan which 
then can execute the dial command that your FastAGI did prepare. I 
prefer to use perl for most AGI/FastAGI solutions, the servers are 
started out of inittab so no forking overhead during call handling. I do 
normally build FastAGI servers around 'select' so the process is either 
working or waiting on requests. I know you will say that means that no 
FastAGI request are served while I wait for database responses. The 
workaround is to start more than one copy of your FastAGI server on 
different ports. Create a global variable in your dialplan ,  increment  
on each call  - do a mod(4)  if you have  4 servers  so you can 
interleave the FastAGI requests between the servers.  If you need 
persistent data for you 'after call' process then use you DB system.
Let your FastAGI server write a dialplan status variable so you can 
retry another FastAGI server in case first one fails.

It's not difficult to get 100+ call setups per second with this approach.
b.r.
Freddi

  


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RE: [Asterisk-Users] Multiple AGI Issues

[Freddi Hansen]

Thanks for the reply. Neat ideas there, but a couple of issues.

1. Don't want to have to jump around between the FastAGI and the dial plan. Our 
plan is to have NO customer data in the dialplan, as all data will be contained 
within MySQL. We don't want to have to make _any_ edits to the dial plan when a 
new customer is added. It's a provisioning nightmare to have to do this. It 
also may not be a Dial() command that gets excuted for a given number dialled. 
It might be Meetme(), Queue() or something else. Jumping back into the dialplan 
and then executing the right command would be hard to maintain. It'd be helpful 
if Asterisk accepted something like the following, which would make it easier, 
but it doesn't...

exten => _X.,1,AGI(//localhost/script.py)
exten => _X.,2,${APP}( ${ARGS} )
  

We have no customer data in the dialplan,  everything is done through mysql.
There are 2 basic ways (at least) to use FastAGI when you  dont want to 
have a multithreaded  FastAGI server. 
1. Create some 'helper' contexts in the dialplan to handle 
applications/actions that may take some time (like meetme,dial a.s.o.). 
Let the FastAGI server set some channel variables that you may need 
(like Destination number, which CallerID to use aso) and  ofcourse the 
context before returning to the dialplan.
2.Use a modified version of  Asterisk.pm which is build around select 
and nonblocking i/o that uses event driven callbacks into your 
application code.(yes threadsafe, it sleeps on a select call until 
events then create the callback.  Since your not familiar with select i 
would recommend using method 1. 




What about findme/followme functionality? Are we going to have to jump 
backwards and forwards between the agi and the dialplan each time (all the 
while maintaining the last number tried in the agi) a new number is tried? We 
could return ALL the numbers to try at once from the AGI I guess, kinda like 
${NUM1}, ${NUM2}, ${NUM3} etc. Oh YUCK!

2. How did you get around the fact that it's quite clearly documented that the 
perl DBI is _not_ thread safe?
  
You can easily use a perl distro compiled without multithread enabled 
eventhough it shouldn't be needed. Again read up on perl IO::select.  
You sleep waiting for event input, execute the job  and then sleep 
again. No multitread needed.  You  may issue sql request that takes 20 
or 50 msec and nothing else is going on within  a single server during 
that time  so no re-entrance into unsafe DBI code.  It also means  other 
calls are not being served from that server during this period that why 
I pointed you to use the server interleaving.

3. I don't have a high enough confidence in the stability of either perl or 
python threading, to allow the FastAGI server to potentially receive several 
dozen calls, and therefore several threads each. If the FastAGI server crashes, 
you lost the ability to place _any_ calls.

  
As described above: no threading needed within the server. We have AGI 
servers that processes 10K+ calls per day per server ( some of the 
servers has 15K perl lines ) they never crash but they are started out 
of /etc/inittab anyway (just in case)

4. Using select() system calls is a little beyond my abilities...

Doug.
I hope to get some time to do a cleanup on my framework for solution 2 
above, it might benefit some other people that like to use agi-perl

b.r.
Freddi




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[Asterisk-Users] RE: AGI Flakyness *sigh*

[Freddi Hansen]

Subject:
[Asterisk-Users] AGI Flakyness *sigh*
From:
"Douglas Garstang" <[EMAIL PROTECTED]>
Date:
Thu, 16 Feb 2006 09:24:26 -0700

To:
"Asterisk Users Mailing List - Non-Commercial Discussion" 




Well, I'm about ready to throw Asterisk across the room.

Can someone tell me WHY, when you've sent a Dial command to Asterisk via AGI, 
if the callee hangs up the call, Asterisk sends a return code, but if the 
caller hangs up, it does not???

This means if an agi script services a call, and after the two parties have 
finished speaking, the person who initiated the call hangs up, the agi script 
is still waiting for a result from stdin waiting waiting. and it 
never gets anything because Asterisk didn't send a result code. Good grief!

Doug.
  

Hi,
stop sounding like those people that blame their c-compiler whenever 
they can't compile their 'Hello world'.
I you want help from this list then you shouldn't suggest that all the 
help full people on this list aren't telling the truth when they tell 
you that AGI actually works.
Start reading the wiki pages on requirements for writing our own python 
program and then if it doesn't work: post the script involved and not 
just a few lines from an error output.

Freddi


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[Asterisk-Users] OT: proliant fedora asterisk

[Freddi Hansen]

HP doesn't support Fedora on Proliant hw so you can't just install their 
ILO and get access

to hw info like cpu/mb/temperature,powersupply status,fan info aso.
I used the link below to get that access, which enabled me to write a 
small script that sends

snmp-traps to hp-ovo.
I did spend quite some time myself until I found this link.

http://www.foo.be/cgi-bin/wiki.pl/GNULinuxCompaqProliantIML

Freddi

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[Asterisk-Users] RE: OH323 user configuration

[Freddi Hansen]

Subject:
[Asterisk-Users] RE: OH323 user configuration
From:
Code Lover <[EMAIL PROTECTED]>
Date:
Wed, 7 Dec 2005 14:06:40 +0300

To:
asterisk-users@lists.digium.com


Hi

Cab Asterisk accept h323 RAS packet( registration) using OH323 channel.

--
Thank You,
Code Lover

 


Asterisk oh323 is an endpoint. (and it can send RAS registration requests)
Install GnuGK together with Asterisk if you want GK functionallity.
Freddi

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RE: [Asterisk-Users] uniqueid with multiple asterisk hosts

[Freddi Hansen]

Hello!

Soon i will add a second asterisk to my setup and of course i want it to use the same 
postgresql-db as the first one. Basically it's about the cdr-uniqueid. Since it could be 
possible that a record with the same uniqueid is written to the cdr-table by both 
machines i'm lookin for a patch that helps asterisk to produce "real unique" 
uniqueid's(don't know why this is not a standard feature anyways).
in bugs.digium.com i found several approaches of which the one over here looks the 
most convincing: http://bugs.digium.com/view.php?id=5825&nbn=11

think i'll stick with it but wanted to hear what other people use for this issue? will 
there be a "real" uniqueid in the official asterisk code at some point? thanks 
for your answers.

regards
christian


Hi,
it's more 2 years since we build our first Asterisk cluster and I have 
been patching releases ever since with regards to the missing systemname 
so it would be really nice someone could get one of the patches into 
cvs.(sorry svn).
I use the 'hostname' during asterisk start to lookup an 'hostindex' to 
have as small key as possible. The reason for this is that I use it as 
part of unique key in 'realtime' to enable realtime DB to be shared 
between multiple Asterisk servers.
We take care of sip NAT issues by taking the hostidx from the newest 
registration in realtime DB and route the call through that server.

Freddi




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Re: [Asterisk-Users] OT: Linux on treo 650

[Freddi Hansen]

works  nice with sjphone (sip) connecting over wifi to *.
Freddi


BTW, thats a nice looking phone.

On 12/14/05, Rusty Dekema <[EMAIL PROTECTED]> wrote:
 


The PPC-6700
(http://www.mobiletechreview.com/Sprint-PPC-6700.htm) that
I am testing right now may not be able to run Asterisk, but it sure can
connect via 802.11b to my Asterisk system. Unfortunately, the operating
system does not seem to provide access to the "earphone" speaker that the
regular phone uses, and instead only to the loudspeaker on the back of the
phone. So if I want to use VoIP on it, I have to have a headset.

Additionally, the device seems to periodically "freeze up" for a period of
time ranging from 1 to 2 seconds during SIP/RTP calls. The call does not
drop, but the interruption makes it pretty unusable. This happens even with
no other applications running.

The above combined with the fact that the 802.11b network in the campus
where I wanted to use this device seems to have a severe jitter problem that
is incorrectable by a reasonably-sized jitter buffer means that I am
probably going to abandon this experiment, but it was interesting
nonetheless.

-Rusty


On 12/14/05, Kerry Garrison < [EMAIL PROTECTED]> wrote:
   


> But can they run Asterisk and create an IAX trunk back to your PBX while
> running a softphone?
>
> I thought not.
> -Kerry
>
>
 




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[Asterisk-Users] Re: CALLERIDNUM (Rehan AllahWala)

[Freddi Hansen]

Do u know how to instert it in the agi ?

   $AGI->exec("SetCIDNum(8504338555)");


but it didn't work
 


   $AGI->exec('Set',"CALLERID(number)=8504338555");

Freddi




 


www.voip-info.org/wiki-asterisk
or you could try the CLI show application Set, and show function
CALLERID


On 12/28/05, Rehan Ahmed <[EMAIL PROTECTED]> wrote:
 


> Hi
>
> Can you send any example of this command like
>
> Set(CALLERID(num)=value)
>
> Thanks
>
> Rehan
>
>
> On 12/28/05, C F <[EMAIL PROTECTED]> wrote:
 


> > in 1.2 and on (or CVS HEAD) you have to use:
> > Set(CALLERID(num)=value)
> >
> > On 12/28/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
   


> > > is it possible rewrite CALLERIDNUM in the ZAP channel? I use
> > >
> > > [int-transfer]
> > >  exten => _00.,1,SetVar(CALLERIDNUM=${CALLNR})
> > >  exten => _00.,2,MYSQL(Connect connid localhost webcdr
> > >  ser91623 cdr) exten => _00.,3,MYSQL(Query resultid
> > >  ${connid} select\
> > >  if((floor(u.credit/p.cost))>1\,ceil((u.credit)/p.cost)*60\,0)\
> > >  as\ sekund\ from\ user\ u\,\ sip\ s\,\ pricelist\ p\ where\
> > >  u.iduser=s.iduser\ and\ s.idsip=\'${CALLERIDNUM}\'\ and\
> > >  p.acode=s.acode\ and\ u.currency=p.currency\ and\
> > >  right(left(\'${EXTEN}\'\,CHAR_LENGTH(
 


> p.ccode)+2)\,CHAR_LENGTH(p.ccode))\
 


> > >  like\ concat(p.ccode\,\'%\')\ order\ by\ p.ccode\ desc\ limit\
> > >  1) exten => _00.,4,MYSQL(Fetch foundRow ${resultid} sekund)
> > >  ; fetch
 


> row
 


> > > ..
> > > ..
> > >
> > > without success. At row 3 have var ${CALLERIDNUM} original
> > > value, not value from ${CALLNR}.
> > >
> > >
> > > 




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RE:[Asterisk-Users] H323 REGISTRATION PROBLEM: Gatekeeper '[EMAIL PROTECTED] ' found but failed to register

[Freddi Hansen]

Hi all

First of all excuse me if i make such a big post, hope
also to write in the right place.

I need to connect my linux/asterisk (10.0.0.252) box
to a Nortel PBX (192.168.1.10) with h323
I'd like to allow some phones to register via sip to
asterisk and
with these to the Nortel PBX wich gives me the
connections to the outside world (phone)

after downloading and compiling the latest asterisk
source from cvs
OpenH323 v1.15.6, PWlib v1.8.7 (Mimas version from
Voxgratia)
and oh323-0.7.3 from
http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.7.3.tar.gz

starting asterisk i get


< snip>

Hi,
I had the same problem in the same configuration. Asterisk finds the gatekeeper 
but it uses the wrong interface when it it should register.
the problem is in the Mimas-patch2 release.
change your pwlib to v1_9_1 and openh323 to version v1_17_2 then your 
registration works (again).

Freddi



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Re: [Asterisk-Users] AGI ?

[Freddi Hansen]

From:
Jon Scottorn <[EMAIL PROTECTED]>
Date:
Tue, 23 May 2006 12:52:02 -0600

To:
Asterisk Users Mailing List - Non-Commercial Discussion 




On Tue, 2006-05-23 at 19:44 +0100, Thomas Kenyon wrote:

Jon Scottorn wrote:
> Hi All,
>
>I have been attempting to get an AGI LCRdialout script to work. 
> Basically what I need to have happen is when someone dials out a

> number the script check to see if it is local if so, go out the ZAP
> channel. If the ZAP channel is busy, go out the IAX channels, if IAX
> is all busy, go out the SIP channels.  Here is a sample of what I have
> in my script. 
Why can't this be handled directly with the dialplan?



It probably can be but I thought It would be quicker and easier with 
AGI.  I thought I was supposed to be able to get the variable 
DIALSTATUS from asterisk.

Is this not true?
Here are the ways I have been trying but with no success.

$AGI->get_variable(DIALSTATUS);
$AGI->get_variable('DIALSTATUS');
$AGI->get_variable("DIALSTATUS");
$AGI->get_variable(${DIALSTATUS});


try:

my $dialstat = $AGI->get_variable('DIALSTATUS');

Freddi


Any other thoughts anyone might have.

Thanks for the help and input.
*/Jon Scottorn/*


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Re: [Asterisk-Users] AGI ?

[Freddi Hansen]

 On Wed, 2006-05-24 at 11:45 -0500, Eric "ManxPower" Wieling wrote:

>>
>> my $dialstat = $AGI->get_variable('DIALSTATUS');
>>
> 
> Hi,
> 
>I have tried that as well.

> Thanks for the suggestion.
> 
> Any other thoughts.


1) What version of Asterisk are you using?



I am running from debian package version 1.2.7.1.dfsg-2

2) Can you get any other dialplan variables?



I have tried getting other variables and no I do not get any results.

3) Are you running the Dial app inside your AGI or before you run your AGI?



here is what my dialplan looks like within extensions.conf
exten => _5NXX,1,AGI,LCRdialout.agi.test|${EXTEN}
exten => _51NXXNXX,1,AGI,LCRdialout.agi.test|${EXTEN}

Here are the two lines that I run from within the AGI script
$AGI->exec("DIAL Zap/g2/$number|25|TW");
$callStatus = $AGI->get_variable('DIALSTATUS');

$AGI->exec("DIAL 
IAX2/$iaxUser\:[EMAIL PROTECTED]/$one$areaCode$number|25|TW");

$callStatus = $AGI->get_variable('DIALSTATUS');


Hi
Here is an example that works fine for me:
   my 
$stat=$AGI->exec('Dial',"SIP/$ses{DST_ContactUsername}/$ses{Dialnumber}|$TimeOut");

   my $dstat=$AGI->get_variable('DIALSTATUS');

Here is the way I normally debug AGI's.:
1. try agi from commandline so you know you dont get syntax errors.
2. start your asterisk from command line (not safe_asterisk + asterisk 
-r)   This  is to get stdout from your agi program.
3. issue 'agi debug' to trace whats going on.   AGI  is a bit sensitive 
to you  sending wrong number of args, it can get out-of-sync so an error 
you make in one line may show up in the next command but it's pretty 
easy to spot with  'agi debug on'.

It might be just that you are giving 1 arg when you exec dial while I use 2.
Freddi



Thanks,
*/Jon Scottorn/*

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[Asterisk-Users] SIP Responsecodes

[Freddi Hansen]

Hi,
It seems as 'the google' has left me today so I am trying the list.
How do I get access to SIP responsecodes from dialplan/agi. Yes I know 
that I should stay with 'DIALSTATUS' but there are cases where I need 
the responsecode like '484 adress incomplete' and not just the 'NO 
ANSWER' DIALSTATUS.

Is there a channel variable/function that skipped over by mistake?
Freddi
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RE: [Asterisk-Users] Sending SIP NOTIFY / How to get remote SIP port?

[Freddi Hansen]

try,

database get SIP/Registry/
it gives you a string which contains the info, then pass it to CUT to 
extract ip-adr and port


Freddi


To do that you need to get the remote ip address and port of the sip peer!

I found the function:

${SIPPEER(exten:ip)

But how can I get the port???

http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sippeer

You have all sorts of info BUT... I see no port!!!  :-( 


I can't believe that you wouldn't be able to get the port via a simple dial
plan function???


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RE: [asterisk-users] RPID

[Freddi Hansen]

Hi,
Here is how I have it working:

If  Alice calls Bob and Bob's phone diverts the call to Carol.
You want Bob to pay for the call and the callerid  shown to Carol to be 
'Alice'


On Bob's server
exten _X.,1,sipaddheader(Divertion:<[EMAIL PROTECTED];user=phone>

The proxy that routes bob's call to Carol  will then charge Bob for the 
call and the From: field will be Alice


If you are an ITSP using Asterisk then you must look for the 'Divertion' 
header in incoming SIP invite's yourselves with a


sipheader(Divertion) command

I have this working in a few different scenarios

I think that the right thing todo would be setting the RDNIS if the 
'Divertion' is present on inbound side but I am not sure about this so I 
am using a private variable and doing this outside the SIP channel


b.r.
Freddi 




Thanks... I did some research and found that it's actually not what I was
wanting (unless I missed something lol).  I'm actually looking for a way to
forward caller id information to the called party on a forwarded call.  I
may just need to dig deeper.

On another note, I did find a patch in mantis that is considered
experimental that does get it to where you can see the caller id of who
you're calling based in the dialplan.  Back to the drawing board though  :) 


Aaron

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Wednesday, September 27, 2006 4:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RPID

DANIEL, AARON MATTHEW wrote:
  
> Has anyone successfully gotten rpid working between two phones through 
> asterisk?
> 
>  
> 
> Aaron Daniel
> 
> Computer Systems Technician
> 
> Sam Houston State University
> 
> [EMAIL PROTECTED]
> 
> (936) 294-4198
> 



Aaron,

RPID is supported in Asterisk but many phones do not support it.
Try 
adding the following to sip.conf:


sendrpid=yes
trustrpid=yes

If it is going to work with your phones, it will just work.  If not,

chances are your phone does not support RPID.  You can always look at a 
SIP debug to make sure it is getting sent.  Even if your phones do not 
support RPID, From: usually works just fine  :) .


--
Kristian Kielhofner
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Re: [asterisk-users] INVITE retransmission by 1.8... (Steve Murphy)

[Freddi Hansen]

I have a site that moved to the latest 1.8 revision, and began to
have problems with phones in "far away places" (South America,
and the MidEast).

What I see is that when a Dial() is issued, the sip channel driver
sends out an INVITE to the phone.  Very soon thereafter,  Asterisk
retransmits the INVITE. The phone sends back a 100 Trying, and
then, usually, a 400 response. I may be misinterpreting what I see,
but I *think* that the response from the phone is delayed just enough
to invoke the retransmission. The phone responds to the second invite
with a 400 code, which Asterisk interprets as an error, and the call 
immediately

goes into hangup.

Has anyone else seen this? It doesn't happen all the time, and only 
with certain
phones. It comes and goes, but when it comes, phones become 
unreachable. It
seems to be in this state the majority of the time for certain phones. 
While most

phones seem far away, some are in Florida.

We replaced the 1.8 version of Asterisk with a 1.6.2 version, and the 
issue has
gone away.  I know, I know, I could give more detail, fill out a bug 
report, but

this is the early stages. I may be misinterpreting what I'm seeing.

 Anyone else seen this sort of thing? Any words of wisdom?


hi,
one of our gateways is used for SIP over satelite links and we se the 
same thing on default installation.

The fix is to change chan_sip.c
#define DEFAULT_RETRANS  to a higher value, we use 3000.

The retransmit timer at the far end (pap2t) is increased to 3 times its 
standard values.


It probably breaks some sip specs but its needed to keep it working when 
roundtrip gets to big.


Freddi

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[asterisk-users] Problem with blank/empty voicemails

[Freddi Hansen]

Hi,
I hope for a hint on this issue.

I had a voicemail running on ast release 1.6.2 latest which i upgraded 
to 1.8.11 latest release.

during this process I did add a couple of fields like minsecs and maxsecs.

I do now get empty emails where the attached voicefile only contains the 
voice header,

the message length written in the email is ok.
If I go to the voicemailbox during the recording then I can se the files 
grow to the filesize i would expect, looks like everything is ok until then.


When I press the '#' or hangup then the email is generated with the 
empty attachment and the voicefiles in the INBOX is now

44 bytes for .wav and
60 byes for .WAV

I have voicemail in mysql and messages stored i filesystem
here is some of the config (which worked ok on release 1.6.2)
attachfmt=wav
deletevoicemail=no
volgain=0.0
minsecs=2
maxsecs=600

[Apr 23 21:59:46] DEBUG[31841] app.c: Locked path 
'/var/spool/asterisk/voicemail/default/88855/INBOX'
[Apr 23 21:59:46] DEBUG[31841] app.c: Unlocked path 
'/var/spool/asterisk/voicemail/default/88855/INBOX'
[Apr 23 21:59:46] DEBUG[31841] channel.c: Scheduling timer at (50 
requested / 50 actual) timer ticks per second
[Apr 23 21:59:46] VERBOSE[31841] file.c: -- 
 Playing 'beep.alaw' (language 'dk')
[Apr 23 21:59:46] DEBUG[31841] channel.c: Scheduling timer at (182 
requested / 182 actual) timer ticks per second
[Apr 23 21:59:46] DEBUG[31841] channel.c: Scheduling timer at (0 
requested / 0 actual) timer ticks per second
[Apr 23 21:59:46] DEBUG[31841] channel.c: Scheduling timer at (0 
requested / 0 actual) timer ticks per second
[Apr 23 21:59:46] DEBUG[31841] channel.c: Scheduling timer at (0 
requested / 0 actual) timer ticks per second
[Apr 23 21:59:46] VERBOSE[31841] app_voicemail.c: -- Recording the 
message
[Apr 23 21:59:46] DEBUG[31841] app.c: play_and_record: , 
/var/spool/asterisk/voicemail/default/88855/tmp/N0NNlv, 'wav49|gsm|wav'

[Apr 23 21:59:46] DEBUG[31841] app.c: Recording Formats: sfmts=wav49
[Apr 23 21:59:46] VERBOSE[31841] app.c: -- x=0, open writing:  
/var/spool/asterisk/voicemail/default/88855/tmp/N0NNlv format: wav49, 
0xb3501378
[Apr 23 21:59:46] VERBOSE[31841] app.c: -- x=1, open writing:  
/var/spool/asterisk/voicemail/default/88855/tmp/N0NNlv format: gsm, 
0xb266b1b0
[Apr 23 21:59:46] VERBOSE[31841] app.c: -- x=2, open writing:  
/var/spool/asterisk/voicemail/default/88855/tmp/N0NNlv format: wav, 
0xb3519948
[Apr 23 21:59:46] DEBUG[31841] dsp.c: Setup tone 1100 Hz, 500 ms, 
block_size=160, hits_required=21
[Apr 23 21:59:46] DEBUG[31841] dsp.c: Setup tone 2100 Hz, 2600 ms, 
block_size=160, hits_required=116
[Apr 23 21:59:46] DEBUG[31841] channel.c: Set channel 
SIP/_Mw_cHFm6vZAyV-00012a5c to read format slin

[Apr 23 22:00:03] VERBOSE[31841] app.c: -- User hung up
[Apr 23 22:00:03] DEBUG[31841] channel.c: Set channel 
SIP/_Mw_cHFm6vZAyV-00012a5c to read format alaw
[Apr 23 22:00:03] DEBUG[31841] app.c: Locked path 
'/var/spool/asterisk/voicemail/default/88855/INBOX'
[Apr 23 22:00:03] DEBUG[31841] app.c: Unlocked path 
'/var/spool/asterisk/voicemail/default/88855/INBOX'
[Apr 23 22:00:03] DEBUG[31841] app_voicemail.c: Attaching file 
'/var/spool/asterisk/voicemail/default/88855/INBOX/msg', format 
'wav', uservm is '2048', global is 2048
[Apr 23 22:00:03] VERBOSE[31841] config.c:   == Parsing 
'/var/spool/asterisk/voicemail/default/88855/INBOX/msg.txt': [Apr 23 
22:00:03] DEBUG[31841] config.c: Parsing 
/var/spool/asterisk/voicemail/default/88855/INBOX/msg.txt

[Apr 23 22:00:03] VERBOSE[31841] config.c:   == Found
[Apr 23 22:00:03] VERBOSE[31841] config.c:   == Parsing 
'/var/spool/asterisk/voicemail/default/88855/INBOX/msg.txt': [Apr 23 
22:00:03] DEBUG[31841] config.c: Parsing 
/var/spool/asterisk/voicemail/default/88855/INBOX/msg.txt

[Apr 23 22:00:03] VERBOSE[31841] config.c:   == Found
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Evaluating 'VM_NAME' (from 'VM_NAME}:
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Result of 'VM_NAME' is 'Freddi Hansen'
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Evaluating 'VM_DUR' (from 'VM_DUR} 
lang besked (number ${VM_MSGNUM})

[Apr 23 22:00:03] DEBUG[31841] pbx.c: Result of 'VM_DUR' is '0:16'
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Evaluating 'VM_MSGNUM' (from 
'VM_MSGNUM})

[Apr 23 22:00:03] DEBUG[31841] pbx.c: Result of 'VM_MSGNUM' is '1'
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Evaluating 'VM_MAILBOX' (from 
'VM_MAILBOX} fra ${VM_CALLERID}, den ${VM_DATE}.  Tak!

[Apr 23 22:00:03] DEBUG[31841] pbx.c: Result of 'VM_MAILBOX' is '88855'
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Evaluating 'VM_CALLERID' (from 
'VM_CALLERID}, den ${VM_DATE}.  Tak!
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Result of '

Re: [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used

[Freddi Hansen]

On 07/12/2012 09:19 AM, Benny Amorsen wrote:

"Kevin P. Fleming"  writes:

That's quite interesting; can you describe a scenario where this 
occurs?


Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24
and eth1 with 10.0.2.1/24. Further imagine that you wish to be able to
move phones between the networks without changing the SIP server
address, so you set 192.168.1.1 as the SIP server no matter which
network they happen to be on.

Now the phones which happen to be connected to eth1 will send a request
to 192.168.1.1. If Asterisk is bound to 0.0.0.0, the reply will come
from 10.0.2.1. This could be solved if Asterisk did a connect() to the
socket and use the same socket for answering. That would tell the system
IP stack that this is in fact a connection, and so the system would
ensure that the reply source IP would be correct.


I must be missing something. If a phone sends a UDP packet to 
192.168.1.1, how does that get routed to (arrive at) the 10.0.2.1 
interface on the Asterisk server? The only way I can imagine that 
happening is if a router in between the phone and the server has been 
told that 192.168.1.0/24 is reachable *through* 10.0.2.1, which seems 
like a bizarre way to construct a network. Getting replies from 
Asterisk *back* to the phone would also require the IP stack on the 
Asterisk server to route those replies back over the 10.0.2.0/24 
interface instead of the 192.168.1.0/24, which doesn't make any sense 
either.


We have since Asterisk 1.2 been using a configuration with 6 NIC's 
bonding to 3 networks, one public internet and 2 private networks.
Routing calls between networks and having phones on all 3 networks is no 
problem.


There is one case though where we do fixup with iptables.
We have 30 virtuel adresses on one of the private networks and when 
Asterisk sends a packet to a destination then the first address of the 
NIC is inserted as source  by the OS.


example
one NIC has ip's
192.168.0.10,192.168.0.20,192.168.30
Telephone (192.168.0.100) sends a packet to Asterisk 192.168.0.30, 
Asterisk sends response to 192.168.0.100 but with source address 
192.168.0.10 as thats the first ip on that NIC.


In Iptables OUTPUT q we do a set-mark to an index into our source ip's
then in POSTROUTING we insert the source adr using the mark

b.r
Freddi




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Re: [asterisk-users] High resident memory with 11.14.0 ?

[Freddi Hansen]

Its up to 5.8G of resident memory with 28321 calls processed.
The OOM killer is going to kill this soon at this rate (8GB RAM machine).
This seems like a pretty serious problem.
It looks like I'll need to restart asterisk every night
Hi the number of cpu cores that you see with top  times 512Mbyte is the 
level of ram that's needed


e.g. a hp-gen8 with 2 octo core cpu's and hyperthreading enabled will be 
( 2 x 8 x 2  x 0,5 gb ) = 16 gb  + a bit exstra.
So from start memory usage increases until it reaches 17.3 gb and then 
stabilizes. at that level.

You can disables hypertreading and cut your ram usage to half of that.

I can't see what hardware you are using but I think you need to check 
that the rule above fits your hardware.


b.r.
Freddi







On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna > wrote:

Hi,
I have an Asterisk server that's been running now for around 2 days.
I've noticed that the resident memory seems to be very high for its 
current call load:


  PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+ COMMAND   
18321 asterisk  20   0 8050m 5.2g 6968 S   
13 66.2 363:11.80 asterisk


$ asterisk -rx "core show channels"

24 active channels

12 active calls

25216 calls processed


This server has a bunch of IAXModems hooked up to it and is mainly 
used as a Fax gateway to hylafax. Is this normal? 5.2Gig of memory 
used after 2 days with only 12 currently active calls?


I am not using any realtime peers.

There are 100 registered SIP peers on this server as well.

Thanks.

-- James

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[asterisk-users] Taskprocessors

[Freddi Hansen]

Hi,
Trying to move traffic from one of our asterisk gateways using 11.21 to 
a newly installed using 13.8.0/13.9.0-rc1.


After half an hour with peers just registering and no calls the 'core 
show taskprocessors' shows between 4500 and 5000 processors.


the ones that there are a lot of is

subp:SIP/   2  0   2

Doesn't look right to me that there should be so many 'hanging/waiting'

using Centos 7 / and sippers are via odbc.

Initialy I had a lot of subm:...  hanging and these processors would 
take 5 seconds to repond to a ping and jobs in their input queue would 
newer be finished.
That problem did disappear when sippeers was updated with 
subscribemwi='yes'.


Can someone give me a hint, I have installed a lot of asterisk systems 
but never stumpled upon this before.


b.r.
Freddi





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Re: [asterisk-users] Pickup(), PickupChan()... PickupQueue()? (Niccol? Belli)

[Freddi Hansen]

Hi,
I'm currently using Pickup() to pickup calls from queues, but this is 
VERY annoying because often users from different queues dialed the 
very same extension (for example they pressed '1' to reach something 
but in different submenus). Other times they didn't dial anything but 
they end up in the very same queue, so the extension to pickup is the 
number they called.
So every time I want to send users to a queue I have to put a Goto() 
before the Queue() app because I need to uniquely identify the 
extension (for example Goto(QueueName,1)).
This is annoying. Really annoying. It also makes the dialplan hard to 
read.
Since we also have PickupChan() is to would be nice to have 
PickupQueue() too. That way we shouldn't care about the extension, we 
should simply write PickupQueue(QueueName). Simple and clear, the 
dialplan thanks.

Hi,
you could use the PICKUPMARK with the Pickup().

before you call the Queue app you set PICKUPMARK=Queuename.
When you want to pickup the call you do Pickup(Queuename@PICKUPMARK) to 
only get calls in the Queue with Queuename.


b.r.
Freddi

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