Re: [asterisk-users] Newbie alert: VoIP hardware

[Stelios Koroneos]

Questions:
[1] Can I use oslec for echo cancellation? I'll have beefy hardware.
Is echo cancellation necessary?

Yes you can use oslec provided that either your distribution has a zaptel
package with the oslec patch (or you build the zaptel drivers + oslec
yourself)
Well without echo cancelation you will probably have a number of calls that
have either very bad sound quality or are simply annoying
With your set i.e 3-4 lines processing requirments are minimal so you should
not worry about that.We have been able to run oslec for 4 lines on a 266Mhz
(no its not Ghz) powerpc embedded board with very good results

Stelios S. Koroneos

Digital OPSiS - Embedded Intelligence
http://www.digital-opsis.com  


 



  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Repo
Sent: Wednesday, May 07, 2008 8:11 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Newbie alert: VoIP hardware


Hello,

Please forgive me for i'm not an asterisk user yet. I've done as much
research as I can .. and have the following questions.

I'm setting up a new office and a home office and i'm shopping for hardware.


Office: 2 analog lines
Hardware: TDM412B (2 FXO, 1FXO)
Link: http://www.voipsupply.com/index.php?cPath=99_555_556
Cost: $303

Home: 1 analog line
Hardware: TDM421B (2 FXS, 1 FXO)
Link: http://www.voipsupply.com/product_info.php?products_id=3980
Cost: $300
[2] Can I get PCI express x1 cards for the same price?

I'm on budget, Any other cards (sangoma? rhino?) that might work well?

I'm sure these questions have been asked before.. :-)

Steve




___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] queue

[Bhrugu Mehta]

hi, all
is there any way in queue app. to execute asterisk app. after Queue() app. i.e

[myplan]
exten => _X.,1,Answer
exten => _X.,n,Queue(myqueue)
exten => _X.,n,Background(file-to-play)

exten => 1,1,Playback(thnks)
exten => 2,2,Playback(by)

Is these possible above situation , how 
thnks, Bhrugu Mehta

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Newbie alert: VoIP hardware

[Steve Repo]

Hello,

Please forgive me for i'm not an asterisk user yet. I've done as much
research as I can .. and have the following questions.

I'm setting up a new office and a home office and i'm shopping for hardware.


Office: 2 analog lines
Hardware: TDM412B (2 FXO, 1FXO)
Link: http://www.voipsupply.com/index.php?cPath=99_555_556
Cost: $303

Home: 1 analog line
Hardware: TDM421B (2 FXS, 1 FXO)
Link: http://www.voipsupply.com/product_info.php?products_id=3980
Cost: $300

Questions:
[1] Can I use oslec for echo cancellation? I'll have beefy hardware.
Is echo cancellation necessary?

[2] Can I get PCI express x1 cards for the same price?

I'm on budget, Any other cards (sangoma? rhino?) that might work well?

I'm sure these questions have been asked before.. :-)

Steve
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] phone status question

[Rilawich Ango]

Hi,
A makes call to B.  B has connection problem with the server (say, the
lan cable is unplugged).
1: A ---> server
2: A <--- server
3: server > B

In 2, server will send the ring to A and it will hear ringing tone.
In 3, server will try to connect B until timeout.

My question is:
A will still wait for B but B is physical unreachable.  Can I set the
number of retry time in server to try to reach the destination instead
of the timeout?

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] better enumlookup handler

[Brian J. Murrell]

Does anyone have a better ENUM lookup handler than the built-in
ENUMLOOKUP() function?  The built-in function does not properly handle
multiple return values such as:

8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u" "E2U+SIP" 
"!^\\+1866(.*)$!sip:[EMAIL PROTECTED]" .
8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u" "E2U+SIP" 
"!^\\+1866(.*)$!sip:[EMAIL PROTECTED]" .

And thus does not handle roll-over should one be unavailable for
whatever reason.

There is this voip-info.org wiki page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+and+multiple+ENUM
+entries but the downloads that it's pointing to seem to be dead.

Sure I could take to writing an AGI script and probably be done it in a
few hours, but why re-invent the wheel?

Thanx,
b.



signature.asc
Description: This is a digitally signed message part
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Melbourne Asterisk night

[Paul Hales]

Tomorrow night is the monthly Asterisk night...in melbourne
(australia)...

The usual stuff - get together, eat, show off tech toys.

At the Pint on Punt, from 7pm.

later,

PaulH





___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Steve Totaro]

On Tue, May 6, 2008 at 11:42 AM, Anthony Francis <[EMAIL PROTECTED]> wrote:
>
>
>
>  Tilghman Lesher wrote:
>  > On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
>  >
>  >> 5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
>  >>
>  >>> On Monday 05 May 2008 11:24, Johansson Olle E wrote:
>  >>>
>   5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
>  
>  > On Monday 05 May 2008 09:45, Johansson Olle E wrote:
>  >
>  >> Another issue that we need to fix with the MYSQL driver is that
>  >> we're
>  >> lacking a connection pool. Everything seems to be handled over one
>  >> connection to Mysql, which causes issues.
>  >>
>  > That's not true.  The MYSQL app generally uses multiple connections,
>  > one
>  > for each channel.  The only way one might use only a single
>  > connection is
>  > by using a global variable to store a single connection id, but that
>  > method
>  > is not documented anywhere, AFAIK.
>  >
>   You talk about the Mysql APP, but is this the case with the Realtime
>   driver as well?
>  
>  >>> No, the native Realtime driver uses a single connection.  The ODBC
>  >>> Realtime
>  >>> driver generally uses a single connection but can be configured to
>  >>> use a
>  >>> separate connection for each query.
>  >>>
>  >> So, we're back to where we started. A developer that can help us with
>  >> a connection
>  >> pool or a separate connection for each query would be a Nice Thing (TM).
>  >>
>  >
>  > What issues are you specifically seeing that merit using multiple
>  > connections?
>  >
>  >
>  I can specify an issue that would merit multiple connections, if the
>  link to your db goes away Asterisk likes to freeze writing CDRs.
>  I have a few remote * servers that this happens to. My solution so far
>  has been to record CDR's to a local DB and then have a
>  perl script that attempts to move them over to my transaction DB. I
>  would suggest this solution to anyone who depends on their CDR records.
>
>  --
>  Thank you and have any kind of day you want,
>
>  Anthony Francis
>  Rockynet VOIP
>

I would not run MySQL on the local box.  I would simple use Asterisk's
csv CDRs and then use some script to import the CSVs into a database
residing on another server using some sort of script.  Depending on
your needs, you could probably run that during low call volume.  I
also think that you adapt the free queue_log to database script by
Queuemetrics to do what you want on the fly.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Storing voicemail on samba share

[Steve Totaro]

On Tue, May 6, 2008 at 5:19 PM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
>
> On Tue, May 06, 2008 at 03:33:31PM -0400, OCG Technical Support wrote:
>  >A client has asked that our asterisk installation leverage their large
>  >investment in their existing data center infrastructure.  We're thinking
>  >about putting the voicemail messages onto a Samba share (on their file
>  >servers).  Any pros/cons to this?  Does network/samba latency create
>  >choppiness?
>
>  I think a lot of that would depend on *how* you put the files there.
>
>  smbclient?  cifs mount?
>
>  I would say write them locally, and then use a looper script to push
>  them to the other server... though on reflection, Asterisk might not be
>  happy with that.
>
>  Cheers,
>  -- jra
>  --
>  Jay R. Ashworth   Baylink  [EMAIL 
> PROTECTED]
>  Designer The Things I Think   RFC 
> 2100
>  Ashworth & Associates http://baylink.pitas.com '87 
> e24
>  St Petersburg FL USA  http://photo.imageinc.us +1 727 647 
> 1274
>
>  Those who cast the vote decide nothing.
>  Those who count the vote decide everything.
>-- (Joseph Stalin)
>

How about using rsync or FTP along with the Samba share?  Just a
thought, no real though or testing involved.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Receptionist SNOM-360

[FaberK]

Hi to all,
I got an Asterisk with 2 BRI(7 pstn numbers and 4 concurrent calls)
and 15 SIP extensions.
The receptionist has a SNOM-360.
How many SIP accounts would you configure on that phone?
Only one would be enough?
One SIP account, has a limit on concurrent calls?
I saw that the SNOM-360 can handle up to eleven SIP accounts.

Thanks

-- 
.:FaberK:.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Al Baker]

Are you saying the * server does NOT TRY to re-establish the BD connection ?

Does your whole * SERVER freeze ?

If  NOT, what happens to you CDR records ?

Anthony Francis wrote:
> Tilghman Lesher wrote:
>   
>> On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
>>   
>> 
>>> 5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
>>> 
>>>   
 On Monday 05 May 2008 11:24, Johansson Olle E wrote:
   
 
> 5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
> 
>   
>> On Monday 05 May 2008 09:45, Johansson Olle E wrote:
>>   
>> 
>>> Another issue that we need to fix with the MYSQL driver is that
>>> we're
>>> lacking a connection pool. Everything seems to be handled over one
>>> connection to Mysql, which causes issues.
>>> 
>>>   
>> That's not true.  The MYSQL app generally uses multiple connections,
>> one
>> for each channel.  The only way one might use only a single
>> connection is
>> by using a global variable to store a single connection id, but that
>> method
>> is not documented anywhere, AFAIK.
>>   
>> 
> You talk about the Mysql APP, but is this the case with the Realtime
> driver as well?
> 
>   
 No, the native Realtime driver uses a single connection.  The ODBC
 Realtime
 driver generally uses a single connection but can be configured to
 use a
 separate connection for each query.
   
 
>>> So, we're back to where we started. A developer that can help us with
>>> a connection
>>> pool or a separate connection for each query would be a Nice Thing (TM).
>>> 
>>>   
>> What issues are you specifically seeing that merit using multiple
>> connections?
>>
>>   
>> 
> I can specify an issue that would merit multiple connections, if the 
> link to your db goes away Asterisk likes to freeze writing CDRs.
> I have a few remote * servers that this happens to. My solution so far 
> has been to record CDR's to a local DB and then have a
> perl script that attempts to move them over to my transaction DB. I 
> would suggest this solution to anyone who depends on their CDR records.
>
>   

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Storing voicemail on samba share

[Jay R. Ashworth]

On Tue, May 06, 2008 at 03:33:31PM -0400, OCG Technical Support wrote:
>A client has asked that our asterisk installation leverage their large
>investment in their existing data center infrastructure.  We’re thinking
>about putting the voicemail messages onto a Samba share (on their file
>servers).  Any pros/cons to this?  Does network/samba latency create
>choppiness?

I think a lot of that would depend on *how* you put the files there.

smbclient?  cifs mount?

I would say write them locally, and then use a looper script to push
them to the other server... though on reflection, Asterisk might not be
happy with that.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] how do I set callerid for incoming iax?

[sean darcy]

Tilghman Lesher wrote:
> On Tuesday 06 May 2008 12:45:42 sean darcy wrote:
>> Using 1.6-rc8.
>>
>> In iax.conf on the calling box, I have:
>>
>> [iax-out]
>> .
>> callerid = "sean" <447>
>>
>> I even also put the same on called box.
>>
>> But I can't seem to set the callerid:
>>
>> exten =>_NXX,1,Answer()
>> exten =>_NXX,n,NoOp(${CALLERID(num)})
>>
>>
>> Answer("IAX2/iax-in-7", "") in new stack
>>   NoOp("IAX2/iax-in-7", "") in new stack
>>
>> So how do I set callerid?
> 
> "iax-out" != "iax-in"
> 

So??

On the calling box,
[iax-out]
type=friend
username=iax-in
secret=password
peercontext=longdistance ; which also does extensions
host=
qualify=yes
trunk=yes
callerid = "sean" <447>

On the called - receiving - box:

[iax-in]
type=friend
username=iax-in
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid = "sean" <447>

and  then the cli shows:

-- Executing [EMAIL PROTECTED]:1] Answer("IAX2/iax-in-1", "") in 
new stack
 -- Executing [EMAIL PROTECTED]:2] NoOp("IAX2/iax-in-1", "") 
in new stack


I must be missing something. The name of the iax.conf context matters 
somehow?

Thanks for any help.

sean


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Julian Yap]

On Tue, May 6, 2008 at 1:38 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote:
>  We are actually running an AsteriskNow appliance with asterisk 1.4.18.1
>  and it's quite unstable.
>  We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX destroy
>  deadlock"
>  and now that we have added a Queue, it's worse than ever. The queue goes
>  stuck quite often
>  (agent are stuck in 'In use' state and if they logoff they can't log-in
>  till an asterisk restart).

There's an IAX issue with the security patch for 1.4.18.1... and 1.4.19.1.

There's another thread on this.

- Julian

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Storing voicemail on samba share

[Matt Watson]

It would probably be wiser to run an IMAP server and do imap storage instead of 
writing to a cifs-mounted directory... or use ODBC storage... assuming they are 
running a database server somewhere.

I don't have any experience with having * write voicemail files to CIFS/SMBFS, 
but I also think its not something I would try... I've personally always found 
that most network file systems don't tend to handle disconnects (server 
reboots, network outages, etc.) very well.  Mind you, it might of come along 
way since the last time I tried.

--
Matt

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical 
Support
Sent: Tuesday, May 06, 2008 3:34 PM
To: 'Asterisk Users List'
Subject: [asterisk-users] Storing voicemail on samba share

A client has asked that our asterisk installation leverage their large 
investment in their existing data center infrastructure.  We're thinking about 
putting the voicemail messages onto a Samba share (on their file servers).  Any 
pros/cons to this?  Does network/samba latency create choppiness?

Thanks,
MD
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Benoit Plessis]

Tzafrir Cohen a écrit :
> On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:
>
>   
>> Here it is, but since the AsteriskNow release has stripped the binary
>> i fear it won't be of much use:
>> 
>
> Is there any "-debug" package for asterisknow's asterisk package?
>
> On RedHat they are generated automatically. On Debian they require some
> extra settings, and has been present in recent Asterisk packages (the
> asterisk-dbg package) but not in all of the smaller modules packages.
>
>   
Nope, already tried this before posting
but nothing like that appears on conary

anyway, i'll be migrating on a debian asap, since i now this
much better and the advantages of AsteriskNow keep reducing

as a matter of fact i already now that some thing that doesn't work 
under AstNow
(my siemens sip hardphones, and my SIP provider (Keyyo) at least) work 
with the
debian packaged asterisk.
Well for the sip provider it's not that it doesn't work, more than the 
only way to have some
sound is to use the 'm' flag of the Dial() command to have the moh 
played during the ringing.
Given that, i got some sound when the call is established ...

-- 
Benoit



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Storing voicemail on samba share

[OCG Technical Support]

A client has asked that our asterisk installation leverage their large
investment in their existing data center infrastructure.  We're thinking
about putting the voicemail messages onto a Samba share (on their file
servers).  Any pros/cons to this?  Does network/samba latency create
choppiness?

 

Thanks,

MD

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] asterisk queue cluster

[Vieri]

I setup two asterisk servers with identical settings
(same extensions, same queues, etc). Each one is
connected to the same amount of incoming/outgoing
links (1 PRI, 4 BRI, 1 IAX  friend, etc, on each box).
Most extensions are sip and they register via DNS SRV
and other methods so that the two servers are load
balanced. Incoming PSTN calls (BRI) reach 50% each
server so that's load balanced too. I use DUNDi with a
IAX friend between the 2 servers to lookup where an
extension was registered (whether on pbx1 or pbx2).
Everything is working as expected (SIP on PBX1 calls
SIP on PBX2 and vice versa; PSTN incoming call reaches
SIP extension or group whether it registered on pbx1
or pbx2, etc.).

My question regards queues.
Incoming PSTN calls have 50% chance of entering, say,
queue number 1000 on pbx1 and 50% on pbx2 (same queue
1000). All calls will be correctly bridged to the
agent, whether it randomly registered on pbx1 or pbx2.
However, how can I make pbx1 and pbx2 "counters
consistent"? ie. first call enters queue 1000 on pbx1
and will have position 1; second call entes queue 1000
and happens to fall on pbx2 and will also have
position 1. The worst case would be if there are, say,
10 callers in queue 1000 on pbx1 and the 11th call
arrives on pbx2 with position 1.

Is there a way of coherently setting up a clustered
queue?
Does anyone have examples/workarounds/links?

Thanks!

Vieri


  

Be a better friend, newshound, and 
know-it-all with Yahoo! Mobile.  Try it now.  
http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Tzafrir Cohen]

On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:

> Here it is, but since the AsteriskNow release has stripped the binary
> i fear it won't be of much use:

Is there any "-debug" package for asterisknow's asterisk package?

On RedHat they are generated automatically. On Debian they require some
extra settings, and has been present in recent Asterisk packages (the
asterisk-dbg package) but not in all of the smaller modules packages.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Mixmonitor recording issue

[Vinícius Fontes]

I had a similar problem when the calls were not recorded when there was a 
transfer. Is that your case? If so, the solution is to start recording on the 
inbound leg of the call for all channels. Just like that:

[default]
exten => _00[2-6]XXX,1,Dial(Local/[EMAIL PROTECTED])

[recording]
exten => 
_00[2-6]XXX,1,MixMonitor(${CALLERID(num)}-${STRFTIME(${EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav)
exten => _00[2-6]XXX,n,Dial(Zap/g1/${EXTEN:1})


Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.

- "Rahul Yadav" <[EMAIL PROTECTED]> escreveu:

> Hi All
> 
> 
> I am using asterisk 1.4.18.1.My problem is that Mixmonitor is not
> recording complete recording.
> Suppose i got a call connected and talking for 3 minutes then
> mixmonitor records only 2 minutes of call.
> This problem happens randomly.Please help as i am suffering very much
> due to this problem.
> 
> Rahul Yadav 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] how do I set callerid for incoming iax?

[Tilghman Lesher]

On Tuesday 06 May 2008 12:45:42 sean darcy wrote:
> Using 1.6-rc8.
>
> In iax.conf on the calling box, I have:
>
> [iax-out]
> .
> callerid = "sean" <447>
>
> I even also put the same on called box.
>
> But I can't seem to set the callerid:
>
> exten =>_NXX,1,Answer()
> exten =>_NXX,n,NoOp(${CALLERID(num)})
>
>
> Answer("IAX2/iax-in-7", "") in new stack
>   NoOp("IAX2/iax-in-7", "") in new stack
>
> So how do I set callerid?

"iax-out" != "iax-in"

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] What is field 'User/ANR'

[Bill Andersen]

*1.4

Sorry for a dumb question, but I'm working with my SIP
provider on a problem and I can't answer this question
for them.  They don't know Asterisk.

When I do a "sip show channels"

What is the "User/ANR" field?

Bill



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] "This e-mail is confidential" ... (was: Re: PRI D-Channel reconfiguration = crash asterisk?)

[Matt Watson]

That's fine... honestly I hate the message myself, however corporate policy is 
corporate policy so there isn't much of a point in discussing it.

That being said, the message does clearly say that the message is for the named 
recipients, in this particular case, the named recipient is a public mailing 
list.  By my action of sending a message to a public mailing list, one can say 
there is implied consent that it gets distributed to whomever the mailing list 
chooses on my behalf.

Thanks,

--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent: Tuesday, May 06, 2008 12:27 PM
To: Asterisk Users
Subject: [asterisk-users] "This e-mail is confidential" ... (was: Re: PRI 
D-Channel reconfiguration = crash asterisk?)

Matt Watson schrieb:

> Disclaimer Statement: This e-mail is confidential and is intended for the 
> above-named recipient(s) only. If you are not the intended recipient and/or 
> have received this e-mail in error, please notify us by telephone and delete 
> this e-mail from your system without retaining a copy in any form. Any 
> unauthorized use or disclosure of this e-mail is prohibited.

Your confidential e-mail is going to end up on Google ...

Regards,
  Philipp Kempgen

--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Mixmonitor recording issue

[Sanjay Rajdev]

I had a similar problem. In my case Asterisk was crashing due to MixMonitor() 
and then automatically restarting. 
I have never found a alternative solution to record the calls. 


Regards, 
Sanjay Rajdev 

- Original Message - 
From: "Rahul Yadav" <[EMAIL PROTECTED]> 
To: asterisk-users@lists.digium.com 
Sent: Tuesday, May 6, 2008 10:54:32 PM GMT +05:30 Chennai, Kolkata, Mumbai, New 
Delhi 
Subject: [asterisk-users] Mixmonitor recording issue 


Hi All 


I am using asterisk 1.4.18.1.My problem is that Mixmonitor is not recording 
complete recording. 
Suppose i got a call connected and talking for 3 minutes then mixmonitor 
records only 2 minutes of call. 
This problem happens randomly.Please help as i am suffering very much due to 
this problem. 

Rahul Yadav 
___ -- Bandwidth and Colocation 
Provided by http://www.api-digital.com -- asterisk-users mailing list To 
UNSUBSCRIBE or update options visit: 
http://lists.digium.com/mailman/listinfo/asterisk-users ___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] how do I set callerid for incoming iax?

[sean darcy]

Using 1.6-rc8.

In iax.conf on the calling box, I have:

[iax-out]
.
callerid = "sean" <447>

I even also put the same on called box.

But I can't seem to set the callerid:

exten =>_NXX,1,Answer()
exten =>_NXX,n,NoOp(${CALLERID(num)})


Answer("IAX2/iax-in-7", "") in new stack
  NoOp("IAX2/iax-in-7", "") in new stack

So how do I set callerid?

sean


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Running Asterisk as root

[Cesar Benjamin Garcia Martinez]

Hum. About the /var/run i do thats changes in the conf and the creation fo
/var/run into /var/lib/asterisk becouse Works :P. Yes, Ubuntu cleans al into
/var/run and that's my solution, I believe is possible touch something in
daemon for do work fine but I consider more simple make 2 folders and modify
one line

Maybe, the init.d script works well if comes from official package, I never
has installed asterisk from package, I prefer from sources.


On Mon, May 05, 2008 at 07:18:08PM -0500, Cesar Benjamin Garcia Martinez
wrote:
> Move to root:
> 
> sudo -s
> 
> type your passwd
> 
> and as root:
> 
> 
> Edit the file /etc/init.d/asterisk
> 
> And uncommet the two lines than sasys something like 
> 
> AST_USER="asterisk"
> AST_GROUP="asterisk"
> 
> You need to create the user asterisk on your system.
> 
> And create another symlink sh to bash:
> 
> cd /bin
> rm -f sh
> ln -s bash sh 

Why is that?

Debian / Ubuntu policy is that a script that is not posix sh should use
/bin/bash. Any script of Asterisk does not fit the policy and has not
bit shot^Wfixed yet?

The fix is to edit the ofending script:

#!/bin/sh  ->  #!/bin/bash

> 
> 
> 
> Edit your /etc/asterisk/asterisk.conf and replace the line:
> 
> astrundir => /var/run 
> 
> With:
> 
> astrundir => /var/lib/asterisk/var/run

/var/run/asterisk

Everything under /var/run is deleted at boot with Ubuntu, so the init.d
script should recreate that directory and give it proper permissions if
it does not exist.

(or use the one from the Asterisk package)

> 
> Create that folder:
> 
> mkdir -p /var/lib/asterisk/var/run

/var/run/asterisk, as mentioned above. and it should be created in the
init.d script .

> 
> and, chown to asterisk:asterisk the folders:
> 
> 
> /var/lib/asterisk/
> /usr/lib/asterisk/

No real need for /usr/lib/asterisk to be owned by Asterisk. It is
read-only. /usr is read-only, as you recall.

> /var/log/asterisk/
> 
> chown -Rv asterisk:asterisk /var/lib/asterisk/
# chown -Rv asterisk:asterisk /usr/lib/asterisk/
> chown -Rv asterisk:asterisk /var/log/asterisk/
> 
> that's all
> 
> 
> 
> Btw... delete the symlink sh -> dash into /bin

NOT

> 
> Start daemon
> 
> 
> /etc/init.d/asterisk start

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

__ NOD32 3078 (20080506) Information __

This message was checked by NOD32 antivirus system.
http://www.eset.com



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Running Asterisk as root

[Cesar Benjamin Garcia Martinez]

Really not.. if only you delete sh, yes, but i say make a symlink from
/bin/bash to /bin/sh

Ubuntu 7.04 and above, comes with the shell dash as sh, ubuntu 6.06 comes
with bash as sh, I got problems to start daemon, when sh points to dash..
safe_asterisk don's start...

I read 1.4.19 don't need anymore safe_asterisk, but, what about if I need
1.2.x ? or 1.4.18 ? I talk for example if I use unicall for E1 (MFCR2) when
I need that versions... someone do?

Oh!!! Now understand... I forget it... when I say about delete sh... i
forget say that is necessary, to create a symlink from /bin/bash to /bin/sh

I'm so sorry :$





-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Tzafrir Cohen
Enviado el: Lunes, 05 de Mayo de 2008 07:35 p.m.
Para: asterisk-users@lists.digium.com
Asunto: Re: [asterisk-users] Running Asterisk as root

On Mon, May 05, 2008 at 07:18:08PM -0500, Cesar Benjamin Garcia Martinez
wrote:

> Btw... delete the symlink sh -> dash into /bin

BAD

THAT BREAKS THE SYSTEM

(leaves it without /bin/sh, making half the scripts fail)

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

__ NOD32 3078 (20080506) Information __

This message was checked by NOD32 antivirus system.
http://www.eset.com



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Mixmonitor recording issue

[Rahul Yadav]

Hi All


I am using asterisk 1.4.18.1.My problem is that Mixmonitor is not recording
complete recording.
Suppose i got a call connected and talking for 3 minutes then mixmonitor
records only 2 minutes of call.
This problem happens randomly.Please help as i am suffering very much due to
this problem.

Rahul Yadav
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Performance issues

[Marcelo Freitas]

I'm not an expert, but FastAGI you use in a live channel over TCP connection 
... Most people agree it is better because doesn't spawn a new process every 
time it is called ... but they also suggest to do not use in the same server ...
 
DeadAGI you cannot use it in a live channel ...
 
By the way, a2billing is a nice software ... Areski does a great job ... I 
neither use it a lot nor in a production high volume environment but I liked it 
...





- Original Message -
Subject: [asterisk-users] Performance issues
From: "chetherston miles" >;[EMAIL PROTECTED]>
Date: Tue, May 6, 2008 10:20



Hello,


We are thinking in use asterisk-java to an billing solution, wich is the better 
choice, and if someone could give us a understandable description about the 
difference between DeadAGI and FastAGI, i found a very interesting project  
called asterisk2billing and they use DeadAGI, anyway wich one scale better?


And there is a tool for test performance?


Thanks,
Roberto

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] "This e-mail is confidential" ... (was: Re: PRI D-Channel reconfiguration = crash asterisk?)

[Philipp Kempgen]

Matt Watson schrieb:

> Disclaimer Statement: This e-mail is confidential and is intended for the 
> above-named recipient(s) only. If you are not the intended recipient and/or 
> have received this e-mail in error, please notify us by telephone and delete 
> this e-mail from your system without retaining a copy in any form. Any 
> unauthorized use or disclosure of this e-mail is prohibited.

Your confidential e-mail is going to end up on Google ...

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] finding Asterisk user group users and enthusiasts in the (Salt Lake City Utah, Chicago IL, Boston MA, Tampa FL)

[Ming Yong]

Hi all,
Are there Asterisk user groups and organized Asterisk enthusiasts in
the following cities: Salt Lake City Utah, Chicago IL, Boston MA,
Tampa FL? Looking to meet up with folks in these cities during the
month of May when I am in those cities.

Drop me a mail if you are looking to shoot the breeze and discuss all
topics asterisk & unified communications. Thanks

Ming

-- 
Ming Yong
CEO, www.voiceroute.org
Druid - Open Source Unified Communications
DID: +1-866-915-2407 ext 301
SIP/email: [EMAIL PROTECTED]

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Unicall - How to automatically block collect calls

[Luis Antonio Prata Barbosa]

I`m using collect call blocking with astunicall from moises and it`s working
properly.

UC_CATEGORY=INTERNATIONAL_DATA  (for brazilian users) indicates a collect
call.

I spent a long time searching a way to do it, but it was only possible with
moises code.

Thank you.

Luis A P Barbosa.

2008/5/6 Oscar Patricio <[EMAIL PROTECTED]>:

> Hi!
>
> Thank you for your answer.
>
> However I would like to know, if there is any other possibility of
> making this work, with older unicall versions.
>
> The reason I ask, is because it's not easy for me to upgrade the
> asterisk and unicall version on my machine, since it is a production
> machine, and I would consider a software upgrade really only if no other
> solution is possible.
>
> Thanks in advance for your answer.
>
> Best regards,
>
> Óscar Patrício
>
>
> Moises Silva escreveu:
>  > The latest version of the driver included in
> > http://www.moythreads.com/astunicall/ comes with a change that will
> > set the variable UC_CATEGORY in your dialplan, Brasil has a special
> > category for those calls, don't remember the name that will show up,
> > but you can make a couple of tests and then drop any call with that
> > specific category.
> >
> > Moy
> >
> > On Mon, May 5, 2008 at 9:14 AM, Oscar Patricio <[EMAIL PROTECTED]>
> wrote:
> >
> >> Hi!
> >>
> >>  I am using asterisk with unicall in brasil.
> >>
> >>  Everything was working fine, but now we want to set up a way to
> >>  automatically drop collect calls, because we have an IVR answering all
> >>  calls automatically!
> >>
> >>  Can you tell me, what I have to configure to block collect calls in
> the
> >>  asterisk?
> >>
> >>  Thank you!
> >>  Best regards,
> >>
> >>  Óscar Patrício
> >>
> >>  ___
> >>  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>
> >>  asterisk-users mailing list
> >>  To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> >
> >
> >
> >
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PRI D-Channel reconfiguration = crash asterisk?

[Matt Watson]

My bad, I also should of mentioned...

That was on Asterisk 1.4.18 and Zaptel 1.4.10

Using a TE220B

--
Matt

From: Matt Watson
Sent: Tuesday, May 06, 2008 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: PRI D-Channel reconfiguration = crash asterisk?

Hello,

I just had to have MTS Allstream fix a new T1 install that we have that we 
aren't running in production yet, but it is attached to a production  machine.

Apparently they setup the T1 with only a 1 B-channel (how useful!)  even though 
we had ordered it fully loaded with 23.  Anyways... they just reconfigured the 
T1 to activate all the T1 channels and this is what I got on my * console:

  == Primary D-Channel on span 1 down
nelson*CLI>
Disconnected from Asterisk server

^^ asterisk crashed.

Unfortunately I didn't have * setup on this box to dump a core file, so the 
only additional debug info I can provide is from my asterisk log file:

 [May  6 11:42:23] VERBOSE[16656] logger.c:   == Primary D-Channel on span 1 
down
[May  6 11:42:23] WARNING[16656] chan_zap.c: No D-channels available!  Using 
Primary channel 24 as D-channel anyway!
[May  6 11:42:23] WARNING[16656] chan_zap.c: The PRI Call have not been 
destroyed

Those are they only 3 relevant lines in the log file.


--
Matt

Disclaimer Statement: This e-mail is confidential and is intended for the 
above-named recipient(s) only. If you are not the intended recipient and/or 
have received this e-mail in error, please notify us by telephone and delete 
this e-mail from your system without retaining a copy in any form. Any 
unauthorized use or disclosure of this e-mail is prohibited.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] PRI D-Channel reconfiguration = crash asterisk?

[Matt Watson]

Hello,

I just had to have MTS Allstream fix a new T1 install that we have that we 
aren't running in production yet, but it is attached to a production  machine.

Apparently they setup the T1 with only a 1 B-channel (how useful!)  even though 
we had ordered it fully loaded with 23.  Anyways... they just reconfigured the 
T1 to activate all the T1 channels and this is what I got on my * console:

  == Primary D-Channel on span 1 down
nelson*CLI>
Disconnected from Asterisk server

^^ asterisk crashed.

Unfortunately I didn't have * setup on this box to dump a core file, so the 
only additional debug info I can provide is from my asterisk log file:

 [May  6 11:42:23] VERBOSE[16656] logger.c:   == Primary D-Channel on span 1 
down
[May  6 11:42:23] WARNING[16656] chan_zap.c: No D-channels available!  Using 
Primary channel 24 as D-channel anyway!
[May  6 11:42:23] WARNING[16656] chan_zap.c: The PRI Call have not been 
destroyed

Those are they only 3 relevant lines in the log file.


--
Matt

Disclaimer Statement: This e-mail is confidential and is intended for the 
above-named recipient(s) only. If you are not the intended recipient and/or 
have received this e-mail in error, please notify us by telephone and delete 
this e-mail from your system without retaining a copy in any form. Any 
unauthorized use or disclosure of this e-mail is prohibited.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Benoit Plessis]

Steve Totaro a écrit :
> On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote:
>   
>> lordfuknowsyou a écrit :
>>
>> 
>>> Vinícius Fontes wrote:
>>>   
>>  >
>>  > I use 1.4.18 with no problems. We have quite a few users(125 total
>>  > between branches), but the call volume at the most has been around 15
>>  > active calls at a time.
>>  >
>>  Any IAX2 phone or mostly SIP ?
>>  Do you use Call Queues ?
>>
>>  We have less user than that, less concurrent call but quite a few
>>  crash/deadlocks
>>
>> 
>
> Try SIP only if you can and report back.  I think you will confirm
> what is pretty much a silent consensus (even among Digium Devs).
>
> Thanks,
> Steve Totaro
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   

I've tried SIP only but i already got one 'stuck' Queue member:
   Members:
  Local/[EMAIL PROTECTED] with penalty 10 (dynamic) (In use) has taken 1 
calls (last was 45 secs ago)
  Local/[EMAIL PROTECTED] with penalty 20 (dynamic) (Not in use) has taken 
no calls yet
   Callers:
  1. Zap/10-1 (wait: 0:18, prio: 0)

[May  6 17:48:35] NOTICE[2047]: app_queue.c:2152 wait_for_answer: No one 
is answering queue 'support' (1/0/0)
asterix*CLI> core show channels
Channel  Location State   
Application(Data)
SIP/rtournier-081ef2 (None)   Up  Bridged 
Call(Local/[EMAIL PROTECTED]

but the other end of the bridged call is long gone



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Anthony Francis]

Tilghman Lesher wrote:
> On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
>   
>> 5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
>> 
>>> On Monday 05 May 2008 11:24, Johansson Olle E wrote:
>>>   
 5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
 
> On Monday 05 May 2008 09:45, Johansson Olle E wrote:
>   
>> Another issue that we need to fix with the MYSQL driver is that
>> we're
>> lacking a connection pool. Everything seems to be handled over one
>> connection to Mysql, which causes issues.
>> 
> That's not true.  The MYSQL app generally uses multiple connections,
> one
> for each channel.  The only way one might use only a single
> connection is
> by using a global variable to store a single connection id, but that
> method
> is not documented anywhere, AFAIK.
>   
 You talk about the Mysql APP, but is this the case with the Realtime
 driver as well?
 
>>> No, the native Realtime driver uses a single connection.  The ODBC
>>> Realtime
>>> driver generally uses a single connection but can be configured to
>>> use a
>>> separate connection for each query.
>>>   
>> So, we're back to where we started. A developer that can help us with
>> a connection
>> pool or a separate connection for each query would be a Nice Thing (TM).
>> 
>
> What issues are you specifically seeing that merit using multiple
> connections?
>
>   
I can specify an issue that would merit multiple connections, if the 
link to your db goes away Asterisk likes to freeze writing CDRs.
I have a few remote * servers that this happens to. My solution so far 
has been to record CDR's to a local DB and then have a
perl script that attempts to move them over to my transaction DB. I 
would suggest this solution to anyone who depends on their CDR records.

-- 
Thank you and have any kind of day you want,

Anthony Francis
Rockynet VOIP


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Benoit Plessis]

Tilghman Lesher a écrit :
> On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote:
>   
>> Tilghman Lesher a écrit :
>> 
>>> On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
>>>   
 lordfuknowsyou a écrit :
 
> Vinícius Fontes wrote:
>
> I use 1.4.18 with no problems. We have quite a few users(125 total
> between branches), but the call volume at the most has been around 15
> active calls at a time.
>   
 Any IAX2 phone or mostly SIP ?
 Do you use Call Queues ?

 We have less user than that, less concurrent call but quite a few
 crash/deadlocks
 
>>> Have you reported these issues on the bugtracker?
>>>   
>> Well, the problem is finding usefull data to report.
>>
>> I've 4 core dumps thats show differents things:
>>
>> two seems to be related to ControlPlayback:
>> #0  0xb705b658 in strcasecmp () from /lib/tls/libc.so.6
>> #1  0x0809c579 in ast_readframe ()
>> #2  0x0809defc in ast_streamfile ()
>> #3  0x0805e786 in ast_control_streamfile ()
>> #4  0xb698be5c in ?? () from
>> /usr/lib/asterisk/modules/app_controlplayback.so
>> #5  0x08298700 in ?? ()
>> #6  0xb470aec0 in ?? ()
>> #7  0xb698c1fc in ?? () from
>> /usr/lib/asterisk/modules/app_controlplayback.so
>> #8  0xb698c1fa in ?? () from
>> /usr/lib/asterisk/modules/app_controlplayback.so
>> #9  0x in ?? ()
>> 
>> 
>
> I'd love to see a 'bt full' on this one.
>   
Here it is, but since the AsteriskNow release has stripped the binary
i fear it won't be of much use:

#0  0xb705b658 in strcasecmp () from /lib/tls/libc.so.6
No symbol table info available.
#1  0x0809c579 in ast_readframe ()
No symbol table info available.
#2  0x0809defc in ast_streamfile ()
No symbol table info available.
#3  0x0805e786 in ast_control_streamfile ()
No symbol table info available.
#4  0xb698be5c in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#5  0x08298700 in ?? ()
No symbol table info available.
#6  0xb470aec0 in ?? ()
No symbol table info available.
#7  0xb698c1fc in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#8  0xb698c1fa in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#9  0x in ?? ()
No symbol table info available.
#10 0x in ?? ()
No symbol table info available.
#11 0x in ?? ()
No symbol table info available.
#12 0x0bb8 in ?? ()
No symbol table info available.
#13 0x2f727669 in ?? ()
No symbol table info available.
#14 0x65696c63 in ?? ()
No symbol table info available.
#15 0x2f73746e in ?? ()
No symbol table info available.
#16 0x6a6e6f62 in ?? ()
No symbol table info available.
#17 0x2d72756f in ?? ()
No symbol table info available.
#18 0x6e656962 in ?? ()
No symbol table info available.
#19 0x756e6576 in ?? ()
No symbol table info available.
#20 0x6568632d in ?? ()
No symbol table info available.
#21 0x6f702d7a in ?? ()
No symbol table info available.
#22 0x62726577 in ?? ()
No symbol table info available.
#23 0x6974756f in ?? ()
No symbol table info available.
#24 0x2d657571 in ?? ()
No symbol table info available.
#25 0x76726573 in ?? ()
No symbol table info available.
#26 0x73656369 in ?? ()
No symbol table info available.
#27 0x696c632d in ?? ()
No symbol table info available.
#28 0x00746e65 in ?? ()
No symbol table info available.
#29 0x0001 in ?? ()
No symbol table info available.
#30 0xb470af20 in ?? ()
No symbol table info available.
#31 0x081aa084 in ?? ()
No symbol table info available.
#32 0x001b in ?? ()
No symbol table info available.
#33 0x0025 in ?? ()
No symbol table info available.
#34 0x0028 in ?? ()
No symbol table info available.
#35 0xb698c1fc in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#36 0x in ?? ()
No symbol table info available.
#37 0xb698c1fa in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#38 0x0829c4a8 in ?? ()
No symbol table info available.
#39 0x0bb8 in ?? ()
No symbol table info available.
#40 0x in ?? ()
No symbol table info available.
#41 0xb470aec0 in ?? ()
No symbol table info available.
#42 0x in ?? ()
No symbol table info available.
#43 0xb698c1fc in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#44 0xb698c1fa in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#45 0x in ?? ()
No symbol table info available.
#46 0x in ?? ()
No symbol table info available.
#47 0x in ?? ()
No symbol table info available.
#48 0x in ?? ()
No symbol table info available.
#49 0x08298700 in ?? ()
No symbol table info available.
#50 0xb705b631 in strcasecmp () from /lib/tls/libc.so.6
No symbol table info available.
#51 0x080c8740 in pbx_substitute_variables_helper ()
No symbol table info available.
#52 0x080cd

Re: [asterisk-users] Asterisk in Production

[Norman Franke]

On May 6, 2008, at 10:20 AM, [EMAIL PROTECTED]  
wrote:



I'm wondering what version of asterisk people use in production
environnement ?
on which distribution ?

And what is your setup like ?

We are actually running an AsteriskNow appliance with asterisk  
1.4.18.1

and it's quite unstable.



I'm running 1.4.19 and it has been pretty stable. Anything before  
1.4.19, however, I found was embarrassingly unstable. I'd often get  
several crashes within an hour. However, since moving to 19 things  
have been better.


I don't run Queues, though, but I do run a custom derivative of  
Queues that fixed some bugs and greatly enhanced its usability for us.


We do tens of thousands of calls per day (mostly inbound) running on  
under Debian, although I had to upgrade the kernel to 2.6.23.11 in  
order to get ztdummy to work on my HP DL380. CPU load remains rather  
low. We are all SIP, no zaptel.


I used to run IAX2 between my three servers (one's a backup and for  
testing, the other handles desk phones and ATAs), but found IAX2  
very, very unreliable. It would hang Asterisk, crash, etc. I just  
replaced it with SIP (and turned off the module) and those problems  
went away.


Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Tilghman Lesher]

On Tuesday 06 May 2008 09:02:47 Steve Totaro wrote:
> On Tue, May 6, 2008 at 9:35 AM, Tilghman Lesher
>
> <[EMAIL PROTECTED]> wrote:
> > On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote:
> >  > While these may not be popular opinions, I still ask, what does
> >  > SwitchVox use? What do some of the guys around here that setup large
> >  > systems use?  Is ABE even using 1.4 yet?
> >
> >  Yes, ABE version C (in release for several months) is using the 1.4
> > codebase.
>
> Does "In Release" equate to "In the Wild" or "In Many Production
> Installations" ?

I sense that there are quite a few people who are running version C and a few
holdouts still running B, but that's based on a wet-finger-in-the-wind
estimation, not on any industry surveys.

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Tilghman Lesher]

On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote:
> Tilghman Lesher a écrit :
> > On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
> >> lordfuknowsyou a écrit :
> >>> Vinícius Fontes wrote:
> >>>
> >>> I use 1.4.18 with no problems. We have quite a few users(125 total
> >>> between branches), but the call volume at the most has been around 15
> >>> active calls at a time.
> >>
> >> Any IAX2 phone or mostly SIP ?
> >> Do you use Call Queues ?
> >>
> >> We have less user than that, less concurrent call but quite a few
> >> crash/deadlocks
> >
> > Have you reported these issues on the bugtracker?
>
> Well, the problem is finding usefull data to report.
>
> I've 4 core dumps thats show differents things:
>
> two seems to be related to ControlPlayback:
> #0  0xb705b658 in strcasecmp () from /lib/tls/libc.so.6
> #1  0x0809c579 in ast_readframe ()
> #2  0x0809defc in ast_streamfile ()
> #3  0x0805e786 in ast_control_streamfile ()
> #4  0xb698be5c in ?? () from
> /usr/lib/asterisk/modules/app_controlplayback.so
> #5  0x08298700 in ?? ()
> #6  0xb470aec0 in ?? ()
> #7  0xb698c1fc in ?? () from
> /usr/lib/asterisk/modules/app_controlplayback.so
> #8  0xb698c1fa in ?? () from
> /usr/lib/asterisk/modules/app_controlplayback.so
> #9  0x in ?? ()
> 

I'd love to see a 'bt full' on this one.

> One is pretty generic:
> #0  0x0809c9bc in ast_closestream ()
> #1  0x08085d91 in ast_hangup ()
> #2  0x080cd3d8 in pbx_builtin_setvar_helper ()
> #3  0x080cf08e in ast_pbx_outgoing_exten ()
> #4  0x080fde65 in ast_inet_ntoa ()
> #5  0xb7eec560 in start_thread () from /lib/tls/libpthread.so.0
> #6  0xb703667e in clone () from /lib/tls/libc.so.6

Ditto, bt full.

> and the latest is thread/iax2 related:
> #0  0xb7ee71c7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
> #1  0xb562a969 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so
> #2  0x0079 in ?? ()
> #3  0x in ?? ()
> #4  0xb547a148 in ?? ()
> #5  0x080f0508 in ast_sched_add_variable ()
> #6  0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so
> #7  0x0012 in ?? ()
> 

This one may need valgrind to track down the problem, but please be sure
to run 1.4.18 or later, as we've already fixed a problem that produced
backtraces similar to this.

> But my main problem is when the system just froze,
> it start mostly by the Queue not working anymore, with member stuck in
> 'in use' stack (should not happen
> with IAX2 agent IIRC, given that we had to build macros using GROUP() to
> detect in use IAX2 agent)
> Then the console (asterisk -rcTvvv) start to freeze (completion doesn't
> work, message stop from being displayed
> and even command output is lost).
>
> And i'm reading http://www.asterisk.org/developers/bug-guidelines which
> speak of using SVN trunk version of asterisk,
> thing i'm not really eager to try on a live system...

I don't see anywhere on that page that recommends that you try SVN trunk,
only the latest SVN (which is probably confusing, but what is meant is to try
the latest SVN in the 1.4 branch, which is the release branch.  Release
candidates and releases are tagged directly off that branch).

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working ...

[Steve Hickel]

These are the instructions that I followed. I did managed to get the
fast busy to go away, but the RDNIS simply does not seem to work. These
are the instructions that I followed on this project. I have run out of
time trying to get Call Manager 4.x to talk to Asterisk 1.4. 

http://www.voip-info.org/wiki/index.php?page_id=2596#editcomments

These instructions although a good start, simply lack the pictures or
images to set up CCM properly, and because of the coding change from
earlier versions, this just doesn't seem to allow voice mail to work.

I have learned a lot about asterisk, but am frustrated by this
experience.

Thanks Sean for the info about the change of the rdnis command format.

Kind regards,

Steve

On Mon, 2008-05-05 at 23:33 -0400, Steve Hickel wrote:
> Sean,
> 
> Here is what I changed. Now I have a fast busy... 
> 
> Steve
> 
>  [demo]
>   exten=s,1,Wait(1)
>   exten=s,n,Answer
>   exten=s,n,Set(TIMEOUT(digit)=5)
>   exten=s,n,Set(TIMEOUT(response)=10)
>   exten=s,n(restart),BackGround(demo-congrats)
>   exten=s,n(instruct),BackGround(demo-instruct)
>   exten=s,n,WaitExten
>   exten=2,1,BackGround(demo-moreinfo)
>   exten=2,n,Goto(s,instruct)
>   exten=3,1,Set(LANGUAGE()=fr)
>   exten=3,n,Goto(s,restart)
>   exten=1000,1,Goto(default,s,1)
>   exten=1234,1,Playback(transfer,skip)
>   exten=1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})
>   exten=1235,1,Voicemail(1234,u)
>   exten=1236,1,Dial(Console/dsp)
>   exten=1236,n,Voicemail(1234,b)
>   exten=#,1,Playback(demo-thanks)
>   exten=#,n,Hangup
>   exten=t,1,Goto(#,1)
>   exten=i,1,Playback(invalid)
>   exten=500,1,Playback(demo-abouttotry)
>   exten=500,n,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
>   exten=500,n,Playback(demo-nogo)
>   exten=500,n,Goto(s,6)
>   exten=600,1,Playback(demo-echotest)
>   exten=600,n,Echo
>   exten=600,n,Playback(demo-echodone)
>   exten=600,n,Goto(s,6)
>   exten=76245,1,Macro(page,SIP/Grandstream1)
>   exten=_7XXX,1,Macro(page,SIP/${EXTEN})
>   exten=7999,1,Set(TIMEOUT(absolute)=60)
>   exten=7999,2,Page(Local/[EMAIL PROTECTED]&Local/[EMAIL 
> PROTECTED]&Local/[EMAIL PROTECTED]/n|d)
>   exten=,1,VoicemailMain
>   exten=,n,Goto(s,6)
> 
>   [general]
>   static=yes
>   writeprotect=no
>   clearglobalvars=no
>   autofallthrough=yes
>   priorityjumping=no
>   
> 
>   [default]
>   exten=_230,1,SetCallerID(${EXTEN:3})
>   exten=_230,2,Dial(SIP/[EMAIL PROTECTED])
>   exten=_230,3,Answer
>   exten=_230,4,Wait,1
>   exten=_230,5,Hangup
>   exten=_231,1,SetCallerID(${EXTEN:3})
>   exten=_231,2,Dial(SIP/[EMAIL PROTECTED])
>   exten=_231,3,Answer
>   exten=_231,4,Wait,1
>   exten=_231,5,Hangup
>   exten=,1,VoiceMailMain
> 
>   [incoming]
>   exten=,1,GotoIf($[${CALLERID(rdnis)}]?2:400)
>   exten=,2,MailboxExists(${CALLERID(rdnis)[EMAIL PROTECTED])
>   exten=,3,Congestion
>   exten=,103,Voicemail(su${CALLERID(rdnis)}
>   exten=,104,Playback(vm-goodbye)
>   exten=,105,Hangup
>   exten=,400,VoicemailMain
> 
> __
> From: Sean Dennis [mailto:[EMAIL PROTECTED]
> To: [EMAIL PROTECTED], Asterisk Users Mailing List -
> Non-Commercial Discussion
> [mailto:[EMAIL PROTECTED]
> Sent: Mon, 05 May 2008 17:58:32 -0400
> Subject: Re: [asterisk-users] Call manager using Asterisk as
> voicemail server (SIP) not working ...
> 
> Steve Hickel wrote:
> > I have sip set up on Callmanager 4.x. When others call my
> ext of 2016 on
> > ccm after a busy or no answer, asterisk voice mail answers
> by saying,
> > "Mailbox  password." I want it to put them into my
> mailbox so they
> > can leave a message. Somehow I must be missing something...
> Please
> > help! 
> >
> > I have spent 19 hours easy on trying to figure this one
> out. 
> >
> > SIP DN is  on CCM 
> > VOICEMAIL on Asterisk is . 
> >
> > Here is my sip.conf: 
> >
> > [general] 
> > context=default 
> > allowoverlap=no 
> > bindport=5060 
> > bindaddr=0.0.0.0 
> > srvlookup=yes 
> > allowexternaldomains=yes 
> > allowexternalinvites=no 
> > allowguest=yes 
> > allowsubscribe=no 
> > allowtransfer=yes 
> > alwaysauthreject=no 
> > autodomain=no 
> > callevents=no 
> > compactheaders=no 
> > dumphistory=no 
> > g726nonstandard=no 
> > ignoreregexpire=no 
> > jbenable=no 
> > jbforce=no 
> > jblog=no 
> > maxcallbitrate=384 
> > maxexpiry=3600 
> > minexpiry=60 
> > nat=no 
> > notifyringing=no 
> > pedantic=no 
> > promis

Re: [asterisk-users] Performance issues

[Matt Watson]

Google is awesome

http://www.voip-info.org/wiki-Asterisk+AGI

--
Matt

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chetherston miles
Sent: Tuesday, May 06, 2008 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Performance issues

Hello,

We are thinking in use asterisk-java to an billing solution, wich is the better 
choice, and if someone could give us a understandable description about the 
difference between DeadAGI and FastAGI, i found a very interesting project  
called asterisk2billing and they use DeadAGI, anyway wich one scale better?

And there is a tool for test performance?

Thanks,
Roberto
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Matt Watson]

I'm using 1.4.18 in production on 2 boxes... one of which being a custom built 
desktop basically, the other being a Dell 1950 III

We are in a migration phase to the Dell box, right now the 1st box is doing 
nothing more than being a PSTN gateway to some FXO lines... basically waiting 
for numbers to be ported off the analog lines and onto the new T1 which is 
connected to the Dell box.

We have the 2 boxes connected by IAX2 trunk.

I had 1.4.19 and 1.4.19.1 running on the Dell box, but it started giving me a 
lot of trouble with the IAX2 trunk, the trunk would (seemingly) go into 
UNREACHABLE status and never come back without restarting asterisk (reload, or 
iax2 reload wouldn’t cut it).  Also, occasionally people trying to make 
outbound calls (and this probably happened on inbound as well), would get a 
"all circuits are busy" message because of the IAX2 channel driver reporting 
congestion on the trunk even though it was up (and not congested)

Unfortunately as this is a production box I didn’t really have time to try and 
debug it so I simply downgraded to .18 since it has proven itself well on the 
1st box.  So far since I;ve downgraded to .18 I haven’t had any problems.

Both installs I have running ontop of Gentoo (wouldn’t recommend it if you are 
new to Linux or don’t like tweak-ability).

That all being said, I'll probably give .20 a try when its released, as I see 
there have been some IAX2 bug fixes in it... but also by the time .20 is 
released I probably will have retired the box being used as a PSTN gateway and 
won’t need the IAX2 trunk anymore.

--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vinícius Fontes
Sent: Tuesday, May 06, 2008 8:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk in Production ?

There were some really unstable Asterisk releases in the 1.4 branch. I 
personally use 1.4.13 or 1.4.15 in production. Every single time I tried 1.4.16 
or higher I had problems.



Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.

- "Steve Totaro" <[EMAIL PROTECTED]> escreveu:

> On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis <[EMAIL PROTECTED]>
> wrote:
> >
> >  Hi,
> >
> >  I'm wondering what version of asterisk people use in production
> >  environnement ?
> >  on which distribution ?
> >
> >  And what is your setup like ?
> >
> >  We are actually running an AsteriskNow appliance with asterisk
> 1.4.18.1
> >  and it's quite unstable.
> >  We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX
> destroy
> >  deadlock"
> >  and now that we have added a Queue, it's worse than ever. The queue
> goes
> >  stuck quite often
> >  (agent are stuck in 'In use' state and if they logoff they can't
> log-in
> >  till an asterisk restart).
> >
> >
> >  regards
> >
>
> I am personally a proponent of Asterisk 1.2.X as I see more and more
> fatal bugs in the 1.4.X code come up on the lists as well as IAX2
> bugs.  I constantly hear "Asterisk 1.4.whatever is much better, but
> the bugs coming out are not just unexpected behavior that one could
> live with, they are segfaults, system crashes, modules not getting
> installed (Zaptel).
>
> I use SIP since I have seen quite a few issues with IAX2 that were
> solved by simply switching to SIP.
>
> The above two yield solid systems under heavy load for me.  OS is not
> so important I do not believe.  I have some running FC8 and more
> running CentOS, both rock solid.  I think the general consensus on OS
> is use what you are most familiar with.
>
> While these may not be popular opinions, I still ask, what does
> SwitchVox use?  What do some of the guys around here that setup large
> systems use?  Is ABE even using 1.4 yet?  All I see in the ABE
> release
> notes is 1.2 although I have heard that ABE should be running 1.4
> "Very Soon" many many moons ago
> http://www.digium.com/en/docs/ABE/README .  So either Digium doesn't
> trust 1.4 enough to use it for ABE or the README is out of date.
>
> Thanks,
> Steve Totaro
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Mindaugas Kezys]

Hello,

Our company did 200+ installations around the globe and had no issues with
stability with correct Asterisk version.

We used most of 1.4. As far as I remember 1.4.16 has some nasty bugs along
with 1.4.19.x (SIP + realtime).

So current stable is 1.4.18.1 (for us).

For load check: http://wiki.kolmisoft.com/index.php/How_fast_MOR_can_perform

It shows how our billing application performs on top of Asterisk (2049
channels) and we can push it even further with some improvements.

We DO NOT RESTART our Asterisk installations daily or weekly. They work for
months.

Regards,
Mindaugas Kezys
http://www.kolmisoft.com



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Benoit Plessis
> Sent: Tuesday, May 06, 2008 2:39 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Asterisk in Production ?
> 
> 
> Hi,
> 
> I'm wondering what version of asterisk people use in production
> environnement ?
> on which distribution ?
> 
> And what is your setup like ?
> 
> We are actually running an AsteriskNow appliance with asterisk 1.4.18.1
> and it's quite unstable.
> We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX destroy
> deadlock"
> and now that we have added a Queue, it's worse than ever. The queue
> goes
> stuck quite often
> (agent are stuck in 'In use' state and if they logoff they can't log-in
> till an asterisk restart).
> 
> 
> regards
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Benoit Plessis]

Tilghman Lesher a écrit :
> On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
>   
>> lordfuknowsyou a écrit :
>> 
>>> Vinícius Fontes wrote:
>>>
>>> I use 1.4.18 with no problems. We have quite a few users(125 total
>>> between branches), but the call volume at the most has been around 15
>>> active calls at a time.
>>>   
>> Any IAX2 phone or mostly SIP ?
>> Do you use Call Queues ?
>>
>> We have less user than that, less concurrent call but quite a few
>> crash/deadlocks
>> 
>
> Have you reported these issues on the bugtracker?
>
>   
Well, the problem is finding usefull data to report.

I've 4 core dumps thats show differents things:

two seems to be related to ControlPlayback:
#0  0xb705b658 in strcasecmp () from /lib/tls/libc.so.6
#1  0x0809c579 in ast_readframe ()
#2  0x0809defc in ast_streamfile ()
#3  0x0805e786 in ast_control_streamfile ()
#4  0xb698be5c in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
#5  0x08298700 in ?? ()
#6  0xb470aec0 in ?? ()
#7  0xb698c1fc in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
#8  0xb698c1fa in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
#9  0x in ?? ()


One is pretty generic:
#0  0x0809c9bc in ast_closestream ()
#1  0x08085d91 in ast_hangup ()
#2  0x080cd3d8 in pbx_builtin_setvar_helper ()
#3  0x080cf08e in ast_pbx_outgoing_exten ()
#4  0x080fde65 in ast_inet_ntoa ()
#5  0xb7eec560 in start_thread () from /lib/tls/libpthread.so.0
#6  0xb703667e in clone () from /lib/tls/libc.so.6


and the latest is thread/iax2 related:
#0  0xb7ee71c7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
#1  0xb562a969 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so
#2  0x0079 in ?? ()
#3  0x in ?? ()
#4  0xb547a148 in ?? ()
#5  0x080f0508 in ast_sched_add_variable ()
#6  0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so
#7  0x0012 in ?? ()



But my main problem is when the system just froze,
it start mostly by the Queue not working anymore, with member stuck in 
'in use' stack (should not happen
with IAX2 agent IIRC, given that we had to build macros using GROUP() to 
detect in use IAX2 agent)
Then the console (asterisk -rcTvvv) start to freeze (completion doesn't 
work, message stop from being displayed
and even command output is lost).

And i'm reading http://www.asterisk.org/developers/bug-guidelines which 
speak of using SVN trunk version of asterisk,
thing i'm not really eager to try on a live system...




___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Sanjay Rajdev]

We are using Asterisk 1.4.13 in production, were we have almost 30 SIP users on 
a Asterisk box, we are also using IAX to communicate between main Asterisk 
server and the other. we use Queues, Conference too. 

Regards, 
Sanjay Rajdev 

- Original Message - 
From: "Benoit Plessis" <[EMAIL PROTECTED]> 
To: asterisk-users@lists.digium.com 
Sent: Tuesday, May 6, 2008 5:08:37 PM GMT +05:30 Chennai, Kolkata, Mumbai, New 
Delhi 
Subject: [asterisk-users] Asterisk in Production ? 


Hi, 

I'm wondering what version of asterisk people use in production 
environnement ? 
on which distribution ? 

And what is your setup like ? 

We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 
and it's quite unstable. 
We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX destroy 
deadlock" 
and now that we have added a Queue, it's worse than ever. The queue goes 
stuck quite often 
(agent are stuck in 'In use' state and if they logoff they can't log-in 
till an asterisk restart). 


regards 

___ 
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- 

asterisk-users mailing list 
To UNSUBSCRIBE or update options visit: 
http://lists.digium.com/mailman/listinfo/asterisk-users 
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Performance issues

[chetherston miles]

Hello,
We are thinking in use asterisk-java to an billing solution, wich is the
better choice, and if someone could give us a understandable description
about the difference between DeadAGI and FastAGI, i found a very interesting
project  called asterisk2billing and they use DeadAGI, anyway wich one scale
better?

And there is a tool for test performance?

Thanks,
Roberto
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Benoit Plessis]

Steve Totaro a écrit :
> On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote:
>   
>>  Any IAX2 phone or mostly SIP ?
>>  Do you use Call Queues ?
>>
>>  We have less user than that, less concurrent call but quite a few
>>  crash/deadlock
>
> Try SIP only if you can and report back.  I think you will confirm
> what is pretty much a silent consensus (even among Digium Devs).
>   
Hi, that's what i was planning seeing all thoses answers.
We initialy choosed IAX2 for the sendurl() support but
i'll set-up a test periode in SIP-only to compare.

> Thanks,
> Steve Totaro
>   
Thanks to you


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[lordfuknowsyou]

Steve Totaro wrote:
>>  I use 1.4.18 with no problems. We have quite a few users(125 total
>>  between branches), but the call volume at the most has been around 15
>>  active calls at a time.
>>
>>
>> 
>
> I would classify that as "Light to Medium Call Volume" or "SMB".
>
> Let me clarify what I consider "High Call Volume".  ~400 simultaneous
> calls, all SIP or 95 on a box doing quad PRI to SIP gateway duty.
> 15k+ calls a day lasting an average of fifteen minutes.
>
> Thanks,
> Steve Totaro
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>   
I agree that it is SMB, never said I was a telco ;]  just in production 
on 1.4.18. we do use sip,iax2 and pri. Our calls do last extended 
periods of time, especially when there are conferences. No call ques, 
and we do realtime voicemail,sip and iax to allow tennants web 
interfaces into the system through the standard 3 tiers.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[lordfuknowsyou]

Benoit Plessis wrote:
> lordfuknowsyou a écrit :
>   
>> Vinícius Fontes wrote:
>>   
>> I use 1.4.18 with no problems. We have quite a few users(125 total 
>> between branches), but the call volume at the most has been around 15 
>> active calls at a time.
>>   
>> 
> Any IAX2 phone or mostly SIP ?
> Do you use Call Queues ?
>
> We have less user than that, less concurrent call but quite a few 
> crash/deadlocks
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
We use SIP and IAX2, we also do fax 2 email using spandsp and rx/txfax. 
I did have a problem with libpri during the upgrade and had to roll back 
to the one I was using prior.

hth

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Jay R. Ashworth]

On Tue, May 06, 2008 at 10:01:54AM -0400, Jay R. Ashworth wrote:
> On Tue, May 06, 2008 at 01:38:37PM +0200, Benoit Plessis wrote:
> > I'm wondering what version of asterisk people use in production 
> > environnement ? on which distribution ?
> > 
> > And what is your setup like ?
> 
> Well, we're running a cluster of about 15 boxes or so with Slack 10 or
> 12 and 1.2.17(?, either 14 or 17) and VICIdial.  Yeah, you'd call it
> production.  :-)

Sorry: we're running zap channels at all the edges (Digium and Sangoma
quad-T cards, primarily), and IAX2 in the middle.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[SIP]

Tilghman Lesher wrote:
> On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote:
>   
>> All I see in the ABE release 
>> notes is 1.2 although I have heard that ABE should be running 1.4
>> "Very Soon" many many moons ago
>> http://www.digium.com/en/docs/ABE/README .  So either Digium doesn't
>> trust 1.4 enough to use it for ABE or the README is out of date.
>> 
>
> The first clue should be that the copyright listed in that file is from 2006.
> Yes, it's very much out of date.
>
>   

Fix it! Beat some of those tech writers into submission!

N.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Steve Totaro]

On Tue, May 6, 2008 at 9:35 AM, Tilghman Lesher
<[EMAIL PROTECTED]> wrote:
> On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote:
>  > While these may not be popular opinions, I still ask, what does
>  > SwitchVox use? What do some of the guys around here that setup large
>  > systems use?  Is ABE even using 1.4 yet?
>
>  Yes, ABE version C (in release for several months) is using the 1.4 codebase.
>
>
>  > All I see in the ABE release
>  > notes is 1.2 although I have heard that ABE should be running 1.4
>  > "Very Soon" many many moons ago
>  > http://www.digium.com/en/docs/ABE/README .  So either Digium doesn't
>  > trust 1.4 enough to use it for ABE or the README is out of date.
>
>  The first clue should be that the copyright listed in that file is from 2006.
>  Yes, it's very much out of date.
>
>  --
>  Tilghman
>

Does "In Release" equate to "In the Wild" or "In Many Production
Installations" ?

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Jay R. Ashworth]

On Tue, May 06, 2008 at 01:38:37PM +0200, Benoit Plessis wrote:
> I'm wondering what version of asterisk people use in production 
> environnement ? on which distribution ?
> 
> And what is your setup like ?

Well, we're running a cluster of about 15 boxes or so with Slack 10 or
12 and 1.2.17(?, either 14 or 17) and VICIdial.  Yeah, you'd call it
production.  :-)

It runs, knock on Formica-laminated particle board, pretty well.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Basic modules of Asterisk

[Sanjay Rajdev]

I just want to Run Asterisk with the basic required modules, What can I do to 
achieve so? 

My only requirement is to run SIP clients and the Dictate Module. 

Regards, 
Sanjay Rajdev 
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IAX issues with 1.4.19.1

[Brian J. Murrell]

On Tue, 2008-05-06 at 08:42 -0500, Tilghman Lesher wrote:
> 
> It's not actually a fix to the security fix.

No, indeed.

> The security fix simply
> highlighted an issue which was already present in Asterisk.

That may be true, but the security fix now depends on that new fix, so
it's tangentially related at least.

> You would
> have seen this same issue prior to the security fix simply by running
> 'iax2 show channels', as that command runs through every single slot
> in the channel array, looking for active channels.  The security issue
> fix simply made that slow search process (already there, nothing new)
> painfully obvious.

Right.  Which to me at least, "tightly couples" it.  IOW, the security
fix, while yes, it fixes the security problem, is quite useless without
this other fix as it makes iax2 unstable.

b.



signature.asc
Description: This is a digitally signed message part
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Steve Totaro]

On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote:
> lordfuknowsyou a écrit :
>
> > Vinícius Fontes wrote:
>  >
>  > I use 1.4.18 with no problems. We have quite a few users(125 total
>  > between branches), but the call volume at the most has been around 15
>  > active calls at a time.
>  >
>  Any IAX2 phone or mostly SIP ?
>  Do you use Call Queues ?
>
>  We have less user than that, less concurrent call but quite a few
>  crash/deadlocks
>

Try SIP only if you can and report back.  I think you will confirm
what is pretty much a silent consensus (even among Digium Devs).

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] DUNDi call impossible in one direction

[Andrea Spadaccini]

Hello everybody,
I've set up DUNDi between two asterisk boxes, and sometimes happens that calls
from machine A can't reach peers in machine B, but calls from B to A work
correctly.

The strange thing is that the CLI command 'dundi show peers' shows correctly
the registered peer in both servers, and in this situation if I make a call
from B to A, suddenly peers in server A are able to call peers in machine B.

Can anyone give me directions?

Thanks in advance,

-- 
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dialplan, Extensions, etc. Worksheet

[Steve Totaro]

I still have not had time to dig and find what I have but there are
several "worksheets" ranging from sizing or initial customer
questionnaires.  This will give you an idea of what kind of hardware
you will need to purchase to put together a (hardware) quote.

Another "worksheet" goes over features, options, dialplan, telco, ISP,
network and some other  things.  This enables you to come up with a
"Scope of Work" and give you an idea of what to charge for labor to
add to your quote.  I highly recommend a detailed Scope of Work since
Asterisk can do "Anything" and that is often used as part of the sales
cycle.  You need to outline exactly what "features and functionality"
are included in the Scope of Work so you and your customer can tweak
it so they do not expect the world.  I find an hourly rate with a max
is the safest way to price labor.

The final "Worksheet" is a combination of "Best Practices", testing
and all of the items from your Scope of Work.  The scope of work and
the final worksheet checklist are to both be signed by you and your
customer.  I call this checklist the "Customer Acceptance" document
which basically says you have delivered and tested what was expected,
anything beyond that is "Extra".  Things like "Rubber Feet Attached"
can obviously be omitted unless you use rubber feet.

Anyways, that is the overview of what I have from several vendors.  I
am not sure where I have them archived but I will get the docs out
there somewhere for download soon.  Things like "Rubber Feet Attached"
can obviously be omitted unless you use rubber feet.

Thanks,
Steve Totaro

On Mon, May 5, 2008 at 7:43 PM, Roderick A. Anderson <[EMAIL PROTECTED]> wrote:
> Darren Wiebe wrote:
>  > If you're willing to cc me a copy I'll be in your debt.
>
>  You bet.
>
>
>  Rod
>  --
>
>
> >
>  > Thanks,
>  >
>  > Darren Wiebe
>  > [EMAIL PROTECTED]
>  >
>  > Steve Totaro wrote:
>  >> On Mon, May 5, 2008 at 5:10 PM, Roderick A. Anderson <[EMAIL PROTECTED]> 
> wrote:
>  >>
>  >>> Steve Totaro wrote:
>  >>>  > On Sun, May 4, 2008 at 1:55 PM, Roderick A. Anderson <[EMAIL 
> PROTECTED]> wrote:
>  >>>  >> Has anyone created a worksheet they can share for designing a 
> dialplan,
>  >>>  >>  extensions, voicemail, etc.
>  >>>  >>
>  >>>  >>  I'm making my way through the O'Reilly Book (dead tree version) and
>  >>>  >>  finding it enlightening.  I have hacked at dialplans created by 
> others
>  >>>  >>  but never actually came up with a design for my own system.  It's 
> sort
>  >>>  >>  of a work in progress made of bits and pieces from all over.
>  >>>  >>
>  >>>  >>  Having a real plan would probably make things easier.
>  >>>  >>
>  >>>  >>
>  >>>  >>  Rod
>  >>>  >>  --
>  >>>  >
>  >>>  > Rod,
>  >>>  >
>  >>>  > You will be glad that you are taking the learning curve plunge down
>  >>>  > the road.  No pain, no gain.
>  >>>  >
>  >>>  > I can certainly say that I am glad I got into Asterisk way before
>  >>>  > there was any real documentation or GUIs for that matter.  It forced
>  >>>  > me to learn the real deal Asterisk through trial and error which is
>  >>>  > invaluable if you plan on really getting into it.
>  >>>  >
>  >>>  > Then again, if you want easy, use a GUI.
>  >>>
>  >>>  Easy isn't what I'm after.  I was hoping for planning worksheets.
>  >>>  Something to go over with a customer (I know I said this was for my
>  >>>  personal system but that is the first step).  How many extensions/
>  >>>  phones/ softphones, and what their /numeric/ extension will be.  An IVR
>  >>>  plan and the text that goes with it, voice-mail handling and mailboxes, 
> etc.
>  >>>
>  >>>  This type of stuff.
>  >>>
>  >>>  So from the minimal number of responses -- yours :-) -- I'm going to
>  >>>  guesstimate no one has anything like this at all or that they can or are
>  >>>  able/willing to share.
>  >>>
>  >>>  Out comes the notepad and the thinking cap.  /-|
>  >>>
>  >>>
>  >>>  Cheers,
>  >>>  Rod
>  >>>  --
>  >>>  >
>  >>>  > Thanks,
>  >>>  > Steve Totaro
>  >>>
>  >>>
>  >>>
>  >> Hey Rod,
>  >>
>  >> I think I may be able to help with worksheets from 3com, NEC, and
>  >> other system vendor's sales channel.  It obviously will not match
>  >> exactly to Asterisk but will give you a great foundation for the
>  >> functions and features that you need to question.
>  >>
>  >> I have my own but I prefer not to put it in the public domain.  It is
>  >> adapted from a conglomeration of many different proprietary systems
>  >> that I have dealt with.  I think many others have the same and
>  >> consider it proprietary internal information for their business.
>  >>
>  >> Let me see what I can dig up from my archives.
>  >>
>  >> Thanks,
>  >> Steve Totaro
>  >>
>  >> ___
>  >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>  >>
>  >> asterisk-users mailing list
>  >> To UNSUBSCRIBE or update options visit:
>  >>http://lists.digium.com/mailman/listinfo/aste

Re: [asterisk-users] Asterisk in Production ?

[Vinícius Fontes]

Mostly SIP, some of my clients have queues and everything is working fine by 
now.



Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.

- "Benoit Plessis" <[EMAIL PROTECTED]> escreveu:

> lordfuknowsyou a écrit :
> > Vinícius Fontes wrote:
> >
> > I use 1.4.18 with no problems. We have quite a few users(125 total
> > between branches), but the call volume at the most has been around
> 15
> > active calls at a time.
> >
> Any IAX2 phone or mostly SIP ?
> Do you use Call Queues ?
> 
> We have less user than that, less concurrent call but quite a few
> crash/deadlocks
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IAX issues with 1.4.19.1

[Tilghman Lesher]

On Tuesday 06 May 2008 07:35:14 Brian J. Murrell wrote:
> On Tue, 2008-05-06 at 13:23 +0100, Julian Lyndon-Smith wrote:
> > Yes.
>
> Hrm.  For those of us that are following along the AST-* train, patching
> as per the AST-* release notices, as a matter of process, wouldn't it
> have been good to republish AST-2008-006 and include this fix along with
> the original patch?
>
> IOW, IMHO, it should be standard practice that when you release a fix to
> a patch in an AST-* security release, that you re-publish the security
> notice complete with the original security fix and the security fix fix.

It's not actually a fix to the security fix.  The security fix simply
highlighted an issue which was already present in Asterisk.  You would
have seen this same issue prior to the security fix simply by running
'iax2 show channels', as that command runs through every single slot
in the channel array, looking for active channels.  The security issue
fix simply made that slow search process (already there, nothing new)
painfully obvious.

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Tilghman Lesher]

On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote:
> While these may not be popular opinions, I still ask, what does
> SwitchVox use? What do some of the guys around here that setup large 
> systems use?  Is ABE even using 1.4 yet?

Yes, ABE version C (in release for several months) is using the 1.4 codebase.

> All I see in the ABE release 
> notes is 1.2 although I have heard that ABE should be running 1.4
> "Very Soon" many many moons ago
> http://www.digium.com/en/docs/ABE/README .  So either Digium doesn't
> trust 1.4 enough to use it for ABE or the README is out of date.

The first clue should be that the copyright listed in that file is from 2006.
Yes, it's very much out of date.

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] using cell phone as an FXO port

[Steve Totaro]

On Mon, May 5, 2008 at 9:29 PM, gmail <[EMAIL PROTECTED]> wrote:
>
>
> Hi all,
> I want to use a cell phone as my FXO line to Asterisk Box ,did anyone
> try this and configured it and how to physically connect it to Asterisk
> server?

Check out chan_mobile.  Super cool.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] using cell phone as an FXO port

[gmail]

Hi all,
I want to use a cell phone as my FXO line to Asterisk Box ,did anyone try 
this and configured it and how to physically connect it to Asterisk server?___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Patrick]

On Tue, 2008-05-06 at 07:58 -0400, Steve Totaro wrote:
[snip]
> While these may not be popular opinions, I still ask, what does
> SwitchVox use?  

Not sure what Asterisk version they use but I saw (iirc) a presentation
on their website that they run switchvox on top of Fedora Core 6. FC6
has been end-of-line for a long, long time... 

Regards,
Patrick


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Tilghman Lesher]

On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
> lordfuknowsyou a écrit :
> > Vinícius Fontes wrote:
> >
> > I use 1.4.18 with no problems. We have quite a few users(125 total
> > between branches), but the call volume at the most has been around 15
> > active calls at a time.
>
> Any IAX2 phone or mostly SIP ?
> Do you use Call Queues ?
>
> We have less user than that, less concurrent call but quite a few
> crash/deadlocks

Have you reported these issues on the bugtracker?

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Tilghman Lesher]

On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
> 5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
> > On Monday 05 May 2008 11:24, Johansson Olle E wrote:
> >> 5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
> >>> On Monday 05 May 2008 09:45, Johansson Olle E wrote:
>  Another issue that we need to fix with the MYSQL driver is that
>  we're
>  lacking a connection pool. Everything seems to be handled over one
>  connection to Mysql, which causes issues.
> >>>
> >>> That's not true.  The MYSQL app generally uses multiple connections,
> >>> one
> >>> for each channel.  The only way one might use only a single
> >>> connection is
> >>> by using a global variable to store a single connection id, but that
> >>> method
> >>> is not documented anywhere, AFAIK.
> >>
> >> You talk about the Mysql APP, but is this the case with the Realtime
> >> driver as well?
> >
> > No, the native Realtime driver uses a single connection.  The ODBC
> > Realtime
> > driver generally uses a single connection but can be configured to
> > use a
> > separate connection for each query.
>
> So, we're back to where we started. A developer that can help us with
> a connection
> pool or a separate connection for each query would be a Nice Thing (TM).

What issues are you specifically seeing that merit using multiple
connections?

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Steve Totaro]

>  I use 1.4.18 with no problems. We have quite a few users(125 total
>  between branches), but the call volume at the most has been around 15
>  active calls at a time.
>
>

I would classify that as "Light to Medium Call Volume" or "SMB".

Let me clarify what I consider "High Call Volume".  ~400 simultaneous
calls, all SIP or 95 on a box doing quad PRI to SIP gateway duty.
15k+ calls a day lasting an average of fifteen minutes.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Benoit Plessis]

lordfuknowsyou a écrit :
> Vinícius Fontes wrote:
>   
> I use 1.4.18 with no problems. We have quite a few users(125 total 
> between branches), but the call volume at the most has been around 15 
> active calls at a time.
>   
Any IAX2 phone or mostly SIP ?
Do you use Call Queues ?

We have less user than that, less concurrent call but quite a few 
crash/deadlocks


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[lordfuknowsyou]

Vinícius Fontes wrote:
> There were some really unstable Asterisk releases in the 1.4 branch. I 
> personally use 1.4.13 or 1.4.15 in production. Every single time I tried 
> 1.4.16 or higher I had problems.
>
>
>
> Att
> Vinícius Fontes
> Desenvolvimento
> Canall Tecnologia em Comunicações Ltda.
>
> - "Steve Totaro" <[EMAIL PROTECTED]> escreveu:
>
>   
>> On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis <[EMAIL PROTECTED]>
>> wrote:
>> 
>>>  Hi,
>>>
>>>  I'm wondering what version of asterisk people use in production
>>>  environnement ?
>>>  on which distribution ?
>>>
>>>  And what is your setup like ?
>>>
>>>  We are actually running an AsteriskNow appliance with asterisk
>>>   
>> 1.4.18.1
>> 
>>>  and it's quite unstable.
>>>  We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX
>>>   
>> destroy
>> 
>>>  deadlock"
>>>  and now that we have added a Queue, it's worse than ever. The queue
>>>   
>> goes
>> 
>>>  stuck quite often
>>>  (agent are stuck in 'In use' state and if they logoff they can't
>>>   
>> log-in
>> 
>>>  till an asterisk restart).
>>>
>>>
>>>  regards
>>>
>>>   
>> I am personally a proponent of Asterisk 1.2.X as I see more and more
>> fatal bugs in the 1.4.X code come up on the lists as well as IAX2
>> bugs.  I constantly hear "Asterisk 1.4.whatever is much better, but
>> the bugs coming out are not just unexpected behavior that one could
>> live with, they are segfaults, system crashes, modules not getting
>> installed (Zaptel).
>>
>> I use SIP since I have seen quite a few issues with IAX2 that were
>> solved by simply switching to SIP.
>>
>> The above two yield solid systems under heavy load for me.  OS is not
>> so important I do not believe.  I have some running FC8 and more
>> running CentOS, both rock solid.  I think the general consensus on OS
>> is use what you are most familiar with.
>>
>> While these may not be popular opinions, I still ask, what does
>> SwitchVox use?  What do some of the guys around here that setup large
>> systems use?  Is ABE even using 1.4 yet?  All I see in the ABE
>> release
>> notes is 1.2 although I have heard that ABE should be running 1.4
>> "Very Soon" many many moons ago
>> http://www.digium.com/en/docs/ABE/README .  So either Digium doesn't
>> trust 1.4 enough to use it for ABE or the README is out of date.
>>
>> Thanks,
>> Steve Totaro
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
I use 1.4.18 with no problems. We have quite a few users(125 total 
between branches), but the call volume at the most has been around 15 
active calls at a time.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IAX issues with 1.4.19.1

[Brian J. Murrell]

On Tue, 2008-05-06 at 13:23 +0100, Julian Lyndon-Smith wrote:
> Yes.

Hrm.  For those of us that are following along the AST-* train, patching
as per the AST-* release notices, as a matter of process, wouldn't it
have been good to republish AST-2008-006 and include this fix along with
the original patch?

IOW, IMHO, it should be standard practice that when you release a fix to
a patch in an AST-* security release, that you re-publish the security
notice complete with the original security fix and the security fix fix.

Some of us don't have the bandwidth to keep upgrading to
release-of-the-week or to watch every commit looking for both the
security fixes and the fixes to the security fixes.

Your consideration of this process enhancement would be greatly
appreciated.

Thanx,
b.

/me goes off to patch his * once again.



signature.asc
Description: This is a digitally signed message part
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IAX issues with 1.4.19.1

[Julian Lyndon-Smith]

Yes.

Julian

Brian J. Murrell wrote:
> On Mon, 2008-05-05 at 16:36 -1000, Julian Yap wrote:
>> That was a bug in the release.
>>
>> From the 1.4.20-rc1 Changelog:
>> 2008-04-30 16:30 + [r114891]  Russell Bryant <[EMAIL PROTECTED]>
> 
> So basically, r114891 was a fix to AST-2008-006?  So if you applied the
> patch for AST-2008-006 you now really need this new fix (r114891) to
> regain the stability that chan_iax2.c had before the AST-2008-006 patch?
> 
> b.
> 
> 
> 
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IAX issues with 1.4.19.1

[Brian J. Murrell]

On Mon, 2008-05-05 at 16:36 -1000, Julian Yap wrote:
> That was a bug in the release.
> 
> From the 1.4.20-rc1 Changelog:
> 2008-04-30 16:30 + [r114891]  Russell Bryant <[EMAIL PROTECTED]>

So basically, r114891 was a fix to AST-2008-006?  So if you applied the
patch for AST-2008-006 you now really need this new fix (r114891) to
regain the stability that chan_iax2.c had before the AST-2008-006 patch?

b.



signature.asc
Description: This is a digitally signed message part
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Running Asterisk as root

[Christian]

On 2008-05-06 at 03:46 Tzafrir Cohen wrote:

>On Mon, May 05, 2008 at 07:18:08PM -0500, Cesar Benjamin Garcia Martinez
>wrote:
>> Move to root:
>> 
>> sudo -s
>> 
>> type your passwd
>> 
>> and as root:
>> 
>> 
>> Edit the file /etc/init.d/asterisk
>> 
>> And uncommet the two lines than sasys something like 
>> 
>> AST_USER="asterisk"
>> AST_GROUP="asterisk"
>> 
>> You need to create the user asterisk on your system.
>> 
>> And create another symlink sh to bash:
>> 
>> cd /bin
>> rm -f sh
>> ln -s bash sh 
>
>Why is that?
>
>Debian / Ubuntu policy is that a script that is not posix sh should use
>/bin/bash. Any script of Asterisk does not fit the policy and has not
>bit shot^Wfixed yet?
>
>The fix is to edit the ofending script:
>
>#!/bin/sh  ->  #!/bin/bash
>
>> 
>> 
>> 
>> Edit your /etc/asterisk/asterisk.conf and replace the line:
>> 
>> astrundir => /var/run 
>> 
>> With:
>> 
>> astrundir => /var/lib/asterisk/var/run
>
>/var/run/asterisk
>
>Everything under /var/run is deleted at boot with Ubuntu, so the init.d
>script should recreate that directory and give it proper permissions if
>it does not exist.
>
>(or use the one from the Asterisk package)
>
>> 
>> Create that folder:
>> 
>> mkdir -p /var/lib/asterisk/var/run
>
>/var/run/asterisk, as mentioned above. and it should be created in the
>init.d script .
>
>> 
>> and, chown to asterisk:asterisk the folders:
>> 
>> 
>> /var/lib/asterisk/
>> /usr/lib/asterisk/
>
>No real need for /usr/lib/asterisk to be owned by Asterisk. It is
>read-only. /usr is read-only, as you recall.
>
>> /var/log/asterisk/
>> 
>> chown -Rv asterisk:asterisk /var/lib/asterisk/
># chown -Rv asterisk:asterisk /usr/lib/asterisk/
>> chown -Rv asterisk:asterisk /var/log/asterisk/
>> 
>> that's all
>> 
>> 
>> 
>> Btw... delete the symlink sh -> dash into /bin
>
>NOT
>
>> 
>> Start daemon
>> 
>> 
>> /etc/init.d/asterisk start
>
>-- 
>   Tzafrir Cohen
>icq#16849755  jabber:[EMAIL PROTECTED]
>+972-50-7952406   mailto:[EMAIL PROTECTED]
>http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
So what instructions are correct?
I don't want to do anything that might not work.
Many thanks,
Christian
>___
>-- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users




___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Vinícius Fontes]

There were some really unstable Asterisk releases in the 1.4 branch. I 
personally use 1.4.13 or 1.4.15 in production. Every single time I tried 1.4.16 
or higher I had problems.



Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.

- "Steve Totaro" <[EMAIL PROTECTED]> escreveu:

> On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis <[EMAIL PROTECTED]>
> wrote:
> >
> >  Hi,
> >
> >  I'm wondering what version of asterisk people use in production
> >  environnement ?
> >  on which distribution ?
> >
> >  And what is your setup like ?
> >
> >  We are actually running an AsteriskNow appliance with asterisk
> 1.4.18.1
> >  and it's quite unstable.
> >  We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX
> destroy
> >  deadlock"
> >  and now that we have added a Queue, it's worse than ever. The queue
> goes
> >  stuck quite often
> >  (agent are stuck in 'In use' state and if they logoff they can't
> log-in
> >  till an asterisk restart).
> >
> >
> >  regards
> >
> 
> I am personally a proponent of Asterisk 1.2.X as I see more and more
> fatal bugs in the 1.4.X code come up on the lists as well as IAX2
> bugs.  I constantly hear "Asterisk 1.4.whatever is much better, but
> the bugs coming out are not just unexpected behavior that one could
> live with, they are segfaults, system crashes, modules not getting
> installed (Zaptel).
> 
> I use SIP since I have seen quite a few issues with IAX2 that were
> solved by simply switching to SIP.
> 
> The above two yield solid systems under heavy load for me.  OS is not
> so important I do not believe.  I have some running FC8 and more
> running CentOS, both rock solid.  I think the general consensus on OS
> is use what you are most familiar with.
> 
> While these may not be popular opinions, I still ask, what does
> SwitchVox use?  What do some of the guys around here that setup large
> systems use?  Is ABE even using 1.4 yet?  All I see in the ABE
> release
> notes is 1.2 although I have heard that ABE should be running 1.4
> "Very Soon" many many moons ago
> http://www.digium.com/en/docs/ABE/README .  So either Digium doesn't
> trust 1.4 enough to use it for ABE or the README is out of date.
> 
> Thanks,
> Steve Totaro
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk in Production ?

[Steve Totaro]

On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote:
>
>  Hi,
>
>  I'm wondering what version of asterisk people use in production
>  environnement ?
>  on which distribution ?
>
>  And what is your setup like ?
>
>  We are actually running an AsteriskNow appliance with asterisk 1.4.18.1
>  and it's quite unstable.
>  We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX destroy
>  deadlock"
>  and now that we have added a Queue, it's worse than ever. The queue goes
>  stuck quite often
>  (agent are stuck in 'In use' state and if they logoff they can't log-in
>  till an asterisk restart).
>
>
>  regards
>

I am personally a proponent of Asterisk 1.2.X as I see more and more
fatal bugs in the 1.4.X code come up on the lists as well as IAX2
bugs.  I constantly hear "Asterisk 1.4.whatever is much better, but
the bugs coming out are not just unexpected behavior that one could
live with, they are segfaults, system crashes, modules not getting
installed (Zaptel).

I use SIP since I have seen quite a few issues with IAX2 that were
solved by simply switching to SIP.

The above two yield solid systems under heavy load for me.  OS is not
so important I do not believe.  I have some running FC8 and more
running CentOS, both rock solid.  I think the general consensus on OS
is use what you are most familiar with.

While these may not be popular opinions, I still ask, what does
SwitchVox use?  What do some of the guys around here that setup large
systems use?  Is ABE even using 1.4 yet?  All I see in the ABE release
notes is 1.2 although I have heard that ABE should be running 1.4
"Very Soon" many many moons ago
http://www.digium.com/en/docs/ABE/README .  So either Digium doesn't
trust 1.4 enough to use it for ABE or the README is out of date.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Predictive dialer - which one would you recommend?

[Matt Florell]

On 5/6/08, Asterisk <[EMAIL PROTECTED]> wrote:
> Hi guys,
>
>  I would like to ask you, if any of you has any experiences with the 
> predictive dialers available for Asterisk? Are open source predictive dialers 
> such as VICIDIAL Dialer any good?
>
>  Which one would you recommend for a ca. 45 seat call center where most of 
> the agents work on both inbound/outbound and are already using their own CTI 
> software (so the predictive dialer software will be an appendix to the 
> existing system and should be integrated with 3rd party software reasonably 
> easy).
>
>  Thanks,
>  Alex

I think that VICIDIAL is pretty good(full disclosure, I wrote it). It
is currently in use at over 700 companies in over 70 countries around
the world and is available in 9 languages. 45 seats blended are no
problem for VICIDIAL, we even have some installtions running that are
over 300 seats.

As for integration with 3rd party software, we have integrated with
many different kinds of web-based and client/server applications for
our clients. Could you explain a little more exactly what functions
that you would want the call center software to perform, and what
functions you want your existing application to perform?

Thanks,

MATT---

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk in Production ?

[Benoit Plessis]

Hi,

I'm wondering what version of asterisk people use in production 
environnement ?
on which distribution ?

And what is your setup like ?

We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 
and it's quite unstable.
We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX destroy 
deadlock"
and now that we have added a Queue, it's worse than ever. The queue goes 
stuck quite often
(agent are stuck in 'In use' state and if they logoff they can't log-in 
till an asterisk restart).


regards

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Predictive dialer - which one would you recommend?

[Asterisk]

Hi guys,

I would like to ask you, if any of you has any experiences with the predictive 
dialers available for Asterisk? Are open source predictive dialers such as 
VICIDIAL Dialer any good?

Which one would you recommend for a ca. 45 seat call center where most of the 
agents work on both inbound/outbound and are already using their own CTI 
software (so the predictive dialer software will be an appendix to the existing 
system and should be integrated with 3rd party software reasonably easy).

Thanks,
Alex


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Unicall - How to automatically block collect calls

[Oscar Patricio]

Hi!

Thank you for your answer.

However I would like to know, if there is any other possibility of 
making this work, with older unicall versions.

The reason I ask, is because it's not easy for me to upgrade the 
asterisk and unicall version on my machine, since it is a production 
machine, and I would consider a software upgrade really only if no other 
solution is possible.

Thanks in advance for your answer.

Best regards,

Óscar Patrício


Moises Silva escreveu:
> The latest version of the driver included in
> http://www.moythreads.com/astunicall/ comes with a change that will
> set the variable UC_CATEGORY in your dialplan, Brasil has a special
> category for those calls, don't remember the name that will show up,
> but you can make a couple of tests and then drop any call with that
> specific category.
>
> Moy
>
> On Mon, May 5, 2008 at 9:14 AM, Oscar Patricio <[EMAIL PROTECTED]> wrote:
>   
>> Hi!
>>
>>  I am using asterisk with unicall in brasil.
>>
>>  Everything was working fine, but now we want to set up a way to
>>  automatically drop collect calls, because we have an IVR answering all
>>  calls automatically!
>>
>>  Can you tell me, what I have to configure to block collect calls in the
>>  asterisk?
>>
>>  Thank you!
>>  Best regards,
>>
>>  Óscar Patrício
>>
>>  ___
>>  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>>  asterisk-users mailing list
>>  To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> 
>
>
>
>   


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Running Asterisk as root

[Alan Lord]

Christian wrote:
> Hi all,
> I have seen discussions on this earlier on, but just want to hear some quick 
> thoughts.
> I am running v1.6 of Asterisk on my Ubuntu installation, I did make config to 
> make it run at boot. Since I've got a firewall and don't have any other 
> servers running I am not worried. I have been htinking about running Asterisk 
> as a seperat user, but haven't done that yet.
> Everything is working fine.
> What do you think?
> Thanks,
> Christian
> 

I'd never run a server app as root. It is just asking for trouble IMHO.

When I built asterisk on my little custom linux server I documented the 
process of setting up as a non-privileged process here. Most of the 
information originally came from the voip-info.org site:

http://www.theopensourcerer.com/2007/10/30/untangle-asterisk-pbx-and-file-server-all-in-one-part-7/

Hope this helps.

Al

-- 
The way out is open!
http://www.theopensourcerer.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Running Asterisk as root

[Andreas van dem Helge]

I totally agree. Someone filed a bugreport for this? Also asterisk
init script should be installed by default too.

I am going to give Cesar's instructions a try (sans removing /bin/sh)
and hope it works!

On Tue, May 6, 2008 at 3:24 AM, Stelios Koroneos
<[EMAIL PROTECTED]> wrote:
> In general, if your asterisk is accesible from the internet its much better
>  to have it run as a non-root process.
>  (My opinion is that this should be the default out-of-the-makefile ;)
>  asterisk behaviour)
>  This is the "norm" for more of the servers/services running on a linux
>  system, and can act as a safety-net when things go bad
>
>
>  Stelios S. Koroneos
>
>  Digital OPSiS - Embedded Intelligence
>  http://www.digital-opsis.com
>
>
>
>
>  > -Original Message-
>  > From: [EMAIL PROTECTED]
>  > [mailto:[EMAIL PROTECTED] On Behalf Of
>  > Christian
>  > Sent: Tuesday, May 06, 2008 3:00 AM
>  > To: asterisk-users@lists.digium.com
>  > Subject: [asterisk-users] Running Asterisk as root
>  >
>  > Hi all,
>  > I have seen discussions on this earlier on, but just want to
>  > hear some quick thoughts.
>  > I am running v1.6 of Asterisk on my Ubuntu installation, I
>  > did make config to make it run at boot. Since I've got a
>  > firewall and don't have any other servers running I am not
>  > worried. I have been htinking about running Asterisk as a
>  > seperat user, but haven't done that yet.
>  > Everything is working fine.
>  > What do you think?
>  > Thanks,
>  > Christian
>  >
>  >
>  > ___
>  > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>  >
>  > asterisk-users mailing list
>  > To UNSUBSCRIBE or update options visit:
>  >http://lists.digium.com/mailman/listinfo/asterisk-users
>  >
>
>
>
>  ___
>  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>  asterisk-users mailing list
>  To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Running Asterisk as root

[Stelios Koroneos]

In general, if your asterisk is accesible from the internet its much better
to have it run as a non-root process.
(My opinion is that this should be the default out-of-the-makefile ;)
asterisk behaviour)
This is the "norm" for more of the servers/services running on a linux
system, and can act as a safety-net when things go bad


Stelios S. Koroneos

Digital OPSiS - Embedded Intelligence
http://www.digital-opsis.com
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Christian
> Sent: Tuesday, May 06, 2008 3:00 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Running Asterisk as root
> 
> Hi all,
> I have seen discussions on this earlier on, but just want to 
> hear some quick thoughts.
> I am running v1.6 of Asterisk on my Ubuntu installation, I 
> did make config to make it run at boot. Since I've got a 
> firewall and don't have any other servers running I am not 
> worried. I have been htinking about running Asterisk as a 
> seperat user, but haven't done that yet.
> Everything is working fine.
> What do you think?
> Thanks,
> Christian
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Johansson Olle E]

5 maj 2008 kl. 19.58 skrev Tilghman Lesher:

> On Monday 05 May 2008 11:24, Johansson Olle E wrote:
>> 5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
>>> On Monday 05 May 2008 09:45, Johansson Olle E wrote:
 Another issue that we need to fix with the MYSQL driver is that  
 we're
 lacking a connection pool. Everything seems to be handled over one
 connection to Mysql, which causes issues.
>>>
>>> That's not true.  The MYSQL app generally uses multiple connections,
>>> one
>>> for each channel.  The only way one might use only a single
>>> connection is
>>> by using a global variable to store a single connection id, but that
>>> method
>>> is not documented anywhere, AFAIK.
>>
>> You talk about the Mysql APP, but is this the case with the Realtime
>> driver as well?
>
> No, the native Realtime driver uses a single connection.  The ODBC  
> Realtime
> driver generally uses a single connection but can be configured to  
> use a
> separate connection for each query.

So, we're back to where we started. A developer that can help us with  
a connection
pool or a separate connection for each query would be a Nice Thing (TM).

/O

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users