[asterisk-users] Upgrade 1.4.19 to 1.6 => segementation fault

[Ronald Wiplinger (Lists)]

During compiling I have not seen an error, however, when I start
asterisk again it ends with:


app_morsecode.so => (Morse code)
  == Registered custom function 'SYSINFO'
 func_sysinfo.so => (System information related functions)
Segmentation fault (core dumped)


How can I figure out what is wrong?

bye

Ronald

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[asterisk-users] Asterisk Instant message passing with eyebeam

[Max Alex]

Hi All,
I am searching about asterisk IM message passing with eyebeam.
but i am not able to send instant message to another registered users.
i am working in asterisk 1.4 branch.
i have tested within call and without call but there is no message recieved.
and every time i got error user not found in eye beam.
and in asterisk i got Method is not implemented.

Can anybody helps me in this?
If any patches are there then please let me know.
Thanks in advance!!
Thanks,
Max Alex
Voip Developer
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[asterisk-users] Need Recording Solution in Asterisk

[Kashif Naeem]

Hello All

One of our client Bank has 900 employees working in different locations.
They need to record all internal and external calls. Can any body suggest
Call Recording Solution for this requirement. We need to know the Hardware /
Bandwidth and  all requirements and costing.

Regards,
-- 
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com

Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766

Email: [EMAIL PROTECTED]
MSN: [EMAIL PROTECTED]
Gmail: [EMAIL PROTECTED]
Skype: kashif.naeem

302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.
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[asterisk-users] CDR Desgin

[Grey Man]

I've taken the liberty of starting a new thread to discuss the design
of the Asterisk CDR mechanism. The discussion has been kindly
initiated by murf putting together a proposal:
http://svn.digium.com/svn/asterisk/team/murf/RFCs.

After reading the proposal I still don't think it's the right way to
go. To my mind adding more channel variables increases the complexity
in a situation that is already overly so. I think it's a mistake to
try and think about all the different call scenarios and come up with
little tricks for the more complicated ones. There will always be
something missed; app_shotgun initiates calls to 100 random numbers
and as soon as three or more calls are answered it will start randonly
transferring them amongst each other at 2 second intervals.

I think it's important to clarify at the outset what a CDR should be.
The most fundamental requirement for CDRs is that they accurately
record the following pieces of information for EVERY call entering or
leaving the system (note every means every and not; "channel" calls
but not "peer" calls).

1. Destination (aka as A Number)
2. AccountCode (aka as B Number)
3. Call Start Time (answer time),
4. Duration.

Of course adding extra information can be very useful and I'm not
proposing any fields be removed from the current implementation
(although for pity's sake one change that should be made it to use a
GUID/UUID for the CDR's uniqueid and save endless confusion).

People that really do need verbose or enhanced CDRs to do things like
tracking a call's flow as it travels in and out of queues, parking
lots etc. would be better off using AMI or the new CEL and not CDRs.
At the very least if problems arise with their call flow tracking they
will still be able to rely on the accuracy of the CDRs to piece it
altogether to work out what's going wrong.

My proposal of creating a 1-to-1 relationship between CDRs and
Asterisk channels already exsits but somewhere along the line it's
going awry. As an experiment, and to actually do something instead of
continually moaning about it, I started commenting out the blocks of
code in res_featrures.c and sip_channel.c that muck around with the
channel CDRs when a transfer occurs. The results of that were that the
CDRs for blind and attended transfers actually got better! They're
still not quite right but are pretty close with only one CDR on each
having a wrong detstination.

Regards,

Greyman.

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Michael Collins]

> Date: Fri, 21 Nov 2008 16:20:28 -0600
> From: Terry Wilson <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Large Asterisk installarions (~10,
000
>   extensions), preferably at universities
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>   
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
> 
> > Yehavi Bourvine wrote:
> >
> >> OK, but I still did not get a reply to my original question: Why
> >> using
> >> SIP registrar in front of Asterisk and not simply use bare
Astersik?
> >> can't it handle the load? (remember - in my case it doesn't handle
> >> the
> >> RTP, only signalling). Can't it handle so much registrations? (I am
> >> using realtime DB, it is has any relevance).
> >
> > My experience has shown that using a dedicated registrar for large
> > installs is more effective;  it doesn't tie up resources on the
> > Asterisk
> > box with all those registration refreshes, for one.  A product built
> > to
> > be a high-throughput standalone registrar will handle the
concurrency
> > requirements and perform better.
> 
> I've looked at doing various things to chan_sip to improve signaling
> performance (hash tables for call lookups, etc.)  I gave up when I
> realized that the overhead of handling the RTP was so far above the
> overhead of processing SIP signaling that it didn't really matter
> much.  The only reason I have ever had to use a SIP registrar (OpenSER
> in my case) was if I needed to load balance calls across multiple
> asterisk servers.  If most of the phones are not separated by a NAT
> from Asterisk (as would be the case in something like a University
> network), the registration timeout could be set to a relatively high
> value w/o causing any problems which would cut down on some of the SIP
> traffic from registrations.
> 
> In fact, I just ran some tests using SIPp and w/o any audio, using
> realtime w/ 10k accounts I can register 100/second while doing 10
> calls/second.  If you are looking just at registrations every 15
> minutes or so, that is 90k devices that could register to asterisk.
> This was using 1.6.0.1 on my little HP amd64 development box--not
> anything near the kind of machine that you would probably install in a
> large installation.  Asterisk just gets faster and faster.  Some of
> the "it isn't good at x" stuff comes from experiences with older
> releases.

In a HA and/or high volume scenario I worry about stuff like this that
has been in tree since 1.0 or earlier and is in 1.6, channel.c lines
3825~3828:

/* XXX This is a seriously wacked out operation.  We're
essentially putting the guts of
   the clone channel into the original channel.  Start by
killing off the original
   channel's backend.   I'm not sure we're going to keep this
function, because
   while the features are nice, the cost is very high in terms
of pure nastiness. XXX */

That's not something I want in my high-end, high-capacity,
high-availability production system!

For smallish installations, this probably isn't a big deal given today's
hardware capabilities. Still, it makes me wonder what other gremlins are
out there that might bite me in a big-time install. 

At least with OSS I can see stuff like this. I shudder to think what
psycho spaghetti code is running on Cisco, Avaya, Nortel, NEC, Shoretel,
etc.

-MC

> 
> If you are lucky enough to have a situation where you can re-invite
> media and keep it off of the asterisk box, it can handle huge loads.
> 
> Terry

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Re: [asterisk-users] MOH Realtime Problem

[Sebastian]

My second problem is resolved, qualify=yes did the trick.

I'm still having problems with MOH

 

 

De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sebastian
Enviado el: Friday, November 21, 2008 9:09 PM
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] MOH Realtime Problem

 

Hi,

 

I'm having 2 problems:

 

1)  MOH in realtime is not working, I have configured it but never go to
look at the database, no warning or error found and I can do a query using
realtime and the family from the cli.

2)  I have SIP phones via realtime, if I register one of them and a call
to a queue comes the call is never delivered to the phone, I have to make a
call from the phone so the phone start getting calls from de queue.

 

 

Any ideas??

 

 

Ast 1.6.0.1

 

 

 

 

 

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[asterisk-users] MoH in a loop

[Robert Augustyn]

Hi all,
Is it possible to have * playing an mp3 file in the way old tape system
worked?
 
 
Sincerely,
Robert Augustyn
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[asterisk-users] MOH Realtime Problem

[Sebastian]

Hi,

 

I'm having 2 problems:

 

1)  MOH in realtime is not working, I have configured it but never go to
look at the database, no warning or error found and I can do a query using
realtime and the family from the cli.

2)  I have SIP phones via realtime, if I register one of them and a call
to a queue comes the call is never delivered to the phone, I have to make a
call from the phone so the phone start getting calls from de queue.

 

 

Any ideas??

 

 

Ast 1.6.0.1

 

 

 

 

 

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Terry Wilson]

> I've looked at doing various things to chan_sip to improve signaling
> performance (hash tables for call lookups, etc.)  I gave up when I
> realized that the overhead of handling the RTP was so far above the
> overhead of processing SIP signaling that it didn't really matter
> much.  The only reason I have ever had to use a SIP registrar (OpenSER

Keep in mind this was in 1.4.  They actually use hash lookups now  
anyway.

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Terry Wilson]

> Yehavi Bourvine wrote:
>
>> OK, but I still did not get a reply to my original question: Why  
>> using
>> SIP registrar in front of Asterisk and not simply use bare Astersik?
>> can't it handle the load? (remember - in my case it doesn't handle  
>> the
>> RTP, only signalling). Can't it handle so much registrations? (I am
>> using realtime DB, it is has any relevance).
>
> My experience has shown that using a dedicated registrar for large
> installs is more effective;  it doesn't tie up resources on the  
> Asterisk
> box with all those registration refreshes, for one.  A product built  
> to
> be a high-throughput standalone registrar will handle the concurrency
> requirements and perform better.

I've looked at doing various things to chan_sip to improve signaling  
performance (hash tables for call lookups, etc.)  I gave up when I  
realized that the overhead of handling the RTP was so far above the  
overhead of processing SIP signaling that it didn't really matter  
much.  The only reason I have ever had to use a SIP registrar (OpenSER  
in my case) was if I needed to load balance calls across multiple  
asterisk servers.  If most of the phones are not separated by a NAT  
from Asterisk (as would be the case in something like a University  
network), the registration timeout could be set to a relatively high  
value w/o causing any problems which would cut down on some of the SIP  
traffic from registrations.

In fact, I just ran some tests using SIPp and w/o any audio, using  
realtime w/ 10k accounts I can register 100/second while doing 10  
calls/second.  If you are looking just at registrations every 15  
minutes or so, that is 90k devices that could register to asterisk.   
This was using 1.6.0.1 on my little HP amd64 development box--not  
anything near the kind of machine that you would probably install in a  
large installation.  Asterisk just gets faster and faster.  Some of  
the "it isn't good at x" stuff comes from experiences with older  
releases.

If you are lucky enough to have a situation where you can re-invite  
media and keep it off of the asterisk box, it can handle huge loads.

Terry

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[asterisk-users] MozIAX - Mozilla IAX2 soft-phone 3sec delay

[Joseph]

Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone.
http://moziax.mozdev.org/

I tried it yesterday on eee pc, connected to asterisk on local LAN and the 
performance is terrible!
The delay is about 2sec or 3sec. and very bad echo. 
I think it is the implementation of their IAX2 in their add on, as I have tried 
external mic. and the same delay problem.

As a comparison I've tried DIAX over dial-up connection and the voice quality 
was acceptable with very little delay.

-- 
#Joseph
GPG KeyID: ED0E1FB7

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[asterisk-users] Setting up to reveive faxes.

[Ken D'Ambrosio]

Hey, all.  When I last was heavily into Asterisk (1.0.x), setting up to
receive faxes was, well, a PITA, what with having to patch the Asterisk
install with various driver patches and this, that, and the other.

Is that still true?  Is there a fax HOWTO out there that reflects Asterisk
1.4.x?

Thanks!

-Ken




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Re: [asterisk-users] hint priority with 50 channels

[Anthony Francis]

Just curious but why would you want to have a lot of devices all have 
the exact same state information?

Philipp Kempgen wrote:
> Loic Didelot schrieb:
>
>   
>> I noticed that my hint priority stops working when I add to many
>> extensions/channels. It looks like everything exceeding 80 characters is
>> discarded. 
>>
>> By stop working I mean the status is and stays "Unavailable".
>>
>>
>> This works
>> exten => *1,hint,SIP/loicvoip1_1&IAX2/loicvoip1_1&SIP/loicvoip1_1_a1
>>
>> This does not work:
>> exten =>
>> *1,hint,SIP/bla1&SIP/bla2&SIP/bla3&SIP/bla4&SIP/bla9&SIP/bla5&SIP/bla6&SIP/bla7&SIP/loicvoip1_1&IAX2/loicvoip1_1&SIP/loicvoip1_1_a1
>>
>>
>> I tested on several asterisk 1.4 versions like 1.4.21*.
>>
>>
>> Is this a bug or something like working as designed?
>> 
>
> It's by design. 80 characters is likely to be the limit.
>
>   
>> Is there another
>> possibility to monitor a bigger number of channels?
>> 
>
> In Asterisk 1.6 you could build something with "Custom" hints
> and DEVICE_STATE().
>
>Philipp Kempgen
>
>   

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

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] ISDN Cause codes

[Martin Smith]

Hi Robert,

I'd recommend the following options for Dial() so that you corroborate
operator messages w/ cause codes:

 1. remove R and r - we've found this can supress operator recordings on
early audio
 2. likewise, remove m to disable MOH

Also, check the values of DIALSTATUS to compare to HANGUPCAUSE.

Good luck,

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Robert Boardman
> Sent: Friday, November 21, 2008 3:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] ISDN Cause codes
> 
> Thanks for the reply
> 
> Could you be a little more specific?
> 
> Thanks
> Robb
> 
> Martin Smith wrote:
> > Hi Robert,
> >
> > I'd suggest tweaking the Dial() arguments so that you (1) 
> allow early
> > audio, (2) don't force it play ringing to the calling party, and (3)
> > modify any other options to be as relaxed as possible. if 
> you make those
> > changes, you'll start hearing the operator message 
> recordings and those
> > are sometimes easier to reference against the cause codes.
> >
> > Cheers,
> >
> >
> > Martin Smith, Systems Developer
> > [EMAIL PROTECTED]
> > Bureau of Economic and Business Research
> > University of Florida
> > (352) 392-0171 Ext. 221 
> >
> >  
> >
> >   
> >> -Original Message-
> >> From: [EMAIL PROTECTED] 
> >> [mailto:[EMAIL PROTECTED] On Behalf Of 
> >> Robert Boardman
> >> Sent: Thursday, November 20, 2008 5:56 PM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: [asterisk-users] ISDN Cause codes
> >>
> >> Hi All
> >>
> >> Just been looking at stats for one of my sites, and I'm 
> >> conserned about 
> >> the number of error cause codes being returned from the telco
> >>
> >> for example
> >>
> >> 12000 calls processed
> >>
> >> 131 are cause code 31* normal. unspecified.*
> >>
> >> 139 are cause code 28 * invalid number format (address 
> incomplete).*
> >>
> >> 112 are cause code 1 *Unallocated (unassigned) number.
> >>
> >> *this adds up to about 3% of calls not completing.
> >>
> >> there are various other codes including 17 busy 34 channel 
> >> unavaliable 
> >> and 44 requested channel unavaliable, which add up to another 1%.*
> >> *
> >> the telco says there is no problem with the line, I'm trying to 
> >> understand what the problem could be
> >>
> >> now  alot of calls complete OK so I don't think is my configs
> >>
> >> Any advice would be appriciated
> >>
> >> Versions
> >> asterisk 1.4.21.1
> >> zaptel 1.4.12.1
> >>
> >>
> >> Robb
> >>
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> >> 
> >
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Re: [asterisk-users] ISDN Cause codes

[Robert Boardman]

Thanks for the reply

Could you be a little more specific?

Thanks
Robb

Martin Smith wrote:
> Hi Robert,
>
> I'd suggest tweaking the Dial() arguments so that you (1) allow early
> audio, (2) don't force it play ringing to the calling party, and (3)
> modify any other options to be as relaxed as possible. if you make those
> changes, you'll start hearing the operator message recordings and those
> are sometimes easier to reference against the cause codes.
>
> Cheers,
>
>
> Martin Smith, Systems Developer
> [EMAIL PROTECTED]
> Bureau of Economic and Business Research
> University of Florida
> (352) 392-0171 Ext. 221 
>
>  
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] 
>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>> Robert Boardman
>> Sent: Thursday, November 20, 2008 5:56 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] ISDN Cause codes
>>
>> Hi All
>>
>> Just been looking at stats for one of my sites, and I'm 
>> conserned about 
>> the number of error cause codes being returned from the telco
>>
>> for example
>>
>> 12000 calls processed
>>
>> 131 are cause code 31* normal. unspecified.*
>>
>> 139 are cause code 28 * invalid number format (address incomplete).*
>>
>> 112 are cause code 1 *Unallocated (unassigned) number.
>>
>> *this adds up to about 3% of calls not completing.
>>
>> there are various other codes including 17 busy 34 channel 
>> unavaliable 
>> and 44 requested channel unavaliable, which add up to another 1%.*
>> *
>> the telco says there is no problem with the line, I'm trying to 
>> understand what the problem could be
>>
>> now  alot of calls complete OK so I don't think is my configs
>>
>> Any advice would be appriciated
>>
>> Versions
>> asterisk 1.4.21.1
>> zaptel 1.4.12.1
>>
>>
>> Robb
>>
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>> 
>
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Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained

[Matt Florell]

On 11/21/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
> > On Fri, Nov 21, 2008 at 10:42 AM, Matt Florell <[EMAIL PROTECTED]> wrote:
>
>
> >> just keep in mind that in
>  >> my opinion the 1.4 tree did not become usable until 1.4.18 when most
>  >> of the major bugs were finally fixed.
>
>
> The longer you drag out the adoption curve, the longer it will take for
>  1.6 to catch up to that state.
>
>  Alex Balashov

We tried using 1.4 many times, and posted many bugs to the tracker.
Some of those bugs were ignored because I was told that I posted too
much information. We tried using most of the 1.4 releases and we did
post our results, even going as far as posting on the dev list and in
IRC, and I was always ignored or not gotten back to. I even offered
several times to donate my time to set up a system at Digium to
reproduce these bugs on demand and still had noone would take up my
offer. I talked to several VPs at Digium in-person and even Mark and
was always referred to someone else and nothing was ever done about
it. Then after 1.4.17 was released is when bug fixing became a higher
priority and they started implementing the release-cantidate process,
and myself and many others participated in that process and 1.4.18
went through several RCs with many many bug fixes and a lot of
testing, and 1.4.18 was the first fully tested release of the 1.4
tree.

As for 1.6, we haven't had anywhere near the time we did with 1.4 to
try to get it working for us, and there is a much steeper upgrade path
from 1.4 to 1.6 than there was between 1.4 and 1.6 which causes a lot
of other small issues in testing and implementation. Hopefully in the
next month or so we will have the time to spend on this.


MATT---

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Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained

[Alex Balashov]

> On Fri, Nov 21, 2008 at 10:42 AM, Matt Florell <[EMAIL PROTECTED]> wrote:

>> just keep in mind that in
>> my opinion the 1.4 tree did not become usable until 1.4.18 when most
>> of the major bugs were finally fixed.

The longer you drag out the adoption curve, the longer it will take for 
1.6 to catch up to that state.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained

[Steve Totaro]

On Fri, Nov 21, 2008 at 10:42 AM, Matt Florell <[EMAIL PROTECTED]> wrote:
> On 11/20/08, Steve Totaro <[EMAIL PROTECTED]> wrote:
>> On Thu, Nov 20, 2008 at 3:38 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>>  > On Thu, Nov 20, 2008 at 08:25:54AM +0100, Olivier wrote:
>>  >> 2008/11/17 Philipp Kempgen <[EMAIL PROTECTED]>
>>  >>
>>  >> > Tilghman Lesher schrieb:
>>  >> > > On Thursday 13 November 2008 08:16:42 Klaus Darilion wrote:
>>  >> > >> Is there somewhere a statement from Digium how long they will 
>> support
>>  >> > >> Asterisk 1.4?
>>  >> > >
>>
>> 0>> > > There is no statement, because we haven't even discussed when
>>
>> the EOL for
>>  >> > > 1.4 will be reached.  Certainly that means it won't happen for at 
>> least
>>  >> > the
>>  >> > > next 60 days, but beyond that, I really don't know.
>>  >> >
>>  >> > For the average non-techie user who does not want to compile
>>  >> > themselves that may sound funny (if not scary).
>>  >> >
>>  >> > When Debian Lenny (featuring Asterisk 1.4) is finally going to be
>>  >> > released that version might not even be supported any more.
>>  >>
>>  >>
>>  >> I think to a large extend, Asterisk is not to be considered as binary
>>  >> distributed at all, as many hardware it supports is not directly managed 
>> by
>>  >> kernel team.
>>  >
>>  > Interesting consideration. Debian Etch and RHEL5 are based on kernel
>>  > 2.6.18, but support quite a few hardware devices not included in that
>>  > kernel.
>>  >
>>  > If this issue bothers you, please help test the alternative timing
>>  > mechanism support now included in trunk.
>>  >
>>  > --
>>  >   Tzafrir Cohen
>>  > icq#16849755  jabber:[EMAIL PROTECTED]
>>  > +972-50-7952406   mailto:[EMAIL PROTECTED]
>>  > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>>  >
>>
>>
>> I still compile and install 1.2 for the most part, for call centers
>>  and large systems.
>>
>>  The few 1.4 installs that I have done have been for "medium" sized
>>  PBXs, say 50-70 phones/users and they have been trouble free for the
>>  most part.  Safe_asterisk may make some troubles transparent.
>>
>>  I am not really sure what 1.4 has over 1.2 for the average PBX installation.
>>
>>  Then you have the OpenPBX guys who forked 1.2, I know they have added
>>  functionality to 1.2, but the following puts me off.  Perhaps
>>  vaporware, perhaps not, it all relies on the devs.  You also have
>>  people like Matt Florell who have continued to add functionality to
>>  1.2 but since Digium won't take them, or the dev doesn't want to sign
>>  over their first born, they are hard to come by but certainly out
>>  there.
>>
>>  1.4 may follow the same path, being forked.
>>
>>  1.6 is not on my radar.
>>
>>
>>  --
>>  Thanks,
>>  Steve Totaro
>>  +18887771888 (Toll Free)
>>  +12409381212 (Cell)
>>  +12024369784 (Skype)
>
> Hello,
>
> We really just maintain a set of patches for 1.2 (just updated
> waitforsilence a couple weeks ago in fact) and we regularly install
> 1.2.30.2 in call center setups. It is rock solid and extremely proven
> in high-call-volume situations.
>
> We have started installing 1.4.21.2 on some systems that are not high
> load as well (1.4.22 has some strange issues with it we have noticed)
> because we do have clients requesting to use 1.4 for some of the nicer
> PBX functionality that it has as well as better SIP support.
>
> We test 1.6 periodically and we are very much looking forward to some
> of the great new features of it, but it crashes very quickly when
> trying to use it in call center situations. just keep in mind that in
> my opinion the 1.4 tree did not become usable until 1.4.18 when most
> of the major bugs were finally fixed.
>
> MATT---
>

As a fellow call center engineerimplementer I completely agree with
Matt's opinion.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio

[Jerry Geis]

I am using an AGI to setup the call to the first person,
then jumping into the dialplan with some Variables set.

Is the AGI messing up my channel???

My dialplan at that point looks like:

exten => 
call_cont,1,Dial(${CONT_CALLAT},${CONT_DIAL_TIMEOUT},${CONT_ONHOLD}tT)

CONT_CALLAT is Zap/1/506 where X is my number
CONT_DIAL_TIMEOUT is 60
CONT_ONHOLD is tT

Seems like this should still be working also.
How do I tell where/how my audio is getting blocked. Internal polycom to 
polycom works fine with this AGI,
the old 1.2 worked fine with this AGI, its just polycom to external 
world with the AGI is giving me a half channel.

Jerry


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Re: [asterisk-users] hint priority with 50 channels

[Philipp Kempgen]

Loic Didelot schrieb:

> I noticed that my hint priority stops working when I add to many
> extensions/channels. It looks like everything exceeding 80 characters is
> discarded. 
> 
> By stop working I mean the status is and stays "Unavailable".
> 
> 
> This works
> exten => *1,hint,SIP/loicvoip1_1&IAX2/loicvoip1_1&SIP/loicvoip1_1_a1
> 
> This does not work:
> exten =>
> *1,hint,SIP/bla1&SIP/bla2&SIP/bla3&SIP/bla4&SIP/bla9&SIP/bla5&SIP/bla6&SIP/bla7&SIP/loicvoip1_1&IAX2/loicvoip1_1&SIP/loicvoip1_1_a1
> 
> 
> I tested on several asterisk 1.4 versions like 1.4.21*.
> 
> 
> Is this a bug or something like working as designed?

It's by design. 80 characters is likely to be the limit.

> Is there another
> possibility to monitor a bigger number of channels?

In Asterisk 1.6 you could build something with "Custom" hints
and DEVICE_STATE().

   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] hint priority with 50 channels

[Danny Nicholas]

Just a guess, but since extensions.conf is basically a "card file", there
may be a 80 character limit to the line or data size.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Loic Didelot
Sent: Friday, November 21, 2008 11:52 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] hint priority with 50 channels

Hi,
I noticed that my hint priority stops working when I add to many
extensions/channels. It looks like everything exceeding 80 characters is
discarded. 

By stop working I mean the status is and stays "Unavailable".


This works
exten => *1,hint,SIP/loicvoip1_1&IAX2/loicvoip1_1&SIP/loicvoip1_1_a1

This does not work:
exten =>
*1,hint,SIP/bla1&SIP/bla2&SIP/bla3&SIP/bla4&SIP/bla9&SIP/bla5&SIP/bla6&SIP/b
la7&SIP/loicvoip1_1&IAX2/loicvoip1_1&SIP/loicvoip1_1_a1


I tested on several asterisk 1.4 versions like 1.4.21*.


Is this a bug or something like working as designed? Is there another
possibility to monitor a bigger number of channels?

Best regards,
Loic Didelot



-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
[EMAIL PROTECTED]
http://www.mixvoip.com


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Re: [asterisk-users] Best way to handle include files?

[Doug]

Thanks, Tzafrir, for your reply!

At 13:25 11/19/2008, Tzafrir Cohen wrote:
 >On Wed, Nov 19, 2008 at 01:14:55PM -0600, Doug wrote:
 >> Hi folks,
 >>
 >> I am building a new box.  Want it to look
 >> pretty much like an older Asterisk 1.2,
 >> Debian box that is in production.  The new
 >> box will used as a test box before we
 >> implement changes to the production box.
 >>
 >> New box:
 >> 
 >> # cat /etc/issue;  uname -a
 >> Debian GNU/Linux 4.0 \n \l
 >>
 >> Linux ServerName 2.6.18-6-686 #1 SMP Mon Oct 13
 >> 16:13:09 UTC 2008 i686 GNU/Linux
 >> 
 >>
 >> I've got Asterisk compiled and running:
 >>
 >> 
 >> # asterisk -rv
 >>== Parsing '/etc/asterisk/asterisk.conf': Found
 >>== Parsing '/etc/asterisk/extconfig.conf': Found
 >> Asterisk 1.2.30.2, Copyright (C) 1999 - 2007 Digium, Inc. and others.
 >> Created by Mark Spencer <[EMAIL PROTECTED]>
 >> Asterisk comes with ABSOLUTELY NO WARRANTY; 
type 'show warranty' for details.
 >> 
 >>
 >> The problem lies when I try to compile rxfax and
 >> txfax.  The compiler jumps out of the
 >>
 >>/usr/src/asterisk/asterisk/asterisk-1.2.30.2/apps/
 >>
 >> directory:
 >>
 >> /bin/sh: curl-config: command not found
 >> cc -fPIC   -c -o app_dial.o app_dial.c
 >> app_dial.c:37:22: error: asterisk.h: No such file or directory
 >> app_dial.c:39: error: expected declaration
 >> specifiers or â...â before string constant
 >>
 >> "asterisk.h" is located:
 >>
 >> # find / -name "asterisk.h"
 >> /usr/src/asterisk/asterisk/asterisk-1.2.30.2/include/asterisk.h
 >>
 >> I am finding that other Asterisk-related
 >> include files are located:
 >>
 >> /usr/include/asterisk/
 >>
 >> but, they have a recent time stamp.  I prefer
 >> a time stamp that indicated the last "real"
 >> modification date.
 >
 >[Use package management rather than gueswork?]

Do you mean:

# apt-get update
Get:1 http://ftp.uwsg.indiana.edu etch Release.gpg [386B]
Hit http://ftp.uwsg.indiana.edu etch Release
Get:2 http://security.debian.org etch/updates Release.gpg [189B]
Get:3 http://security.debian.org etch/updates Release [37.6kB]
Ign http://ftp.uwsg.indiana.edu etch/main Packages/DiffIndex
Ign http://ftp.uwsg.indiana.edu etch/non-free Packages/DiffIndex
Ign http://ftp.uwsg.indiana.edu etch/main Sources/DiffIndex
Ign http://ftp.uwsg.indiana.edu etch/non-free Sources/DiffIndex
Hit http://ftp.uwsg.indiana.edu etch/main Packages
Hit http://ftp.uwsg.indiana.edu etch/non-free Packages
Hit http://ftp.uwsg.indiana.edu etch/main Sources
Hit http://ftp.uwsg.indiana.edu etch/non-free Sources
Ign http://security.debian.org etch/updates/main Packages/DiffIndex
Ign http://security.debian.org etch/updates/contrib Packages/DiffIndex
Ign http://security.debian.org etch/updates/non-free Packages/DiffIndex
Ign http://security.debian.org etch/updates/main Sources/DiffIndex
Ign http://security.debian.org etch/updates/contrib Sources/DiffIndex
Ign http://security.debian.org etch/updates/non-free Sources/DiffIndex
Get:4 http://security.debian.org etch/updates/main Packages [291kB]
Hit http://security.debian.org etch/updates/contrib Packages
Hit http://security.debian.org etch/updates/non-free Packages
Get:5 http://security.debian.org etch/updates/main Sources [45.9kB]
Hit http://security.debian.org etch/updates/contrib Sources
Hit http://security.debian.org etch/updates/non-free Sources
Fetched 375kB in 1s (309kB/s)
Reading package lists... Done

http://www.google.com/search?q=asterisk+%22package+management%22

 >
 >>
 >> Researching on the Web, some people suggest
 >> copying all the include files to:
 >>
 >> /usr/include/asterisk/
 >
 >This is indeed normally installed by 'make install' of Asterisk.

Right.  Why do the .h files have the install date
instead of the last modified date?

 >
 >>
 >> Others suggest making a symbolic link that
 >> translates:
 >>
 >> /usr/include/asterisk/
 >>
 >> to:
 >>
 >> /usr/src/asterisk/asterisk/asterisk-1.2.30.2/include/
 >
 >Why do you actually want to keep the build directory around?

Well, we've got plenty of disk space, and it
gives a historical record of upgrades.

Why wouldn't we want to keep them around?


This seems to imply that the include files should
be copied into:

/usr/include/asterisk/

Is this correct?



 >
 >(Note that Asterisk modules don't link at build time with and Asterisk
 >component (e.g.: library), and hence the sterisk-devel only includes
 >only the header files)

I am confused.  Are you saying that when compiling
or recompiling the Asterisk modules don't link?
I am not exactly sure what you are saying.

Again, what is the best way to handle include files
so that rxfax and txfax will compile, and will allow
for future upgrades of Asterisk versions?


 >
 >--
 >   Tzafrir Cohen
 >icq#16849755  jabber:[EMAIL PROTECTED]

Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained

[Matt Florell]

On 11/21/08, Tilghman Lesher <[EMAIL PROTECTED]> wrote:
> On Friday 21 November 2008 09:42:12 Matt Florell wrote:
>  > On 11/20/08, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
> > >  You also have
>  > >  people like Matt Florell who have continued to add functionality to
>  > >  1.2 but since Digium won't take them, or the dev doesn't want to sign
>  > >  over their first born, they are hard to come by but certainly out
>  > >  there.
>  >
>
> > We really just maintain a set of patches for 1.2 (just updated
>  > waitforsilence a couple weeks ago in fact) and we regularly install
>  > 1.2.30.2 in call center setups. It is rock solid and extremely proven
>  > in high-call-volume situations.
>  >
>  > We have started installing 1.4.21.2 on some systems that are not high
>  > load as well (1.4.22 has some strange issues with it we have noticed)
>  > because we do have clients requesting to use 1.4 for some of the nicer
>  > PBX functionality that it has as well as better SIP support.
>  >
>  > We test 1.6 periodically and we are very much looking forward to some
>  > of the great new features of it, but it crashes very quickly when
>  > trying to use it in call center situations. just keep in mind that in
>  > my opinion the 1.4 tree did not become usable until 1.4.18 when most
>  > of the major bugs were finally fixed.
>
>
> Are you reporting these crashes in 1.6?  I'd like to know where they are, so
>  we can track them down and fix them.

By the time we get around to testing for any length of time there is
always another version released(including RCs and betas), we haven't
tested on the most recent 1.6 release and we don't really have the
resources to do intense debugging and bug reporting on 1.6 anytime in
the near future. We have tested two of the original beta releases as
well as the first 1.6.0.1 RC and they all had crash problems. We have
also had issues adjusting to using Dahdi on 1.6 since it is manditory
and you cannot use zaptel as you can with 1.4.

I am hoping to set up a system over the Holidays that I will only put
1.6 on that I will be able to do bug testing on, but from our
experience it is not easy to move from 1.2/1.4 over to 1.6 and back
again in a timely manner because of all of the major changes made to
Asterisk between 1.4 and 1.6.

MATT---

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Re: [asterisk-users] TrixBox problem...

[Andres]

Sounds like a deadlock.  Its not a Trixbox issue but an asterisk one.  
Your best bet is to upgrade asterisk itself.  No need to reinstall 
Trixbox.  Just download the asterisk source and compile/install.

And yes, we have installed several Trixbox systems that became unstable 
eventually.  The solution was to upgrade asterisk. 

Andres
http://www.neuroredes.com

Gregory Malsack wrote:

> Hey Everyone,
>
>  
>
> Here’s an email I received from a client who has a trixbox system that 
> has contracted with me for some custom dialplan programming.
>
>  
>
> 
>
> While I was away at a conference on Tuesday, our server crashed same 
> as before (it was “responsive” to GUI interface but no calls could go 
> in/out).  Our staff rebooted it Wednesday morning then I got a call 
> this morning that the phones were very staticky so they rebooted again 
> and that resolved the issues.  
>
>  
>
> I’m ready to resort to a nightly Cron reboot at 4am but that’s just 
> avoiding the underlying problem… would like to figure out why it’s 
> happening in the first place.
>
> 
>
>  
>
> Anyone else seem to have experienced a similar problem?
>
>  
>
> The clients system is a supermicro P4 2.4ghz, 1gb ram, 250gb hard 
> drive. The client is using eyebeam softphones, and is using sip 
> trunking to his phone companies services over a T1 line. This is the 
> second time he has called telling me that the system is locking up, 
> but apache seems to still work, the gui seems to still work, it just 
> simply doesn’t route calls. After they reboot everything is fine. 
> However, as noted above, they seem to have to reboot the system an 
> awful lot lately.
>
>  
>
> Thanks,
>
> Greg
>
>
> No virus found in this outgoing message.
> Checked by AVG.
> Version: 7.5.549 / Virus Database: 270.9.9/1803 - Release Date: 
> 11/21/2008 9:37 AM
>
>
>
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Re: [asterisk-users] TrixBox problem...

[Tim Nelson]

What is the call volume on this box? Depending on the version of Asterisk, 
maybe there are some memory leaks present causing calls to fail but everything 
else to keep working? 

Tim Nelson 
Systems/Network Support 
Rockbochs Inc. 
(218)727-4332 x105 

- "Gregory Malsack" wrote: 
> 


Hey Everyone, 



Here’s an email I received from a client who has a trixbox system that has 
contracted with me for some custom dialplan programming. 



 

While I was away at a conference on Tuesday, our server crashed same as before 
(it was “responsive” to GUI interface but no calls could go in/out). Our staff 
rebooted it Wednesday morning then I got a call this morning that the phones 
were very staticky so they rebooted again and that resolved the issues. 



I’m ready to resort to a nightly Cron reboot at 4am but that’s just avoiding 
the underlying problem… would like to figure out why it’s happening in the 
first place. 

 



Anyone else seem to have experienced a similar problem? 



The clients system is a supermicro P4 2.4ghz, 1gb ram, 250gb hard drive. The 
client is using eyebeam softphones, and is using sip trunking to his phone 
companies services over a T1 line. This is the second time he has called 
telling me that the system is locking up, but apache seems to still work, the 
gui seems to still work, it just simply doesn’t route calls. After they reboot 
everything is fine. However, as noted above, they seem to have to reboot the 
system an awful lot lately. 



Thanks, 

Greg 
> 

No virus found in this outgoing message. 
> Checked by AVG. 
> Version: 7.5.549 / Virus Database: 270.9.9/1803 - Release Date: 11/21/2008 
> 9:37 AM 
> 
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Re: [asterisk-users] Full Duplex

[Matthew Fredrickson]

Matt Riddell wrote:
> On 18/11/2008 9:46 a.m., Matthew Fredrickson wrote:
>> Singer X.J. Wang wrote:
>>> We've had the same issue. For calls that go between a SIP connection 
>>> (desktop phones) and Zaptel connections, there was a lot of problems 
>>> with half duplex. We switched
>>> from the Digium card to the Sangoma card and the problem went away.
>> Just for the record, he said that it happened regardless of protocol (IP 
>> to IP calls do not use the card based echo cancellers).
>>
>> Sorry for the problem you had.  However, I think that if you use a 
>> current version of our echo canceller board, you will find your issues 
>> resolved.  In fact, for a significant number of Digium's boards, Sangoma 
>> uses the exact same hardware echo canceller.
> 
> A few months ago, I had a similar problem and needed to pass:
> 
> vpmnlptype=4 vpmnlpmaxsupp=11
> 
> to resolve it. If I upgraded zaptel would this be fixed?

In a newer version of the firmware, it's very likely, although you would 
have to talk to technical support directly about it right now to get the 
updated version.

Matthew Fredrickson
Digium, Inc.


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[asterisk-users] TrixBox problem...

[Gregory Malsack]

Hey Everyone,

 

Here’s an email I received from a client who has a trixbox system that has 
contracted with me for some custom dialplan programming.

 



While I was away at a conference on Tuesday, our server crashed same as before 
(it was “responsive” to GUI interface but no calls could go in/out).  Our staff 
rebooted it Wednesday morning then I got a call this morning that the phones 
were very staticky so they rebooted again and that resolved the issues.   

 

I’m ready to resort to a nightly Cron reboot at 4am but that’s just avoiding 
the underlying problem… would like to figure out why it’s happening in the 
first place.



 

Anyone else seem to have experienced a similar problem?

 

The clients system is a supermicro P4 2.4ghz, 1gb ram, 250gb hard drive. The 
client is using eyebeam softphones, and is using sip trunking to his phone 
companies services over a T1 line. This is the second time he has called 
telling me that the system is locking up, but apache seems to still work, the 
gui seems to still work, it just simply doesn’t route calls. After they reboot 
everything is fine. However, as noted above, they seem to have to reboot the 
system an awful lot lately.

 

Thanks,

Greg


No virus found in this outgoing message.
Checked by AVG. 
Version: 7.5.549 / Virus Database: 270.9.9/1803 - Release Date: 11/21/2008 9:37 
AM
 
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Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained

[Tilghman Lesher]

On Friday 21 November 2008 09:42:12 Matt Florell wrote:
> On 11/20/08, Steve Totaro <[EMAIL PROTECTED]> wrote:
> >  You also have
> >  people like Matt Florell who have continued to add functionality to
> >  1.2 but since Digium won't take them, or the dev doesn't want to sign
> >  over their first born, they are hard to come by but certainly out
> >  there.
>
> We really just maintain a set of patches for 1.2 (just updated
> waitforsilence a couple weeks ago in fact) and we regularly install
> 1.2.30.2 in call center setups. It is rock solid and extremely proven
> in high-call-volume situations.
>
> We have started installing 1.4.21.2 on some systems that are not high
> load as well (1.4.22 has some strange issues with it we have noticed)
> because we do have clients requesting to use 1.4 for some of the nicer
> PBX functionality that it has as well as better SIP support.
>
> We test 1.6 periodically and we are very much looking forward to some
> of the great new features of it, but it crashes very quickly when
> trying to use it in call center situations. just keep in mind that in
> my opinion the 1.4 tree did not become usable until 1.4.18 when most
> of the major bugs were finally fixed.

Are you reporting these crashes in 1.6?  I'd like to know where they are, so
we can track them down and fix them.

-- 
Tilghman

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Re: [asterisk-users] Limit the number of users in a meetmeconference?

[Atis Lezdins]

On Fri, Nov 21, 2008 at 5:46 PM, Danny Nicholas <[EMAIL PROTECTED]> wrote:
> Armed with a little more information, here is a more realistic reply.
> In the 1.6.0.1 code, app_meetme.c defines maxusers in line 369 and sets the
> max value in line 870 to 0x7fff.
> Therefore changing line 870 would allow you to limit the maxusers.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
> Sent: Friday, November 21, 2008 9:05 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Limit the number of users in a
> meetmeconference?
>
> Hi Dan -
>
>>> I found the "maxusers" defined in meetme.c, but I'm
>>> not sure how this value is set.  Does anybody know
>>> if one can limit the number of users permitted in a
>>> meetme conference?  I know there's MeetmeCount(), but
>>> I'd rather avoid the dialplan logic and just set
>>> maxusers instead.
>>
>> That feature is primarily used with RealTime conferences.
>> The maxusers value is read from a database and enforced
>> on RealTime enable conferences.  This presumes you are
>> looking at 1.6.X or Trunk code...
>
> Ah.  No realtime for me, so I guess I'll just stick with using
> MeetmeCount() in the dialplan.  Thanks for the info!
>
>
> - Noah
>

If it's in realtime, then it should also work from config file. If
it's not, then file a bug.

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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[asterisk-users] hint priority with 50 channels

[Loic Didelot]

Hi,
I noticed that my hint priority stops working when I add to many
extensions/channels. It looks like everything exceeding 80 characters is
discarded. 

By stop working I mean the status is and stays "Unavailable".


This works
exten => *1,hint,SIP/loicvoip1_1&IAX2/loicvoip1_1&SIP/loicvoip1_1_a1

This does not work:
exten =>
*1,hint,SIP/bla1&SIP/bla2&SIP/bla3&SIP/bla4&SIP/bla9&SIP/bla5&SIP/bla6&SIP/bla7&SIP/loicvoip1_1&IAX2/loicvoip1_1&SIP/loicvoip1_1_a1


I tested on several asterisk 1.4 versions like 1.4.21*.


Is this a bug or something like working as designed? Is there another
possibility to monitor a bigger number of channels?

Best regards,
Loic Didelot



-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
[EMAIL PROTECTED]
http://www.mixvoip.com


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Re: [asterisk-users] Log level of 500 Server Internal Error.

[Tom Moore]

Hi,
I've started noticing these messages today myself specifically with
Broadvox.
Are you using this carrier or someone else?

Tom 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins
Sent: Friday, November 21, 2008 12:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Log level of 500 Server Internal Error.

Hi,

VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error"

I just noticed that i sometimes get those back from provider. They are
currently general SIP message log entries with verbose level 3.

I wonder if such SIP fails could generate at least WARNING in log?
Currently i'm checking logs for warnings and errors, so i probably
have missed those.. It would be great indication that something is not
ok - either outgoing trunk or local phone is bad.

Any opinions?

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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No virus found in this incoming message.
Checked by AVG - http://www.avg.com 
Version: 8.0.175 / Virus Database: 270.9.9/1803 - Release Date: 11/21/2008
9:37 AM


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Re: [asterisk-users] Log level of 500 Server Internal Error.

[Alex Balashov]

Atis Lezdins wrote:
> On Fri, Nov 21, 2008 at 7:32 PM, Alex Balashov
> <[EMAIL PROTECTED]> wrote:
>> Atis Lezdins wrote:
>>> Hi,
>>>
>>> VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error"
>>>
>>> I just noticed that i sometimes get those back from provider. They are
>>> currently general SIP message log entries with verbose level 3.
>>>
>>> I wonder if such SIP fails could generate at least WARNING in log?
>>> Currently i'm checking logs for warnings and errors, so i probably
>>> have missed those.. It would be great indication that something is not
>>> ok - either outgoing trunk or local phone is bad.
>> That would generate a lot of debate about what sorts of signaling error
>> classes are useful to include in the fixed logs and which aren't.
>>
>> Best thing to do is just to run your own packet capture and grep for
>> things of interest to you.
>>
> 
> Yes, that's what i would like to start.
> 
> If a call fails, i think it's reasonable enough to log a warning
> message. If i haven't seen this before, how would i know that it's bad
> and search for it? IMHO it's a good indication for network problem (as
> was midget packet warning recently)

Define "fails."  There are many different scenarios applicable to many 
different people's situations, and I doubt Asterisk can be set up to log 
them all.  SIP also has a complicated state machine;  sometimes call 
"failures" can occur further up the setup flow and not as an immediate 
failure response.

That, I think, is what I was trying to put forth as a possible reason 
why Asterisk doesn't do what you're asking, which is otherwise a fairly 
"obvious" thing to do.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Log level of 500 Server Internal Error.

[Atis Lezdins]

On Fri, Nov 21, 2008 at 7:32 PM, Alex Balashov
<[EMAIL PROTECTED]> wrote:
> Atis Lezdins wrote:
>> Hi,
>>
>> VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error"
>>
>> I just noticed that i sometimes get those back from provider. They are
>> currently general SIP message log entries with verbose level 3.
>>
>> I wonder if such SIP fails could generate at least WARNING in log?
>> Currently i'm checking logs for warnings and errors, so i probably
>> have missed those.. It would be great indication that something is not
>> ok - either outgoing trunk or local phone is bad.
>
> That would generate a lot of debate about what sorts of signaling error
> classes are useful to include in the fixed logs and which aren't.
>
> Best thing to do is just to run your own packet capture and grep for
> things of interest to you.
>

Yes, that's what i would like to start.

If a call fails, i think it's reasonable enough to log a warning
message. If i haven't seen this before, how would i know that it's bad
and search for it? IMHO it's a good indication for network problem (as
was midget packet warning recently)

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] Log level of 500 Server Internal Error.

[Alex Balashov]

Atis Lezdins wrote:
> Hi,
> 
> VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error"
> 
> I just noticed that i sometimes get those back from provider. They are
> currently general SIP message log entries with verbose level 3.
> 
> I wonder if such SIP fails could generate at least WARNING in log?
> Currently i'm checking logs for warnings and errors, so i probably
> have missed those.. It would be great indication that something is not
> ok - either outgoing trunk or local phone is bad.

That would generate a lot of debate about what sorts of signaling error 
classes are useful to include in the fixed logs and which aren't.

Best thing to do is just to run your own packet capture and grep for 
things of interest to you.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Alex Balashov]

Alex Balashov wrote:
> Yehavi Bourvine wrote:
> 
>> OK, but I still did not get a reply to my original question: Why using 
>> SIP registrar in front of Asterisk and not simply use bare Astersik? 
>> can't it handle the load? (remember - in my case it doesn't handle the 
>> RTP, only signalling). Can't it handle so much registrations? (I am 
>> using realtime DB, it is has any relevance).
> 
> My experience has shown that using a dedicated registrar for large 
> installs is more effective;  it doesn't tie up resources on the Asterisk 
> box with all those registration refreshes, for one.  A product built to 
> be a high-throughput standalone registrar will handle the concurrency 
> requirements and perform better.
> 

Some of this is just a general design principle, nothing specific to 
registration.  Once a VoIP platform gets to be big enough, a lot of the 
logical elements that go into it get centralised into distinct and 
dedicated components that are federated into a delivery platform.  It's 
no longer considered a good idea at that point to have one process 
perform many different functions that have varying concurrency and 
blocking implications.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] Log level of 500 Server Internal Error.

[Atis Lezdins]

Hi,

VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error"

I just noticed that i sometimes get those back from provider. They are
currently general SIP message log entries with verbose level 3.

I wonder if such SIP fails could generate at least WARNING in log?
Currently i'm checking logs for warnings and errors, so i probably
have missed those.. It would be great indication that something is not
ok - either outgoing trunk or local phone is bad.

Any opinions?

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Alex Balashov]

Yehavi Bourvine wrote:

> OK, but I still did not get a reply to my original question: Why using 
> SIP registrar in front of Asterisk and not simply use bare Astersik? 
> can't it handle the load? (remember - in my case it doesn't handle the 
> RTP, only signalling). Can't it handle so much registrations? (I am 
> using realtime DB, it is has any relevance).

My experience has shown that using a dedicated registrar for large 
installs is more effective;  it doesn't tie up resources on the Asterisk 
box with all those registration refreshes, for one.  A product built to 
be a high-throughput standalone registrar will handle the concurrency 
requirements and perform better.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Yehavi Bourvine]

> > I know that in the past there have been people on this list who have

> done very large scale asterisk deployments.  Not sure if any of them
> > are still around to comment.
> >
> > With that many extensions, I'll second using a SIP registrar like
> > Freeswitch or OpenSer.  Just use asterisk to provide the services.
>
>
OK, but I still did not get a reply to my original question: Why using SIP
registrar in front of Asterisk and not simply use bare Astersik? can't it
handle the load? (remember - in my case it doesn't handle the RTP, only
signalling). Can't it handle so much registrations? (I am using realtime DB,
it is has any relevance).

   Thanks! __Yehavi:
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Re: [asterisk-users] upgrade from 1.2 to 1.4 and now half channelaudio

[Danny Nicholas]

You could try un-commenting "duplex=2" in rpt.conf and changing it to
duplex=3.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Friday, November 21, 2008 11:05 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] upgrade from 1.2 to 1.4 and now half
channelaudio

>
> You could trying changing this in sip.cfg
>  To
> 
>   

Just tried that - rebooted my polycom and still half audio.
Thanks,

Jerry

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Gonzalo Servat]

On Fri, Nov 21, 2008 at 2:49 PM, Noah Miller <[EMAIL PROTECTED]>wrote:

> > And FreeSWITCH can't handle that?
>
> Freeswitch can provide many PBX features with additional modules, but
> asterisk can provide more, and its implementations of such items are
> more time tested.  One of freeswitch's big strengths is its ability to
> handle many SIP registrations.  This is not asterisk's strength (at
> least not historically).  One of Asterisk's big strengths is its
> multitude of services and features.  This is not freeswitch's
> strength.  Combine freeswitch and asterisk to get the best of both
> worlds.
>

I was preparing a reply that would argue the need to have Asterisk involved
if FreeSWITCH is there, but given the name of the list and the potential to
piss people off, I'll leave it at that.

- Gonzalo
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Re: [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio

[Jerry Geis]

>
> You could trying changing this in sip.cfg
>  To
> 
>   

Just tried that - rebooted my polycom and still half audio.
Thanks,

Jerry

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Re: [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio

[Danny Nicholas]

You could trying changing this in sip.cfg
mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Friday, November 21, 2008 10:28 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio

Hi all,

I upgraded from asterisk 1.2.23 and zaptel 1.2.19
to asterisk 1.4.18 and zaptel 1.4.12.1
I use polycom 501 phones internally.

Everything seems fine. I can pick up the phone and call out,
calls coming in work just fine.

The issue I see is when the system first calls me,
then calls someone else. This works if its polycom to polycom. I hear 
audio full channel.
If I do  polycom to external line like a cell I only get HALF channel audio.
At this time they can hear me but I cannot hear them.

What might this be???

Jerry

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Alex Balashov]

Gonzalo Servat wrote:
> On Fri, Nov 21, 2008 at 1:48 PM, Alex Balashov 
> <[EMAIL PROTECTED] > wrote:
> 
> Gonzalo Servat wrote:
>  > On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller
> <[EMAIL PROTECTED] 
>  >  >> wrote:
>  >
>  > [..snip..]
>  >
>  > With that many extensions, I'll second using a SIP registrar like
>  > Freeswitch or OpenSer.  Just use asterisk to provide the
> services.
>  >
>  >
>  > Is Asterisk even needed?
> 
> Potentially, no.  But if you intend to provide subscriber/PBX features,
> it is needed as a UA feature box(s).
> 
> 
> And FreeSWITCH can't handle that?

I suppose FreeSWITCH could, if you're so inclined.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Alex Balashov]

Gonzalo Servat wrote:
> On Fri, Nov 21, 2008 at 1:48 PM, Alex Balashov 
> <[EMAIL PROTECTED] > wrote:
> 
> Gonzalo Servat wrote:
>  > On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller
> <[EMAIL PROTECTED] 
>  >  >> wrote:
>  >
>  > [..snip..]
>  >
>  > With that many extensions, I'll second using a SIP registrar like
>  > Freeswitch or OpenSer.  Just use asterisk to provide the
> services.
>  >
>  >
>  > Is Asterisk even needed?
> 
> Potentially, no.  But if you intend to provide subscriber/PBX features,
> it is needed as a UA feature box(s).
> 
> 
> And FreeSWITCH can't handle that?

I suppsoe FreeSWITCH could, if you're so inclined.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Noah Miller]

>> > Is Asterisk even needed?
>>
>> Potentially, no.  But if you intend to provide subscriber/PBX features,
>> it is needed as a UA feature box(s).
>
> And FreeSWITCH can't handle that?

Freeswitch can provide many PBX features with additional modules, but
asterisk can provide more, and its implementations of such items are
more time tested.  One of freeswitch's big strengths is its ability to
handle many SIP registrations.  This is not asterisk's strength (at
least not historically).  One of Asterisk's big strengths is its
multitude of services and features.  This is not freeswitch's
strength.  Combine freeswitch and asterisk to get the best of both
worlds.


- Noah

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Gonzalo Servat]

On Fri, Nov 21, 2008 at 1:48 PM, Alex Balashov <[EMAIL PROTECTED]>wrote:

> Gonzalo Servat wrote:
> > On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller <[EMAIL PROTECTED]
> > > wrote:
> >
> > [..snip..]
> >
> > With that many extensions, I'll second using a SIP registrar like
> > Freeswitch or OpenSer.  Just use asterisk to provide the services.
> >
> >
> > Is Asterisk even needed?
>
> Potentially, no.  But if you intend to provide subscriber/PBX features,
> it is needed as a UA feature box(s).
>

And FreeSWITCH can't handle that?

- Gonzalo
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[asterisk-users] OT - SIP message encoding to enhance text display

[Olivier]

Hi,

I've read RFC3428 which presents SIP MESSAGE.
Is there any extension or encoding scheme working with SIP MESSAGE that
would enhance text display with blinking or underlining attributes ?
This could be useful to notify SIP hardphone users with some important
events such being in Do Not Disturb mode.

Regards
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[asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio

[Jerry Geis]

Hi all,

I upgraded from asterisk 1.2.23 and zaptel 1.2.19
to asterisk 1.4.18 and zaptel 1.4.12.1
I use polycom 501 phones internally.

Everything seems fine. I can pick up the phone and call out,
calls coming in work just fine.

The issue I see is when the system first calls me,
then calls someone else. This works if its polycom to polycom. I hear 
audio full channel.
If I do  polycom to external line like a cell I only get HALF channel audio.
At this time they can hear me but I cannot hear them.

What might this be???

Jerry

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[RE Kushner List Account]

Noah Miller wrote:
>
>> and is only one of the roads that
>> leads to Hell (I prefer Patterson Lake Road myself since I drive in from
>> the North East).
>> 
>
> Hmm.  You must live near Ann Arbor.
>
>   

No, northern suburbs of Detroit.  M-59 to US-23 S to M-36 W..To S. 
Howell St..Patterson Lake Rd..To Hell

Ann Arbor is quite a bit South of Hell.  Actually it's been some time 
since I've been to Hell but I'm sure it's frozen over today ;-)

-Ron


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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Alex Balashov]

Gonzalo Servat wrote:
> On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller <[EMAIL PROTECTED] 
> > wrote:
> 
> [..snip..]
> 
> With that many extensions, I'll second using a SIP registrar like
> Freeswitch or OpenSer.  Just use asterisk to provide the services.
> 
> 
> Is Asterisk even needed?

Potentially, no.  But if you intend to provide subscriber/PBX features, 
it is needed as a UA feature box(s).

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Re: [asterisk-users] Limit the number of users in a meetmeconference?

[Danny Nicholas]

Armed with a little more information, here is a more realistic reply.
In the 1.6.0.1 code, app_meetme.c defines maxusers in line 369 and sets the
max value in line 870 to 0x7fff.  
Therefore changing line 870 would allow you to limit the maxusers.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent: Friday, November 21, 2008 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Limit the number of users in a
meetmeconference?

Hi Dan -

>> I found the "maxusers" defined in meetme.c, but I'm
>> not sure how this value is set.  Does anybody know
>> if one can limit the number of users permitted in a
>> meetme conference?  I know there's MeetmeCount(), but
>> I'd rather avoid the dialplan logic and just set
>> maxusers instead.
>
> That feature is primarily used with RealTime conferences.
> The maxusers value is read from a database and enforced
> on RealTime enable conferences.  This presumes you are
> looking at 1.6.X or Trunk code...

Ah.  No realtime for me, so I guess I'll just stick with using
MeetmeCount() in the dialplan.  Thanks for the info!


- Noah

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Gonzalo Servat]

On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller <[EMAIL PROTECTED]>wrote:

> [..snip..]

With that many extensions, I'll second using a SIP registrar like
> Freeswitch or OpenSer.  Just use asterisk to provide the services.
>

Is Asterisk even needed?

- Gonzalo
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Re: [asterisk-users] Ping

[Atis Lezdins]

On Fri, Nov 21, 2008 at 4:59 PM, Sebastian Milioto <[EMAIL PROTECTED]> wrote:
> Ping
>
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Pong

GMail's preview looks fun - "Ping -- Bandwidth and Colocation Provided
by http://www.api-digital.com";

Regards,
Atis


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VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
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Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained

[Matt Florell]

On 11/20/08, Steve Totaro <[EMAIL PROTECTED]> wrote:
> On Thu, Nov 20, 2008 at 3:38 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>  > On Thu, Nov 20, 2008 at 08:25:54AM +0100, Olivier wrote:
>  >> 2008/11/17 Philipp Kempgen <[EMAIL PROTECTED]>
>  >>
>  >> > Tilghman Lesher schrieb:
>  >> > > On Thursday 13 November 2008 08:16:42 Klaus Darilion wrote:
>  >> > >> Is there somewhere a statement from Digium how long they will support
>  >> > >> Asterisk 1.4?
>  >> > >
>
> 0>> > > There is no statement, because we haven't even discussed when
>
> the EOL for
>  >> > > 1.4 will be reached.  Certainly that means it won't happen for at 
> least
>  >> > the
>  >> > > next 60 days, but beyond that, I really don't know.
>  >> >
>  >> > For the average non-techie user who does not want to compile
>  >> > themselves that may sound funny (if not scary).
>  >> >
>  >> > When Debian Lenny (featuring Asterisk 1.4) is finally going to be
>  >> > released that version might not even be supported any more.
>  >>
>  >>
>  >> I think to a large extend, Asterisk is not to be considered as binary
>  >> distributed at all, as many hardware it supports is not directly managed 
> by
>  >> kernel team.
>  >
>  > Interesting consideration. Debian Etch and RHEL5 are based on kernel
>  > 2.6.18, but support quite a few hardware devices not included in that
>  > kernel.
>  >
>  > If this issue bothers you, please help test the alternative timing
>  > mechanism support now included in trunk.
>  >
>  > --
>  >   Tzafrir Cohen
>  > icq#16849755  jabber:[EMAIL PROTECTED]
>  > +972-50-7952406   mailto:[EMAIL PROTECTED]
>  > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>  >
>
>
> I still compile and install 1.2 for the most part, for call centers
>  and large systems.
>
>  The few 1.4 installs that I have done have been for "medium" sized
>  PBXs, say 50-70 phones/users and they have been trouble free for the
>  most part.  Safe_asterisk may make some troubles transparent.
>
>  I am not really sure what 1.4 has over 1.2 for the average PBX installation.
>
>  Then you have the OpenPBX guys who forked 1.2, I know they have added
>  functionality to 1.2, but the following puts me off.  Perhaps
>  vaporware, perhaps not, it all relies on the devs.  You also have
>  people like Matt Florell who have continued to add functionality to
>  1.2 but since Digium won't take them, or the dev doesn't want to sign
>  over their first born, they are hard to come by but certainly out
>  there.
>
>  1.4 may follow the same path, being forked.
>
>  1.6 is not on my radar.
>
>
>  --
>  Thanks,
>  Steve Totaro
>  +18887771888 (Toll Free)
>  +12409381212 (Cell)
>  +12024369784 (Skype)

Hello,

We really just maintain a set of patches for 1.2 (just updated
waitforsilence a couple weeks ago in fact) and we regularly install
1.2.30.2 in call center setups. It is rock solid and extremely proven
in high-call-volume situations.

We have started installing 1.4.21.2 on some systems that are not high
load as well (1.4.22 has some strange issues with it we have noticed)
because we do have clients requesting to use 1.4 for some of the nicer
PBX functionality that it has as well as better SIP support.

We test 1.6 periodically and we are very much looking forward to some
of the great new features of it, but it crashes very quickly when
trying to use it in call center situations. just keep in mind that in
my opinion the 1.4 tree did not become usable until 1.4.18 when most
of the major bugs were finally fixed.

MATT---

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[asterisk-users] [SOLVED] TDM400 (?) zap hangup

[Roderick A. Anderson]

Roderick A. Anderson wrote:
> And if that ain't confusing I don't know what would be.
> 
> I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago 
> and ended up never using it.  Passed it along to a friend who is having 
> some problems with it.  (He isn't on this list.)
> 
> We've both tried searches using Google but haven't been able to find
> anything that helps.  So this is more a question of
> what-the-heck-should-we-be-searching-for. :-)
> 
> The TDM400 works taking inbound calls and gives a dial tone when the
> phone is picked up but as soon as a key is pressed the line (Asterisk
> says) hangs up.  Asterisk is configured based on a working system but
> that one only has one module inbound (FXO?)  The outbound settings are
> based on docs from voip-info.org.
> 
> Does this ring a bell for anyone?  No pun intended.
> 
> Unfortunately the system is 35 miles away and I haven't got the logs
> handy so I can't provide more right now.  Just hoping for a clue on
> search terms.

Thanks to Tzafrir Cohen and Jared Smith we've solved the problem.

It was a "A Series of Unfortunate Events".  The main one was, there was 
no (and then an incorrect) context= for the ZAP channel.  The incorrect 
one came about because of a miss-communication while testing.  We were 
able to dial-out but the logic in the dialplan to select a context for 
local calls, toll-free, etc. was missing.  Once we got the channel set 
to the correct context all was well.


Again thanks,
Rod
-- 
> TIA,
> Rod


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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Alex Balashov]

Noah Miller wrote:

> With that many extensions, I'll second using a SIP registrar like
> Freeswitch or OpenSer.  Just use asterisk to provide the services.

Third.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Noah Miller]

> Due diligence is required on anything 10,000 people are going to be
> pounding on. Undersizing is common,

I think due diligence is THE key with any open source solution,
including asterisk.  I'll admit that I pretty badly screwed up one
asterisk installation because I didn't adequately prepare it (shipped
it to the customer and had their IT staff install - bad plan).  And
while I've never done a system anywhere near 10K extensions, I've had
good experiences with some large-ish installations because I budgeted
in the time for research and testing.

I know that in the past there have been people on this list who have
done very large scale asterisk deployments.  Not sure if any of them
are still around to comment.

With that many extensions, I'll second using a SIP registrar like
Freeswitch or OpenSer.  Just use asterisk to provide the services.


> and is only one of the roads that
> leads to Hell (I prefer Patterson Lake Road myself since I drive in from
> the North East).

Hmm.  You must live near Ann Arbor.


- Noah

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Re: [asterisk-users] Limit the number of users in a meetme conference?

[Noah Miller]

Hi Dan -

>> I found the "maxusers" defined in meetme.c, but I'm
>> not sure how this value is set.  Does anybody know
>> if one can limit the number of users permitted in a
>> meetme conference?  I know there's MeetmeCount(), but
>> I'd rather avoid the dialplan logic and just set
>> maxusers instead.
>
> That feature is primarily used with RealTime conferences.
> The maxusers value is read from a database and enforced
> on RealTime enable conferences.  This presumes you are
> looking at 1.6.X or Trunk code...

Ah.  No realtime for me, so I guess I'll just stick with using
MeetmeCount() in the dialplan.  Thanks for the info!


- Noah

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[asterisk-users] Ping

[Sebastian Milioto]

Ping
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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Alex Balashov]

Al Baker wrote:

> Remember - You are going from a CARRIER GRADE purpose built piece of 
> hardware with Software built under a rigid CMM with extensive 
> "soak-testing" to software that has been developed under , shall we say, 
> a somewhat less rigid and stringent methodology.
> You will be moving from an environment supported by hundreds of highly 
> trained people, some with decades of TELCO experience
> to one where you support comes from a somewhat less seasoned group of 
> individuals.

But in choosing "carrier grade" (everyone calls their stuff that) 
vendors you are also going to a much smaller installed base and much 
lower total reporting and QA pool.  I would take the sheer number and 
dynamism of the Asterisk installed base over their comparatively limited 
deployments, even if we grant the unsubstantiated premise that the 
latter is developed under a less rigid and stringent methodology.

Let me put it this way:  if I wrote a piece of software and sold it to 
10 customers, it won't matter for overall product quality that I fix the 
problems they report and maintain it for them under the guidance of a 
"rigid" and "stringent" methodology.  That's nice.  Hope it fixes their 
problems.  It is really of comparatively minor benefit to prospective 
future adopters.  It's not nearly as valuable as simply doing the best I 
can with bug reports and test cases from hundreds of users.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities

[Alex Balashov]

Alex Balashov wrote:
> Jason Aarons (US) wrote:
> 
>> Just switching from Nortel to something else may not eliminate 
>> hardware/software failures, or prevent those without experience from 
>> pushing the enter key at the wrong time. 
> 
> One also has to keep in mind - Asterisk, like any large open-source 
> project, gets a lot more QA, patches and bug fixes than any commercial 
> product sold in the intra-industrial channel (i.e. excluding consumer 
> mass-market stuff) ever will!  It has a massive installed base, many 
> users reporting bugs through an open and easy to understand process, and 
> a large community either directly or derivatively involved in 
> contributing fixes and testing code.
> 
> How much installed base from which to harness that kind of large-scale 
> technical feedback does Nortel have?  Avaya?  Cisco?
> 
> Asterisk has by far the best QA mechanism.  In terms of potential bugs 
> that impact "mission-critical" availability, I would feel better using 
> it than any of these black-box, proprietary vendor solutions any day.
> 

Also, if there is a show-stopping bug, it can be addressed in a 
relatively expedient manner, especially if you are paying Digium for 
support.

With the other guys, you're going to have to wait for Service Pack 8 
Patchlevel 4 Release 2 Build 3789 in 12-24 months.  It might have a fix. 
  Maybe.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] A way to run extenrnotify when IMAP events take place...

[Philipp Kempgen]

Danny Nicholas schrieb:
> Here is a "Dirty" solution - create a PERL or other script to "listen" for
> changes to voicemail DB/Dir.  When VM is deleted, launch script to turn off
> Cisco MWI (should be simple since you are turning on with script).   Not
> "Best" solution, just workable one.

Yeah. If all else should fail there are various dirty solutions
such as listening to events on the manager interface, INotify,
implementing a SMDI interface yourself ...

   Philipp Kempgen

-- 
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Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities

[Alex Balashov]

Jason Aarons (US) wrote:

> Just switching from Nortel to something else may not eliminate 
> hardware/software failures, or prevent those without experience from 
> pushing the enter key at the wrong time. 

One also has to keep in mind - Asterisk, like any large open-source 
project, gets a lot more QA, patches and bug fixes than any commercial 
product sold in the intra-industrial channel (i.e. excluding consumer 
mass-market stuff) ever will!  It has a massive installed base, many 
users reporting bugs through an open and easy to understand process, and 
a large community either directly or derivatively involved in 
contributing fixes and testing code.

How much installed base from which to harness that kind of large-scale 
technical feedback does Nortel have?  Avaya?  Cisco?

Asterisk has by far the best QA mechanism.  In terms of potential bugs 
that impact "mission-critical" availability, I would feel better using 
it than any of these black-box, proprietary vendor solutions any day.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities

[Jason Aarons (US)]

Just switching from Nortel to something else may not eliminate
hardware/software failures, or prevent those without experience from
pushing the enter key at the wrong time.  You have to consider the two
professionals actually cost considerably more than just salary, due to
taxes, 401k, benefits and after they gain knowledge of your Asterisk
will they go to a VAR to make big money?  It's a catch-22.

 

I would start with a small pilot of anything (Nortel VoIP, Cisco VoIP,
Asterisk) and use it to build your knowledge/experience and then make an
informed decision versus a rush decision based upon how last week went
with the legacy equipment.  You might find you need to plan for PoE long
term, or have network issues, etc, etc.

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yehavi
Bourvine
Sent: Friday, November 21, 2008 6:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Large Asterisk installarions (~10,000
extensions), preferably at universities

 


Thanks to everyone who replies so far!

 

We have Nortel PBX'es with a support contract from one of the local VARs
(Nortel does not give direct support here). In the last two weeks we had
one of our exchanges down for three half days; one was after a failure,
and the other two were when the technician came to fix remainders of the
original problem and just did a "5 seconds restart which won't even cut
calls". Yeh, the "5 seconds" took 6-7 hours... BTW, they still do not
know what was the original problem.

 

So, why won't we save the big bucks we pay them, hire two professionals
(who cost less) and support an open source code by ourselves? This way
we depend on ourselves only.

 

 Thanks, __Yehavi:


 

2008/11/21 Grygoriy Dobrovolskyy <[EMAIL PROTECTED]>

 

2008/11/21 Yehavi Bourvine <[EMAIL PROTECTED]>

Hello,

 

  Our university has to upgrade soon its old Nortel PBX's which
holds around 10,000 extensions tied to 5 PBXes. Up to now we thought
about commercial solutions but now there is a window openning for open
source solution. However, I need examples to convince that this solution
is feasible, and preferably at other universities.

 

Are there any pointers for such installations?

 

   Thanks! __Yehavi:

 

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Hello very interesting project you have, however asterisk is not
a registry server, i suggest that you use opensips/opense/kamalio for
your registrar, from where you dispatch to you asterisk servers, inside
a good environment with a controlled network and nice tagged voip flow
you could acheve a good results. 


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Re: [asterisk-users] A way to run extenrnotify when IMAP events take place...

[Danny Nicholas]

Here is a "Dirty" solution - create a PERL or other script to "listen" for
changes to voicemail DB/Dir.  When VM is deleted, launch script to turn off
Cisco MWI (should be simple since you are turning on with script).   Not
"Best" solution, just workable one.  I'm doing similar thing with my VM - I
look for new VM, run soz to turn up volume, then send new wav file to
recipient in e-mail so they can listen in Media Player or on Iphone instead
of dialing in.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: Friday, November 21, 2008 6:58 AM
To: Asterisk Users
Subject: Re: [asterisk-users] A way to run extenrnotify when IMAP events
take place...

Jeffrey Phelps schrieb:
> But how do I get it to run a script??  I don't have any SMDI Interfaces,
> so I wouldn't be able to put anything in the config...

I thought Cisco CallManager had SMDI so that might have been an
alternative solution.
All I can tell is that the normal externnotify command in
voicemail.conf works well for me - without IMAP storage that
is.
Sorry for not being of any help here. Maybe someone else can
comment.

> Jeffrey Phelps schrieb:
> 
>> I have IMAP voicemail working with Exchange 2003 using a single
> username
> 
>> and password for multiple mailboxes.
> 
>  
> 
>> Right now, I am setting up asterisk to use voicemail with my Cisco
> Call
> 
>> Manager (Which I detest BTW...) and I have everything working, EXCEPT:
> 
>  
> 
>> I cannot get my externnotify script to run when any changes have been
> 
>> made to the VoiceMail...
> 
>  
> 
>> Scenario:
> 
>  
> 
>> Bob gets a call  -> Bob rejects call to voicemail
> 
>  
> 
>> Caller leaves Bob a voicemail  -> externnotify calls script which
> turns
> 
>> on his Cisco MWI.
> 
>  
> 
>> Bob checks Voicemail  ->  Bob deletes Voicemail  -> asterisk says that
> 
>> the voicemail was deleted, but doesn't run my script again to turn off
> 
>> the Cisco MWI.
> 
>  
> 
>> I would just like to know if there is any work around for this.
> 
>> OR.  Maybe Someone is working on adding this into the code
> 
>> so that it works...
> 
>  
> 
>> I'm running * 1.6.1-beta2
> 
>  
> 
> afaicr I read something which might be related in doc/smdi.txt.


   Philipp Kempgen

-- 
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Re: [asterisk-users] ISDN Cause codes

[Martin Smith]

Hi Robert,

I'd suggest tweaking the Dial() arguments so that you (1) allow early
audio, (2) don't force it play ringing to the calling party, and (3)
modify any other options to be as relaxed as possible. if you make those
changes, you'll start hearing the operator message recordings and those
are sometimes easier to reference against the cause codes.

Cheers,


Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Robert Boardman
> Sent: Thursday, November 20, 2008 5:56 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] ISDN Cause codes
> 
> Hi All
> 
> Just been looking at stats for one of my sites, and I'm 
> conserned about 
> the number of error cause codes being returned from the telco
> 
> for example
> 
> 12000 calls processed
> 
> 131 are cause code 31* normal. unspecified.*
> 
> 139 are cause code 28 * invalid number format (address incomplete).*
> 
> 112 are cause code 1 *Unallocated (unassigned) number.
> 
> *this adds up to about 3% of calls not completing.
> 
> there are various other codes including 17 busy 34 channel 
> unavaliable 
> and 44 requested channel unavaliable, which add up to another 1%.*
> *
> the telco says there is no problem with the line, I'm trying to 
> understand what the problem could be
> 
> now  alot of calls complete OK so I don't think is my configs
> 
> Any advice would be appriciated
> 
> Versions
> asterisk 1.4.21.1
> zaptel 1.4.12.1
> 
> 
> Robb
> 
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[asterisk-users] SPA2100 transfer to ASTERISK CID

[Sebastian Milioto]

Hi all,

I have around 100 SPA2100 registered in my provider openSER.
I'd like to add an Asterisk registered into openSER, to the network, to
deploy voicemail service for those SPAs.
Due to administration access levels, I have no access to SER box, so I'm
wondering if that possible:

- Some foreign user (say A) calls one of my SPA (say B).
- B don't answer. So.. B  SPA is setted up to transfer in 20 seconds on no
answer to the number in Asterisk.

All ok so far, however, in Asterisk I receive de caller ID = A, but I need B
CID. Having B caller ID I could let A leave the message into B mailbox.

Can anybody helpme with that please?

Thanks very much in advance

Sebastian
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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[RE Kushner List Account]

Yehavi Bourvine wrote:
>
> Thanks to everyone who replies so far!
>  
> We have Nortel PBX'es with a support contract from one of the local 
> VARs (Nortel does not give direct support here). In the last two weeks 
> we had one of our exchanges down for three half days; one was after a 
> failure, and the other two were when the technician came to fix 
> remainders of the original problem and just did a "5 seconds restart 
> which won't even cut calls". Yeh, the "5 seconds" took 6-7 hours... 
> BTW, they still do not know what was the original problem.
>  

Well, as everyone knows people don't generally have long memories, but I 
do.  Back in the 1980s Northern Telecom installed a DMS-100 for 
Ameritech in Southfield, Michigan. Somewhere along the way the switching 
matrix was somehow undersized, fingers pointed both ways and heated 
words were exchanged between companies.  Internally the switch was 
called Yo-Yo by the Ameritech employees.

Just because you spent millions of dollars on a solution doesn't 
guarantee you five nines. Sometimes it costs millions more just to make 
it work right, both in fines and vendor fixes.

Because of the Southfield fiasco when the largest exchange came up years 
later the contract went to AT&T Bell Labs to install the largest 5ESS in 
the world in the Royal Oak exchange. Nortel wasn't even considered. 
Nortel also lost quite a bit of Ameritech business to the Siemens EWSD. 
Ameritech's Ohio Bell wouldn't even touch the DMS-100 for many years 
because of what happened in Southfield. This was at a time when all the 
switches needed to go digital to reduce power costs and comply with 
federal law, so there was a urgency to get hardware in place ASAP. Even 
with the pressure the DMS-100 was avoided.

Due diligence is required on anything 10,000 people are going to be 
pounding on. Undersizing is common, and is only one of the roads that 
leads to Hell (I prefer Patterson Lake Road myself since I drive in from 
the North East).

-Ron


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Re: [asterisk-users] A way to run extenrnotify when IMAP events take place...

[Philipp Kempgen]

Jeffrey Phelps schrieb:
> But how do I get it to run a script??  I don't have any SMDI Interfaces,
> so I wouldn't be able to put anything in the config...

I thought Cisco CallManager had SMDI so that might have been an
alternative solution.
All I can tell is that the normal externnotify command in
voicemail.conf works well for me - without IMAP storage that
is.
Sorry for not being of any help here. Maybe someone else can
comment.

> Jeffrey Phelps schrieb:
> 
>> I have IMAP voicemail working with Exchange 2003 using a single
> username
> 
>> and password for multiple mailboxes.
> 
>  
> 
>> Right now, I am setting up asterisk to use voicemail with my Cisco
> Call
> 
>> Manager (Which I detest BTW...) and I have everything working, EXCEPT:
> 
>  
> 
>> I cannot get my externnotify script to run when any changes have been
> 
>> made to the VoiceMail...
> 
>  
> 
>> Scenario:
> 
>  
> 
>> Bob gets a call  -> Bob rejects call to voicemail
> 
>  
> 
>> Caller leaves Bob a voicemail  -> externnotify calls script which
> turns
> 
>> on his Cisco MWI.
> 
>  
> 
>> Bob checks Voicemail  ->  Bob deletes Voicemail  -> asterisk says that
> 
>> the voicemail was deleted, but doesn't run my script again to turn off
> 
>> the Cisco MWI.
> 
>  
> 
>> I would just like to know if there is any work around for this.
> 
>> OR.  Maybe Someone is working on adding this into the code
> 
>> so that it works...
> 
>  
> 
>> I'm running * 1.6.1-beta2
> 
>  
> 
> afaicr I read something which might be related in doc/smdi.txt.


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Yehavi Bourvine]

Thanks to everyone who replies so far!

We have Nortel PBX'es with a support contract from one of the local VARs
(Nortel does not give direct support here). In the last two weeks we had one
of our exchanges down for three half days; one was after a failure, and the
other two were when the technician came to fix remainders of the original
problem and just did a "5 seconds restart which won't even cut calls". Yeh,
the "5 seconds" took 6-7 hours... BTW, they still do not know what was the
original problem.

So, why won't we save the big bucks we pay them, hire two professionals (who
cost less) and support an open source code by ourselves? This way we depend
on ourselves only.

 Thanks, __Yehavi:


 2008/11/21 Grygoriy Dobrovolskyy <[EMAIL PROTECTED]>

>
>
> 2008/11/21 Yehavi Bourvine <[EMAIL PROTECTED]>
>
>>   Hello,
>>
>>   Our university has to upgrade soon its old Nortel PBX's which holds
>> around 10,000 extensions tied to 5 PBXes. Up to now we thought about
>> commercial solutions but now there is a window openning for open source
>> solution. However, I need examples to convince that this solution is
>> feasible, and preferably at other universities.
>>
>> Are there any pointers for such installations?
>>
>>Thanks! __Yehavi:
>>
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>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> Hello very interesting project you have, however asterisk is not a registry
> server, i suggest that you use opensips/opense/kamalio for your registrar,
> from where you dispatch to you asterisk servers, inside a good environment
> with a controlled network and nice tagged voip flow you could acheve a good
> results.
>
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Re: [asterisk-users] SVN - DIGIUM

[Grygoriy Dobrovolskyy]

server problem's

2008/11/21 Luis Morales <[EMAIL PROTECTED]>

> Does any know what happens with svn repository on svn.digium.com ?
>
> --
>
> -
> Luis Morales
> Consultor de Tecnologia
> Cel: +(58)416-4242091
>
> -
> "Empieza por hacer lo necesario, luego lo que es posible... y de
> pronto estarás haciendo lo imposible"
>
> Leonardo Da'Vinci
>
> -
>
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[asterisk-users] PSTN Gateway setup

[Valentin Bud]

Hello list,

 I recently bought a Linksys SPA400 as a PSTN gateway. The gateway is
connected to
an * server and i have 10 users using this setup. I do have some
problems in establishing
a call to an outside location (call that goes through the SPA400). The
first attempt doesn't
get through.

 I suspect the spa400 being the source of the problem. The Linksys
SPA400 has a lot of
params on the PSTN side. I don't know where can i find this
information. I have called the
provider but the front line doesn't have a clue about this technical
stuff and i can't get in
touch with a tech guy from there. My telecommunications / PSTN
knowledge is almost
0 (i tend to improve it).

 I'm talking about Line Impedance and stuff like that. I live in
Romania and i suppose there are
some global (or at least local / country) standards which state the
values that can be used.

 I have configured the Linksys to use Line 1 and using the last
firmware from them i have
chosen the zone to be European FSK in the Voice tab.

 My assumptions that SPA400 is the source of the problem came from the
fact that on my
test setup in my office it worked like a charm. When i deployed the
solution to the client
the problem arose. The only different thing between my office and the
clients' one is the
PSTN provider.

 For Romanian guys: my office connects to RTC and the clients' office
connects to UPC (through an Allied Telesys which provides media
conversion from fiber
to ethernet also pstn), just is case it matters and make sense.e

 Please give me a hint on any kind of documentation i can read, i'm
really lost with this
problem i have :(.

thank you and have a great day,
v

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Grygoriy Dobrovolskyy]

2008/11/21 Yehavi Bourvine <[EMAIL PROTECTED]>

> Hello,
>
>   Our university has to upgrade soon its old Nortel PBX's which holds
> around 10,000 extensions tied to 5 PBXes. Up to now we thought about
> commercial solutions but now there is a window openning for open source
> solution. However, I need examples to convince that this solution is
> feasible, and preferably at other universities.
>
> Are there any pointers for such installations?
>
>Thanks! __Yehavi:
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

Hello very interesting project you have, however asterisk is not a registry
server, i suggest that you use opensips/opense/kamalio for your registrar,
from where you dispatch to you asterisk servers, inside a good environment
with a controlled network and nice tagged voip flow you could acheve a good
results.
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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[EdPimentl]

Strongly suggest to consider a Freeswitch/OpenSER implementation instead.
Regarding purpose built and supported software.sometimes throwning
billions of CMM software development to a product does not guarantee a good
product... look at Micro$oft Vista.
E
http://Gpro.ws
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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Gordon Henderson]

On Fri, 21 Nov 2008, Al Baker wrote:

> Remember - You are going from a CARRIER GRADE purpose built piece of
> hardware with Software built under a rigid CMM with extensive
> "soak-testing" to software that has been developed under , shall we say,
> a somewhat less rigid and stringent methodology.
> You will be moving from an environment supported by hundreds of highly
> trained people, some with decades of TELCO experience
> to one where you support comes from a somewhat less seasoned group of
> individuals.
> 10,000 extensions ???
> On Asterisk ???
> You pays your money, you takes you chances.

I know what a few friends who work/study in the astrophysics department of 
a university half an hour up the road from me would rather have - their 
new "carrier grade" switch built under a rigid CMM, etc. fails about once 
a month right now. Recently it was because it was start of term and it 
couldn't handle the additional call-load. They used to forward me emails 
from their support department as a bit of a joke, but they've stopped 
doing it now as it's way beyond a joke.

They paid their money, took their chance with a full-commercial system and 
blew-it. I just wish I could get in there now, but it's too "political" a 
situation for an outsider to get anywhere.

Whats equally annoying is that their"Innovations Centre" (a sort of 
business "incubator" unit for graduates) has a very clever Asterisk system 
installed, but the main university seems oblivious to it all.

Gordon

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