date:20080226

[asterisk-users] duplicated voicemail messages

2008-02-26 Thread Craig Kowald

Hello,

It has happened to me twice now that duplicated voicemail messages are
automatically created, every minute.

I have been unable to reliably repeat it (so far), but the basic flow
seems to be:

1. a call comes in via my TDM400P (PSTN line)

2. the call is not answered and goes to voicemail

3. the caller does not really leave a message, just 10 seconds or so of
silence. At least, that is all I end up with.

4. Every minute from that point on, a new voicemail message is created.
All of the messages are 10 seconds of silence, so I assume they are just
duplicates of the original message.


The first time this happened, my mailbox was completely filled with
blank messages.

The second time, it just stopped after 25 minutes. In this case I ended
up with a CDR indicating that the call was answered and lasted for 25
minutes - although the final destination (dst column) of the call was
't' (which I assume means timeout, not that that makes any sense to me).


So, has anybody else ever had a situation where duplicate voicemail
messages are created ? And if so, what did you do about it ?


Regards,

Craig Kowald.



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Re: [asterisk-users] problem transferring calls some of the times

2008-02-26 Thread Ian


Hi Raúl

It would seem that we might have the same problem here. As I also get 
"No Answer" records in the cdr database for the calls that failed. I 
just checked against a tester I did yesterday that failed.


See the extract of the call from the CDR database, does it look anything 
like yours?


calldate 
 
	clid 
 
	src 
 
	dst 
 
	dcontext 
 
	channel 
 
	dstchannel 
 
	lastapp 
 
	lastdata 
 
	duration 
 
	billsec 
 
	disposition 
 
	amaflags 
 
	accountcode 
 
	uniqueid 
 
	userfield 
 

2008-02-26 14:11:59 	  	  	s 	incoming_calls 	Zap/4-1 
SIP/300-009218b0 	Dial 	SIP/300 	116 	106 	ANSWERED 	3 	  	  	 
2008-02-26 14:14:12 	"Luzaan lyn 1" <300> 	300 	301 	internal 
SIP/300-00835200 	SIP/301-00843750 	Dial 	SIP/301|30 	19 	9 
ANSWERED 	3 	  	  	 
2008-02-26 14:18:09 	301 	301 	s 	internal 	SIP/316-00919110 
 	  	  	4 	0 	NO ANSWER 	3 	  	  	 




Raúl Gómez C. said the following on 26-Feb-08 05:29 PM:

Ian,

I'm having *THE SAME PROBLEM* and I've noticed that when a transfer 
fail (only happens when receptionist dial an external number) the call 
is marker as "NO ANSWER" in the CDR, even when the call *HAS BEEN 
ANSWERED* by the other party (the callee). See my previous post below.



http://lists.digium.com/pipermail/asterisk-users/2008-February/206228.html

http://lists.digium.com/pipermail/asterisk-users/2008-February/206533.html


So I think Asterisk doesn't want to transfer calls that hasn't been 
answered, *OR* maybe are the phone itself (since I have the EXACT 
phones you have GXP-2000), that is causing the problem.
My testing and log whatching got me believing its a problem with 
"Zombie" calls becuase of reregistering on the Grandstream phones. Dont 
know if you have noticed it as well, almost always happens at the top of 
the hour.


BTW: I have the same Asterisk, Zaptel and Libpri versions as you have.
I downgraded from 1.4.8 to 1.4.7.1 in favour of being able to dial using 
DTMF.


Please check your CDR and look if the calls that has failed to 
transfer are marked as "NO ANSWER".

You are spot on here.



Thanks, I hope we can solve this anytime soon...
So do I, I am glad that there is someone else with this problem, I think 
we can help each other in this matter.


Regards
Ian


-

[asterisk-users] Call recording problems from queue

2008-02-26 Thread Scott Gifford

Hello,

I'm trying to set up call recording for a queue.  Right now the
recording appears to work correctly, but when I call and chatter for a
minute or so, at the end of the call I end up with a very small file
(less than 100 bytes), which contains about .06 seconds of silence.
If I talk for another minute, this file will get up to 200 bytes or
so.

In my queue configuration, I have:

[testq]
monitor-format = gsm
monitor-type = MixMonitor
...

I can see what looks like MixMonitor starting and stopping at the right
time:

-- IAX2/sgifford-3 answered Zap/1-1
  == Begin MixMonitor Recording Zap/1-1
-- Hungup 'IAX2/sgifford-3'
  == Spawn extension (incoming, 3772, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
  == End MixMonitor Recording Zap/1-1

I have tried turning debugging up very high (like 50) and I still
don't see any clues.

I'm using Asterisk 1.4.18 built from source.  The incoming lines are
Zap on a Sangoma card.  The queue members are IAX clients.  The calls
are being sent as GSM.

Does anybody have an idea what could be going wrong, or where to look
to debug this problem?

Thanks!

Scott.

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Re: [asterisk-users] [URGENT] Zap channels fail to load

2008-02-26 Thread Louwrens Benadé

Well, on an E1 PRI config your D-channel is indeed assigned to channel 16,
the center channel. On a T1, your data channel is on channel 24, the last
channel.

Did you restore your zaptel config from samples or another source?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andres Jimenez
Sent: 26 February 2008 06:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [URGENT] Zap channels fail to load

On Tue, Feb 26, 2008 at 1:35 PM, Louwrens Benadé <[EMAIL PROTECTED]>
wrote:
> Why does this look suspiciously like a T1 line? Are you sure this is a
>  fractional E1?

My provider names the line a PRA, but this is understood anywhere as a
PRI (no fractional).

>From the Asterisk configuration point of view, is there any other
difference between the fractional and full PRI apart from the number
of channels?

Do you guys know of any particular setting our provider (Eircom,
Ireland) could require?

I have a problem with DTMF when the PRI is the one carrying the call:
"1s" and "2s" are not transmitted. If the call is internal or carryed
by and IAX trunk + SIP it works nicely.

Regards,

-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
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Re: [asterisk-users] beta4: outgoing call causes Red Alarm on TDM400P

2008-02-26 Thread sean darcy

sean darcy wrote:
> Tzafrir Cohen wrote:
>> On Sun, Feb 24, 2008 at 05:38:43PM -0700, James Finstrom wrote:
>>> -BEGIN PGP SIGNED MESSAGE-
>>> Hash: SHA1
>>>
>>>
>>>
>>> Sean,
>>>
>>> I believe the alarm is generated by the bits flipping .
>>> In "kewl"  is hangup so every time you hang-up you could
>>> potentially alarm. 
>> That is: if the FXS uses KS, then upon hangup it will deny power for a
>> while. The FXO on the other side identifies this as if somebody
>> disconnected the wire, and is temporarily in alarm.
>>
>>> I don't know what the timer delay is but I think
>>> anything over a second would be safe otherwise you will see red alarms
>>> but they will probably be more of an annoyance than a serious issue.
>> Unless you actually want to keep the line open even upon recieving the
>> power denial. I can't think of a practical use for this.
>>
> 
> I swapped to a different power connector for the card. Rebooted.
> 
> Still the same problem on the very first call after reboot.
> 
> Beginning to think this is a beta4 issue.
> 
Not beta4, but zaptel-1.4.8. zaptel svn 3883 solved it.

sean


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Re: [asterisk-users] one CDR instead of multiple CDR

2008-02-26 Thread Dovid B


- Original Message - 
From: "Atis Lezdins" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Tuesday, February 05, 2008 4:24 PM
Subject: Re: [asterisk-users] one CDR instead of multiple CDR


> On 2/5/08, Arjan Kroon | Mobillion <[EMAIL PROTECTED]> wrote:
>> This is a part of our programma.
>>
>> [begin]
>> exten => s,1, h324m_gw([EMAIL PROTECTED])
>>
>> [video]
>> exten => s,1,h324m_gw_answer()
>> exten => s,2,Wait(3)
>> exten => s,3,Goto(intro,s,1)
>>
>> [intro]
>> exten => s,1,mp4play(intro.3gp)
>> exten => #,n,Goto(einde,s,1)
>>
>> [einde]
>> exten => s,n, Hangup()
>
> Seems like a side-effect of using local channels. You could try adding
> NoCDR() in context video, and see if that helps, and you still get
> valid call durations. Or as alternative - add NoCDR in context begin,
> as it completes almost immediately. However i don't see where third
> channel is raised.. Could you provide debug logs of affected call from
> /var/log/asterisk/full (enabling "full" in logger.conf).
>
> Regards,
> Atis
>
>
>>
>>
>> When I use this dialplan and during the intro.3gp I press the #-key the
>> call will be ended.
>> But I got three different CDR's.
>>
>> Does anybody know how I can use one CDR instead of 3 different CDR's
>>
>> Kind Regards,
>>
>>
>> Arjan Kroon

If he uses NoCDR() and the person calling in hangs up when the call is at 
that context then no recard of the call will be logged ? 



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Re: [asterisk-users] cannot dial out with latest zaptel and kernel 2.6.24

2008-02-26 Thread John covici

I updated zaptel and I can dial out, but when someone calls in it
won't hangup unless my extension hangs up, which was not true before.
This is a better state than before, thanks much for fixing so far.

on Tuesday 02/26/2008 Shaun Ruffell([EMAIL PROTECTED]) wrote
 > John Covici wrote:
 > > Hi.  I am using asterisk 1.4 (latest as of today) and zaptel 1.4
 > > (latest as of today) and I cannot dial out using my 400P card with one
 > > fxs module and one fxo module.  I am using kernel 2.6.24 and get the
 > > following log entries:
 > > [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Executing [EMAIL 
 > > PROTECTED]:23] Dial("Zap/1-1", "ZAP/4/www411|300|wW") in new stack
 > > [Feb 25 17:28:13] DEBUG[25071] chan_zap.c: Dialing 'www411'
 > > [Feb 25 17:28:13] DEBUG[25071] chan_zap.c: Deferring dialing...
 > > [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Called 4/www411
 > > [Feb 25 17:28:14] WARNING[25071] chan_zap.c: Detected alarm on channel 4: 
 > > No Alarm
 > > [Feb 25 17:28:14] VERBOSE[25071] logger.c: -- Hungup 'Zap/4-1'
 > > [Feb 25 17:28:14] VERBOSE[25071] logger.c:   == Everyone is busy/congested 
 > > at this time (1:0/0/1)
 > > 
 > > Any assistance on this would be appreciated.
 > > 
 > 
 > It looks like this might have been a combination of zaptel generating 
 > battery alarms which asterisk 1.4 didn't recognize.
 > 
 > Could you try updating just zaptel and see if you still see the alarm?

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]

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Re: [asterisk-users] Configuring modem pools in Asterisk [WAS: Connecting a Rolm CBX to Asterisk via T1?]

2008-02-26 Thread Craig Guy

It should look more like this:

exten => fax,1,Dial(IAX2/iaxmodem1/${NumberCalled}|20)
exten => fax,n,Dial(IAX2/iaxmodem2/${NumberCalled}|20)
exten => fax,n,Dial(IAX2/iaxmodem3/${NumberCalled}|20)
exten => fax,n,Dial(IAX2/iaxmodem4/${NumberCalled}|20)
exten => fax,n,Busy()

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua Kinard
Sent: Wednesday, 27 February 2008 7:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring modem pools in Asterisk [WAS:
Connecting a Rolm CBX to Asterisk via T1?]


Okay, T1 card issue sorted out.  New Lesson: Stay Away from TigerJet chips.

Next up, modem pool -- I wanted to know if the below config looked anywhere
near half-sane for defining in asterisk what is essentially a small pool of
four waiting modems that will handle faxes if another modem is busy:

exten => _X.,1,Dial(IAX2/iaxmodem0/${EXTEN})
exten => _X.,2,Busy
exten => _X.,3,Hangup

exten => _X.,4,Dial(IAX2/iaxmodem1/${EXTEN})
exten => _X.,5,Busy
exten => _X.,6,Hangup

exten => _X.,7,Dial(IAX2/iaxmodem2/${EXTEN})
exten => _X.,8,Busy
exten => _X.,9,Hangup

exten => _X.,10,Dial(IAX2/iaxmodem3/${EXTEN})
exten => _X.,11,Busy
exten => _X.,12,Hangup

This seemed logical, but redundant.  I've seen the usage of macro's to
condense stuff like that, but I wasn't sure how to have it auto-determine
which modem to use (i.e., iaxmodem0 through iaxmodem3).  In my mind, I'm
thinking of this in the form of a for loop:

for each modem in iaxmodem0..iaxmodem3
is it busy?
Yes: Continue
No:  Answer
done
done

Is something like that representable in asterisk-speak?


Also pondering ahead for working on outbound faxing, I'm assuming a
[fax-out] context would be somewhat similar as the above, just a different
set of iaxmodems (4-7)?


Thanks!,

--jkinard

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Re: [asterisk-users] cannot dial out with latest zaptel and kernel 2.6.24

2008-02-26 Thread Shaun Ruffell

John Covici wrote:
> Hi.  I am using asterisk 1.4 (latest as of today) and zaptel 1.4
> (latest as of today) and I cannot dial out using my 400P card with one
> fxs module and one fxo module.  I am using kernel 2.6.24 and get the
> following log entries:
> [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Executing [EMAIL 
> PROTECTED]:23] Dial("Zap/1-1", "ZAP/4/www411|300|wW") in new stack
> [Feb 25 17:28:13] DEBUG[25071] chan_zap.c: Dialing 'www411'
> [Feb 25 17:28:13] DEBUG[25071] chan_zap.c: Deferring dialing...
> [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Called 4/www411
> [Feb 25 17:28:14] WARNING[25071] chan_zap.c: Detected alarm on channel 4: No 
> Alarm
> [Feb 25 17:28:14] VERBOSE[25071] logger.c: -- Hungup 'Zap/4-1'
> [Feb 25 17:28:14] VERBOSE[25071] logger.c:   == Everyone is busy/congested at 
> this time (1:0/0/1)
> 
> Any assistance on this would be appreciated.
> 

It looks like this might have been a combination of zaptel generating 
battery alarms which asterisk 1.4 didn't recognize.

Could you try updating just zaptel and see if you still see the alarm?


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Re: [asterisk-users] TE120P echo cancellation problem

2008-02-26 Thread arkda

Just a heads up, the echo cancellation problem disappeared with Asterisk
1.4.15, zaptel 1.4.8, and libpri 1.4.3.

Still having other problems with the TE120P, but all OT from echo
cancellation.

On Mon, Feb 25, 2008 at 7:45 PM, arkda <[EMAIL PROTECTED]> wrote:

> Sorry, 1.4. Keep forgetting 1.2 is still around. These were built with:
>
> svn co http://svn.digium.com/svn/libpri/branches/1.4/ libpri
> svn co http://svn.digium.com/svn/zaptel/branches/1.4/ zaptel
> svn co http://svn.digium.com/svn/asterisk/branches/1.4/ asterisk
>
> There has never been another version of zaptel, Asterisk, or libpri on
> this server, it was built for testing this TE120P before we purchased more
> and rolled them out to customers.
>
> I'm am however considering rebuilding it with Asterisk .15, zaptel 1.4.8,
> and libpri 1.4.3 since we've had Digium G729 codec issues with anything
> newer than .15 on Asterisk.
>
>
> On Mon, Feb 25, 2008 at 7:23 PM, Kevin P. Fleming <[EMAIL PROTECTED]>
> wrote:
>
> > arkda wrote:
> >
> > > Asterisk revision 104093, zaptel revision 3849, libpri revision 529
> > all
> > > from svn.
> >
> > Revisions of what branches? We need to know the URL you checked out for
> > each of these, not just the revision number.
> >
> > Also, have you confirmed in your kernel message log that the version of
> > Zaptel running in memory is the one you downloaded and compiled? Have
> > you tried removing all previously built Zaptel modules from
> > /lib/modules/*/* and reinstalling it?
> >
> > --
> > Kevin P. Fleming
> > Director of Software Technologies
> > Digium, Inc. - "The Genuine Asterisk Experience" (TM)
> >
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>
>
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Re: [asterisk-users] How do I tell if T.38 was used?

2008-02-26 Thread arkda

You can interrogate the SIP information for some of this using the SIP debug
command on the CLI along with the udptl debug command. It's not perfect but
it works for what you're looking for.

On Tue, Feb 26, 2008 at 3:21 PM, Robert Moskowitz <[EMAIL PROTECTED]>
wrote:

> I am running Trixbox 2.4 which has Asterisk 1.4.18-1
>
> I have kind of followed:
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38
>
> I added to sip_general_custom.conf
>
> ;NEEDED!!!
> t38pt_udptl = yes
>
> I did not add this to the actual SIP extension, as I assumed this being
> general it applies to all sip extensions, and doing a sip show peer ext#
> did indeed come up with t38pt_udptl = yes
>
> The fax is attached to a Grandstream 488, so I set it for fax mode: T.38
>
> I did leave DTMF as inband (can't find any docs on what to use for this).
> my rx_fax works just fine, but it did for fax pass-through.
>
> So how do I determine if T.38 was negotiated?
>
>
>
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Re: [asterisk-users] How can I call cheap to UK cell phones

2008-02-26 Thread Zeeshan Zakaria

If I am originate from Canada, how can I benefit from these cheap rates?

On Tue, Feb 26, 2008 at 7:20 PM, Steve Kennedy <[EMAIL PROTECTED]>
wrote:

> On Tue, Feb 26, 2008 at 01:38:31PM -0600, Erik Anderson wrote:
>
> > On Tue, Feb 26, 2008 at 1:19 PM, Zeeshan Zakaria <[EMAIL PROTECTED]>
> wrote:
> > > Greetings,
> > > How can I call cheap to UK cell phones. I am located in Toronto,
> Canada, but
> > > need to call UK cell phones both from Toronto and London.
> > I'd guess you could get an account with one of these providers:
> >
> http://voip-info.org/wiki/view/VOIP+Service+Providers+Business+Europe#UnitedKingdom
>
> The termination rates are set by Ofcom  around 6-7p
> per minute dependent on network.
>
> BT offer blended rates (i.e. same rates for mobile/landline/etc), but
> you have to originate from outside UK.
>
> Steve
>
> --
> NetTek Ltd  UK mob +44-(0)7775 755503
> UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
> Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
> Euro Tech News Blog http://eurotechnews.blogspot.com
>
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-- 
Zeeshan A Zakaria
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Re: [asterisk-users] Anybody installed Asterisk in a Virtuozzo VPS system???

2008-02-26 Thread Daniel Pittman

Alan <[EMAIL PROTECTED]> writes:

> I have a small VPS server in www.eapps.com and im doing some research
> in order to install Asterisk in that server..
>
> Does anybody has installed Asterisk in a Virtuozzo VPS System??

I have done so, with success, for a SIP-only installation.  Well, into
OpenVZ which is the open source, free version of the same.

The only specific requirement was to install the dummy zaptel timer into
the hardware kernel, add the device nodes in the VE, then use the VZ
tools to allow the VE access to those devices.

Without that there was no timer (dummy, in my case) which resulted in
drifting send times for SIP connections.

Regards,
Daniel

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Re: [asterisk-users] How can I call cheap to UK cell phones

2008-02-26 Thread Steve Kennedy

On Tue, Feb 26, 2008 at 01:38:31PM -0600, Erik Anderson wrote:

> On Tue, Feb 26, 2008 at 1:19 PM, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:
> > Greetings,
> > How can I call cheap to UK cell phones. I am located in Toronto, Canada, but
> > need to call UK cell phones both from Toronto and London.
> I'd guess you could get an account with one of these providers:
> http://voip-info.org/wiki/view/VOIP+Service+Providers+Business+Europe#UnitedKingdom

The termination rates are set by Ofcom  around 6-7p
per minute dependent on network.

BT offer blended rates (i.e. same rates for mobile/landline/etc), but
you have to originate from outside UK.

Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com

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Re: [asterisk-users] chan_ss7 0.10

2008-02-26 Thread Andrew Furey

On 27/02/2008, Joel Solanki <[EMAIL PROTECTED]> wrote:
> I tried 3 times to send this message. It goes out but i dont recieve mail
> sent on asterisk-users@lists.digium.com but when someone replies to that
> email i recieve the email like you did.
> I thought mails were not going to mailling list to tried 3 times.  But it is
> strange. As far as i know if i sent mail to [EMAIL PROTECTED]
> then even i should also recieve my email right ?

I don't know whether the list server is refraining from sending it to
yourself, or if (more likely) Gmail is deciding not to show it since
you already know what it says... but yes, it happens for me too (as
well as on another mailing list I'm on). Standard Gmail behaviour, at
least.

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett

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Re: [asterisk-users] Had it with Dell Garbage

2008-02-26 Thread Steve Totaro

Sangoma cards work a treat in a HP DL380 or 320/260 for that matter.
I just like having two power supplies and hot swap RAID 5 plus a few
extra slots.

Thanks,
Steve Totaro

On Tue, Feb 26, 2008 at 5:51 PM, Joshua Kinard <[EMAIL PROTECTED]> wrote:
>
>
> Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very
> likely, 380's as well).  I just learned this the hard way.
>
> --J
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Norman Franke
> Sent: Tuesday, February 26, 2008 5:27 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Had it with Dell Garbage
>
>
>
>
> On Feb 26, 2008, at 4:13 PM, [EMAIL PROTECTED] wrote:
>
>
>
> On Tue, Feb 26, 2008 at 3:10 PM, Matt <[EMAIL PROTECTED]> wrote:
>
>
> I've had it with Dell server garbage.They seem to change RAID
>
> controllers as much as I change socks, and then the controllers don't work
>
> with Linux, unless you load a new driver.They sell servers with a PCI-e
>
> slot in them, but then you get it and find out the RAID controller is using
>
> the PCI-e slot!   Their sales folks are dumber than rocks, and they change
>
> them more often than I change underwear.
>
>  [end rant].
>
>
>
>
> Can anyone recommend an IBM or Gateway server that you have used with
>
> Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has
>
> room for one or two PCI-express interface cards?
>
>
>
>
>
>
>
> HP DL380 is my baby.
>
>
>
>
> Thanks,
>
> Steve Totaro
>
> Ditto. We've been using HPs for a while without problem. I'm currently using
> a DL380 (a recent quad processor one) and it screams.
>
>
> -Norman
>
>
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Re: [asterisk-users] beta4: outgoing call causes Red Alarm on TDM400P

2008-02-26 Thread sean darcy

Tzafrir Cohen wrote:
> On Sun, Feb 24, 2008 at 05:38:43PM -0700, James Finstrom wrote:
>> -BEGIN PGP SIGNED MESSAGE-
>> Hash: SHA1
>>
>>
>>
>> Sean,
>>
>> I believe the alarm is generated by the bits flipping .
>> In "kewl"  is hangup so every time you hang-up you could
>> potentially alarm. 
> 
> That is: if the FXS uses KS, then upon hangup it will deny power for a
> while. The FXO on the other side identifies this as if somebody
> disconnected the wire, and is temporarily in alarm.
> 
>> I don't know what the timer delay is but I think
>> anything over a second would be safe otherwise you will see red alarms
>> but they will probably be more of an annoyance than a serious issue.
> 
> Unless you actually want to keep the line open even upon recieving the
> power denial. I can't think of a practical use for this.
> 

I swapped to a different power connector for the card. Rebooted.

Still the same problem on the very first call after reboot.

Beginning to think this is a beta4 issue.

sean


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Re: [asterisk-users] chan_ss7 0.10

2008-02-26 Thread Joel Solanki

Hi,

I tried 3 times to send this message. It goes out but i dont recieve mail
sent on asterisk-users@lists.digium.com but when someone replies to that
email i recieve the email like you did.
I thought mails were not going to mailling list to tried 3 times.  But it is
strange. As far as i know if i sent mail to
[EMAIL PROTECTED] even i should also recieve my email
right ?

may be i can be wrong. Any i am happy that message has reached the
asterisk-users :) Hope to recieve some feedback soon

Regards,
Joel

On Wed, Feb 27, 2008 at 3:24 AM, Rob Hillis <[EMAIL PROTECTED]> wrote:

> Posting the same question three times at 12 hour intervals will not get
> you a faster reply.
>
> Especially not to an original email that was written in November of last
> year.
>
>
> Joel Solanki wrote:
> > Hi marek,
> >
> > Thanks for the update.
> > I have Sangoma A104D and wanted to use ss7 signalling. I came accross
> > chan_ss7 but found sifira is not in active development.  But is this
> > chan_ss7 stable and can be used in production server implementation.
> > We are going to have 2 to 3 boxes with ss7 signalling using sangoma
> > but I am
> > looking for open source ss7 implementation which is chan_ss7. so need to
> > know about stability and recommendation for using on production server.
> >
> >
> > Please provide your recommendation & suggestions.
> >
> >
> > Regards,
> > Joel
> >
>
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[asterisk-users] Anything like SipT38SwitchOver in Asterisk?

2008-02-26 Thread Robert Moskowitz

Is there something equivalent to SipT38SwitchOver in Asterisk (in 
callweaver)...

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Re: [asterisk-users] Had it with Dell Garbage

2008-02-26 Thread Joshua Kinard

Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very 
likely, 380's as well).  I just learned this the hard way.

--J

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Norman Franke
Sent: Tuesday, February 26, 2008 5:27 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Had it with Dell Garbage

On Feb 26, 2008, at 4:13 PM, [EMAIL PROTECTED] wrote:

On Tue, Feb 26, 2008 at 3:10 PM, Matt < [EMAIL PROTECTED]> wrote:

I've had it with Dell server garbage.They seem to change RAID

controllers as much as I change socks, and then the controllers don't work

with Linux, unless you load a new driver.They sell servers with a PCI-e

slot in them, but then you get it and find out the RAID controller is using

the PCI-e slot!   Their sales folks are dumber than rocks, and they change

them more often than I change underwear.

 [end rant].

Can anyone recommend an IBM or Gateway server that you have used with

Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has

room for one or two PCI-express interface cards?

HP DL380 is my baby.

Thanks,

Steve Totaro

Ditto. We've been using HPs for a while without problem. I'm currently using a 
DL380 (a recent quad processor one) and it screams. 

-Norman

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Re: [asterisk-users] Configuring modem pools in Asterisk [WAS: Connecting a Rolm CBX to Asterisk via T1?]

2008-02-26 Thread Joshua Kinard


Okay, T1 card issue sorted out.  New Lesson: Stay Away from TigerJet chips.

Next up, modem pool -- I wanted to know if the below config looked anywhere 
near half-sane for defining in asterisk what is essentially a small pool of 
four waiting modems that will handle faxes if another modem is busy:

exten => _X.,1,Dial(IAX2/iaxmodem0/${EXTEN})
exten => _X.,2,Busy
exten => _X.,3,Hangup

exten => _X.,4,Dial(IAX2/iaxmodem1/${EXTEN})
exten => _X.,5,Busy
exten => _X.,6,Hangup

exten => _X.,7,Dial(IAX2/iaxmodem2/${EXTEN})
exten => _X.,8,Busy
exten => _X.,9,Hangup

exten => _X.,10,Dial(IAX2/iaxmodem3/${EXTEN})
exten => _X.,11,Busy
exten => _X.,12,Hangup

This seemed logical, but redundant.  I've seen the usage of macro's to condense 
stuff like that, but I wasn't sure how to have it auto-determine which modem to 
use (i.e., iaxmodem0 through iaxmodem3).  In my mind, I'm thinking of this in 
the form of a for loop:

for each modem in iaxmodem0..iaxmodem3
is it busy?
Yes: Continue
No:  Answer
done
done

Is something like that representable in asterisk-speak?


Also pondering ahead for working on outbound faxing, I'm assuming a [fax-out] 
context would be somewhat similar as the above, just a different set of 
iaxmodems (4-7)?


Thanks!,

--jkinard

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[asterisk-users] How is reinvite triggered

2008-02-26 Thread Robert Moskowitz

Particularly WRT T.38 fax.

Supposedly, when fax tones are detected, Asterisk is to do a reinvite 
asking for T.38.

Here is what I am using in my dialplan:

[custom-fax1]
exten => s,1,Answer
exten => s,n,StopPlayTones
exten => s,n,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => s,n,rxfax(${FAXFILE})
exten => s,n,Hangup
exten => h,1,system(/var/lib/asterisk/bin/fax-process.pl --to 
${FAX_TO_EMAIL} --from ${FAX_RX_FROM} --subject "Fax from 
${URIENCODE(${CALLERID(number)})} ${URIENCODE(${CALLERID(name)})}" 
--attachment fax_${URIENCODE(${CALLERID(number)})}.pdf --type 
application/pdf --file ${FAXFILE});
exten => h,2,Hangup()



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Re: [asterisk-users] Had it with Dell Garbage

2008-02-26 Thread Norman Franke

On Feb 26, 2008, at 4:13 PM, [EMAIL PROTECTED]  
wrote:



On Tue, Feb 26, 2008 at 3:10 PM, Matt <[EMAIL PROTECTED]> wrote:

I've had it with Dell server garbage.They seem to change RAID
controllers as much as I change socks, and then the controllers  
don't work
with Linux, unless you load a new driver.They sell servers  
with a PCI-e
slot in them, but then you get it and find out the RAID controller  
is using
the PCI-e slot!   Their sales folks are dumber than rocks, and  
they change

them more often than I change underwear.
 [end rant].

Can anyone recommend an IBM or Gateway server that you have used with
Asterisk and are happy with, and which will support RAID-1 or  
RAID-5 and has

room for one or two PCI-express interface cards?



HP DL380 is my baby.

Thanks,
Steve Totaro


Ditto. We've been using HPs for a while without problem. I'm  
currently using a DL380 (a recent quad processor one) and it screams.


-Norman

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Re: [asterisk-users] Had it with Dell Garbage

2008-02-26 Thread Daniel Cole

We currently using the x service servers as well, never had any problems with 
them.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, 27 February 2008 7:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Had it with Dell Garbage

On Tue, Feb 26, 2008 at 3:20 PM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
>
> On Tue, Feb 26, 2008 at 3:10 PM, Matt <[EMAIL PROTECTED]> wrote:
>  > I've had it with Dell server garbage.They seem to change RAID
>  > controllers as much as I change socks, and then the controllers don't work
>  > with Linux, unless you load a new driver.They sell servers with a PCI-e
>  > slot in them, but then you get it and find out the RAID controller is using
>  > the PCI-e slot!   Their sales folks are dumber than rocks, and they change
>  > them more often than I change underwear.
>  >  [end rant].
>  >
>  > Can anyone recommend an IBM or Gateway server that you have used with
>  > Asterisk and are happy with, and which will support RAID-1 or RAID-5 and 
> has
>  > room for one or two PCI-express interface cards?
>  >
>
>  HP DL380 is my baby.
>
>  Thanks,
>  Steve Totaro
>

IBM X series are also great, I have deployed many, I just have a thing for HP.

Thanks,
Steve Totaro

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Re: [asterisk-users] Realtime Queue Status for Agents

2008-02-26 Thread Todd Houle

Do you mean that an agent on the phone doesn't know if there are  
callers in line behind the current one or not?  I had that question as  
well.  Perhaps not the best way, but I solved it by taking an extra  
phone and setting it up as a static agent in the closet.  Then I made  
a light on my phone notify me of a call on that closet phone.  This  
way, the agent could see the other phone 'ring' via a light on their  
phone and know a caller was waiting.

You could also do this via a web page that can monitor a queue.
http://sourceforge.net/project/showfiles.php?group_id=178880&release_id=454063

   Todd

On Feb 25, 2008, at 4:35 PM, Ken Leland III wrote:

> Hello,
>
> I am having a problem which stems from the fact that the Agents that
> handle my call queues are unaware if there are are people waiting in  
> the
> queue.  I have found the following configuration options but they are
> not very helpful.
>
> in queues.conf:
> announce=xxx
> The "announce = XXX" option in queues.conf makes Asterisk play the XXX
> announcement to the member of the queue who picks up the call,  This  
> is
> helpful but does not help if there is only one agent and the queue  
> fills
> up quickly *after* a call has been answered.
>
> periodic-announce
> this would be perfect except it is played to the caller instead of the
> agent.
>
> any ideas?
>

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Re: [asterisk-users] Cannot hear voice through SIP Phone from one side

2008-02-26 Thread Todd Houle

On Feb 5, 2008, at 3:32 PM, Sanjoy Rath wrote:

> I have a asterisk server. Two SIP Soft XLites are connected to the  
> server. I am able to make
> calls from one SIP Phones to the other SIP Phones and landlines  
> successfully. The SIP Soft Phone on th eother side can hear my voice  
> but I cannot hear their voice.
>
> They can call my local cell phone as well. Samething, they can hears  
> my voice, I cannot hear their voice.
>

You can read this:
http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html

In short, adding code similar to the following to your sip.conf should  
fix it.

[general]
externip = 99.230.47.247
localnet = 192.168.2.0/255.255.255.0

   Todd


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Re: [asterisk-users] chan_ss7 0.10

2008-02-26 Thread Rob Hillis

Posting the same question three times at 12 hour intervals will not get
you a faster reply.

Especially not to an original email that was written in November of last
year.


Joel Solanki wrote:
> Hi marek,
>
> Thanks for the update.
> I have Sangoma A104D and wanted to use ss7 signalling. I came accross
> chan_ss7 but found sifira is not in active development.  But is this
> chan_ss7 stable and can be used in production server implementation.
> We are going to have 2 to 3 boxes with ss7 signalling using sangoma
> but I am
> looking for open source ss7 implementation which is chan_ss7. so need to
> know about stability and recommendation for using on production server.
>
>
> Please provide your recommendation & suggestions.
>
>
> Regards,
> Joel
>

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Re: [asterisk-users] Sip trunk mystery

2008-02-26 Thread Dirk Enrique Seiffert

Hi Jared,

>
> Notice how the Contact Header and the SDP all have the IP address of
> 192.168.8.3?  If your firewall isn't masquerading (rewriting) those
> addresses as the SIP traffic goes through it, then the device on the
> other end is going to try to contact 192.168.8.3, and I'm guessing it's
> going to have a hard time doing that.  (This would also explain why
> you're seeing outbound traffic only in your tcpdump traces.)
>

My firewall is masquerading, anyhow I modified also the
externalip=mypublicIP . Now it looks like this:

Retransmitting #4 (NAT) to 190.144.151.212:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP mypublicIP:5060;branch=z9hG4bK01757b08;rport
From: "901" ;tag=as69ce5a7a
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Grandstream BT100 1.0.4.49
Max-Forwards: 70
Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch",
algorithm=MD5, uri="sip:[EMAIL PROTECTED]",
nonce="120406066426160405702174408208",
response="a6c72a47ea39c1200f2add823369ebce", opaque=""
Date: Tue, 26 Feb 2008 21:20:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

What else might might be wrong here?


Thanks

Enrique
Cartagena - Colombia
http://www.sipcolombia.com


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Re: [asterisk-users] Had it with Dell Garbage

2008-02-26 Thread Conrad Wood

I second Sun and supermicro.

Sun was really cool on the management facilities, the linux
compatibility and the speed was nice too.

Supermicro (opteron series) always amazes me how fast they are. They
really *feel* fast ;)

Only ever used support on supermicro and it was excellent. My box froze
and after I send them the opteron built-in exception log they identified
a problem with one of the DIMMs, told me which one and sent me a
replacement. No fuss with the old one either, I threw it away.

Conrad



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Re: [asterisk-users] Had it with Dell Garbage

2008-02-26 Thread Nick Seraphin

I haven't bought from them recently, but I also have bought many servers
and desktop systems from J & N.  I have at least 3 servers they built that
are over 8 years old and still running in production.  I've bought like 8
servers, and a half-dozen desktop systems from them since around 1996 or
1997.  Again, I can't speak for anything recently, but they have been a
legitimate company, reliable, and honest to deal with.  I used to deal
directly with the owner (Jerry Jacobsen) -- now they have a lot more
employees so you're likely better off just dealing with whoever answers
the phone because they're a lot bigger now.  They're definitely not a
fly-by-night company... it should be safe for you to give them a try if
you like what they offer.

Lately I buy Supermicro 1U servers and add my own cpu/ram/hd.  I had a
batch of bad motherboards, but other than that I like the quality so far.
One word of warning... their warranty is stupid.  The warranty starts the
day they sell the product to the reseller/warehouse company... NOT to the
end user.  So if you buy from Tech Data or Newegg and it sits on the shelf
in their warehouse for a month, you only get an 11 month warranty because
the clock starts ticking when it leaves Supermicro's plant.  They consider
Tech Data or Newegg to be their customer, not you.

If it sits in the warehouse for a year, then you get NO WARRANTY.

Based on my dealings with J&N however, I think they would honor the full
warranty even if Supermicro doesn't, if you buy it from J&N.  They're
definitely not perfect (in the early days they were slow to ship, but that
seems to be better now...  and last time I requested a custom quote they
never got back to me) but they won't cheat you and they stand behind what
they sell.  I'd deal with them again if I felt they had what I needed at
the right price (or even reasonably close).

-- Nick

On Tue, 26 Feb 2008, John covici wrote:

> I had a server built for me by J and N Computer Services
> http://www.jncs.com which is using a Super Micro c2sbe MB which I
> think has what you need plus 4 PCI-32 slots!  Its a nice MB and I have
> an e8400 cpu in it.
> 
> 
> 
> on Tuesday 02/26/2008 Matt([EMAIL PROTECTED]) wrote
>  > I've had it with Dell server garbage.They seem to change RAID
>  > controllers as much as I change socks, and then the controllers don't work
>  > with Linux, unless you load a new driver.They sell servers with a PCI-e
>  > slot in them, but then you get it and find out the RAID controller is using
>  > the PCI-e slot!   Their sales folks are dumber than rocks, and they change
>  > them more often than I change underwear.
>  > [end rant].
>  > 
>  > Can anyone recommend an IBM or Gateway server that you have used with
>  > Asterisk and are happy with, and which will support RAID-1 or RAID-5 and 
> has
>  > room for one or two PCI-express interface cards?
>  > I've had it with Dell server garbage.    They seem to 
> change RAID controllers as much as I change socks, and then the controllers 
> don't work with Linux, unless you load a new driver.    
> They sell servers with a PCI-e slot in them, but then you get it and find out 
> the RAID controller is using the PCI-e slot!   Their sales folks 
> are dumber than rocks, and they change them more often than I change 
> underwear.
>  > [end rant].Can anyone recommend an IBM or Gateway server that you 
> have used with Asterisk and are happy with, and which will support RAID-1 or 
> RAID-5 and has room for one or two PCI-express interface cards?
>  > ___
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>  > 
>  > asterisk-users mailing list
>  > To UNSUBSCRIBE or update options visit:
>  >http://lists.digium.com/mailman/listinfo/asterisk-users
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>  John Covici
>  [EMAIL PROTECTED]
> 
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Re: [asterisk-users] Attatch monitor recording to a voicemail

2008-02-26 Thread Ben Willcox

Jared Smith wrote:

> No, unfortunately this was done under NDA, but the general gist goes like 
> this:

As it happens, I deployed my solution to this on our live PBX today, which I
wrote with some help from another asterisk-users user. Here is what I
came up with:

Firstly, in features.conf I don't use the normal automon function in the
featuremap, but an applicationmap:

[applicationmap]
recordtovm =>*1,self,Macro,recordtovm


Then in extensions.conf we have the following additions:

[globals]
DYNAMIC_FEATURES=>recordtovm

[macro-recordtovm]
exten => 
s,1,Set(MONITOR_FILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)})
exten => s,n,Set(DYNAMIC_FEATURES=recordtovm)
exten => s,n,MixMonitor(${MONITOR_FILENAME}.wav,b,/etc/asterisk/recordtovm.pl 
${CALLERID(num)} ${MONITOR_FILENAME}.wav)

And finally the perl script recordtovm.pl in /etc/asterisk/ is as follows:

#!/usr/bin/perl -w
#
use strict;

my $monitordir="/var/spool/asterisk/monitor/";
my $vmdir="/var/spool/asterisk/voicemail/default/";
my $vmfolder="INBOX";
my $vmbox=$ARGV[0];
my $vmpath=$vmdir."$vmbox/"."$vmfolder";
my $monitorfilename=$ARGV[1];

opendir (DIR, $vmpath);
my @files = grep(/\.txt$/,readdir(DIR));
closedir(DIR);
my @sortedfiles = sort {$b cmp $a} @files;
my $vmid;
if ($sortedfiles[0] =~ /^(msg)(\d\d\d\d)(.txt)/)
{
$vmid=$2;
$vmid++;
}
else
{
$vmid="";
};

open VMFILE,"> $vmpath/msg$vmid.txt";
print VMFILE ";\n";
print VMFILE "; Message Information file\n";
print VMFILE ";\n";
print VMFILE "[message]\n";
print VMFILE "origmailbox=$vmbox\n";
print VMFILE "context=\n";
print VMFILE "macrocontext=\n";
print VMFILE "exten=s\n";
print VMFILE "priority=\n";
print VMFILE "callerchan=\n";
print VMFILE "callerid=\n";
print VMFILE "origdate=\n";
print VMFILE "origtime=\n";
print VMFILE "category=\n";
print VMFILE "duration=\n";
close VMFILE;

if ($ARGV[1])
{
system("mv $monitordir"."$monitorfilename $vmpath/msg$vmid.wav");
};


Seems to work pretty well, we have the Record button on our SNOM phones mapped 
to DTMF *1,
so its a single press to start recording. The perl script doesn't populate the 
origdate and
origtime fields at the moment so you'll need to add this if you want the time 
and date saving
with the message.

Hope this helps,

Ben





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Re: [asterisk-users] Had it with Dell Garbage

2008-02-26 Thread Mike Trest - Personal

Steve,
I have fielded several hundred Asterisk and related VoIP boxes.
I buy SuperMicro 1-U units mostly.  I have also used their larger
units with RAID and a full load of ULTRA SCSI (for MySql application).

I like these because, after bad experience with DELL/COMPAQ/HP/IBM
compatibility issue, the supermicro systems always load and work with
all of the Fedora kernels will just with their RAID controllers.

..mike..


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Re: [asterisk-users] Had it with Dell Garbage

2008-02-26 Thread Matt

2.6 CentOS 4

>
>
> I can't speak to the PCIe issue, but I've never in my life had
> compatibility issues with the Dell RAID controllers.  What kernel are
> you on?
>
> > Can anyone recommend an IBM or Gateway server that you have used with
> > Asterisk and are happy with, and which will support RAID-1 or RAID-5 and
> has
> > room for one or two PCI-express interface cards?
>
> Gateway server?  Ew.
>
> Have you looked into the new Sun servers?  I've been researching them
> lately, and they have some compelling offerrings.  They also offer
> full support for linux as well...
>
> -erik
>
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Re: [asterisk-users] SippySkype

2008-02-26 Thread Steven

I have gotten this to work and am glad to see an open source solution.

I have tested it as a gateway to our asterisk IVR.

I will need to have three to five skype instances running to keep away from a 
potential busy signal.

Does anyone know of a good solution for this?

Does anyone know of a good virtualization option that will support sound well 
enough for this?




-- 
-- 
Steven

http://www.connectech.org/



"James Finstrom" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> found this today, I am not a skype user but have read on chan_skype
> and don't like aspects of how it is implemented. My thoughts on it are
> only theoretical as I haven't used it I just cringe at adding X to a
> server. Anyhow there is a new project called sippyskype that appears
> to do a similar sort of thing with a couple differences.
>
> 1. Its FREE (as in beer)
> 2. It runs as a sip proxy so you can load it on a desktop or if you
> happen to have a windoze box you can put it there then asterisk can
> make a sip connection to it and your off... or on
>
> Again I am not a skype user so this may not be as cool as it sounds
> but if you are you may consider it.
>
> http://www.mhspot.com/mhspot/sippyskype.htm
>
> - --
> James Finstrom
> Rhino Equipment Corp.
> All Rhino products are made in America, Come with a Money Back gurantee
> and have a 5 Year warranty. Quality and Toughness built in!!
> Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826
> IP: asterisk.rhinoequipment.com ~ FWD: 633686
>
> THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY
> MATERIAL and is thus for use only by the intended recipient. If you
> received
> this in error, please contact the sender and delete the email and its
> attachments from all computers.
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.6 (GNU/Linux)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFHvESIdloC7YyaIOoRAr7qAJsGJIJvlmUGlo7WfebZVpzynDZVSACfQgwo
> YH747F21Mma5Ye8RhEsEVvA=
> =G7sV
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>
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Re: [asterisk-users] Had it with Dell Garbage

2008-02-26 Thread Erik Anderson

On Tue, Feb 26, 2008 at 2:10 PM, Matt <[EMAIL PROTECTED]> wrote:
> I've had it with Dell server garbage.They seem to change RAID
> controllers as much as I change socks, and then the controllers don't work
> with Linux, unless you load a new driver.They sell servers with a PCI-e
> slot in them, but then you get it and find out the RAID controller is using
> the PCI-e slot!   Their sales folks are dumber than rocks, and they change
> them more often than I change underwear.
>  [end rant].

Ouch!  :-)

I can't speak to the PCIe issue, but I've never in my life had
compatibility issues with the Dell RAID controllers.  What kernel are
you on?

> Can anyone recommend an IBM or Gateway server that you have used with
> Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has
> room for one or two PCI-express interface cards?

Gateway server?  Ew.

Have you looked into the new Sun servers?  I've been researching them
lately, and they have some compelling offerrings.  They also offer
full support for linux as well...

-erik

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[asterisk-users] Had it with Dell Garbage

2008-02-26 Thread John covici

I had a server built for me by J and N Computer Services
http://www.jncs.com which is using a Super Micro c2sbe MB which I
think has what you need plus 4 PCI-32 slots!  Its a nice MB and I have
an e8400 cpu in it.



on Tuesday 02/26/2008 Matt([EMAIL PROTECTED]) wrote
 > I've had it with Dell server garbage.They seem to change RAID
 > controllers as much as I change socks, and then the controllers don't work
 > with Linux, unless you load a new driver.They sell servers with a PCI-e
 > slot in them, but then you get it and find out the RAID controller is using
 > the PCI-e slot!   Their sales folks are dumber than rocks, and they change
 > them more often than I change underwear.
 > [end rant].
 > 
 > Can anyone recommend an IBM or Gateway server that you have used with
 > Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has
 > room for one or two PCI-express interface cards?
 > I've had it with Dell server garbage.    They seem to 
 > change RAID controllers as much as I change socks, and then the controllers 
 > don't work with Linux, unless you load a new driver.    
 > They sell servers with a PCI-e slot in them, but then you get it and find 
 > out the RAID controller is using the PCI-e slot!   Their sales 
 > folks are dumber than rocks, and they change them more often than I change 
 > underwear.
 > [end rant].Can anyone recommend an IBM or Gateway server that you 
 > have used with Asterisk and are happy with, and which will support RAID-1 or 
 > RAID-5 and has room for one or two PCI-express interface cards?
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 > asterisk-users mailing list
 > To UNSUBSCRIBE or update options visit:
 >http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]

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Re: [asterisk-users] Had it with Dell Garbage

2008-02-26 Thread Steve Totaro

On Tue, Feb 26, 2008 at 3:20 PM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
>
> On Tue, Feb 26, 2008 at 3:10 PM, Matt <[EMAIL PROTECTED]> wrote:
>  > I've had it with Dell server garbage.They seem to change RAID
>  > controllers as much as I change socks, and then the controllers don't work
>  > with Linux, unless you load a new driver.They sell servers with a PCI-e
>  > slot in them, but then you get it and find out the RAID controller is using
>  > the PCI-e slot!   Their sales folks are dumber than rocks, and they change
>  > them more often than I change underwear.
>  >  [end rant].
>  >
>  > Can anyone recommend an IBM or Gateway server that you have used with
>  > Asterisk and are happy with, and which will support RAID-1 or RAID-5 and 
> has
>  > room for one or two PCI-express interface cards?
>  >
>
>  HP DL380 is my baby.
>
>  Thanks,
>  Steve Totaro
>

IBM X series are also great, I have deployed many, I just have a thing for HP.

Thanks,
Steve Totaro

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Re: [asterisk-users] Sip trunk mystery

2008-02-26 Thread Jared Smith

On Tue, 2008-02-26 at 12:31 -0500, Dirk Enrique Seiffert wrote: 
> I aquired an account with a reseller net-voz.com: I did some testing with
> the account directly from a Snom300 phone - works without a problem,
> (behind the nat) I spent hours testing and adjusting the trunk
> configuration for net-voz, maybe sombody on the list can give a helpful hint:

I'll take a stab at it.

> First of all: Registry works!

Registering to another host doesn't mean anything when it comes to
sending them a call.  Registration only tells them your IP address and
port so that they can send calls *to you*.


> On the astersik CLI the logs show:
> 
> Audio is at 192.168.8.3 port 14800
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 190.144.151.212:5060:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport
> From: "901" ;tag=as3c6dfee5
> To: 
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 103 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch",
> algorithm=MD5, uri="sip:[EMAIL PROTECTED]",
> nonce="12040419552605702055508208",
> response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque=""
> Date: Tue, 26 Feb 2008 16:09:09 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 260
> 
> v=0
> o=root 2381 2382 IN IP4 192.168.8.3
> s=session
> c=IN IP4 192.168.8.3
> t=0 0
> m=audio 14800 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> ---

Notice how the Contact Header and the SDP all have the IP address of
192.168.8.3?  If your firewall isn't masquerading (rewriting) those
addresses as the SIP traffic goes through it, then the device on the
other end is going to try to contact 192.168.8.3, and I'm guessing it's
going to have a hard time doing that.  (This would also explain why
you're seeing outbound traffic only in your tcpdump traces.)

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[asterisk-users] How do I tell if T.38 was used?

2008-02-26 Thread Robert Moskowitz

I am running Trixbox 2.4 which has Asterisk 1.4.18-1

I have kind of followed: 
http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38

I added to sip_general_custom.conf

;NEEDED!!!
t38pt_udptl = yes

I did not add this to the actual SIP extension, as I assumed this being 
general it applies to all sip extensions, and doing a sip show peer ext# 
did indeed come up with t38pt_udptl = yes

The fax is attached to a Grandstream 488, so I set it for fax mode: T.38

I did leave DTMF as inband (can't find any docs on what to use for this).
my rx_fax works just fine, but it did for fax pass-through.

So how do I determine if T.38 was negotiated?



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Re: [asterisk-users] Had it with Dell Garbage

2008-02-26 Thread Steve Totaro

On Tue, Feb 26, 2008 at 3:10 PM, Matt <[EMAIL PROTECTED]> wrote:
> I've had it with Dell server garbage.They seem to change RAID
> controllers as much as I change socks, and then the controllers don't work
> with Linux, unless you load a new driver.They sell servers with a PCI-e
> slot in them, but then you get it and find out the RAID controller is using
> the PCI-e slot!   Their sales folks are dumber than rocks, and they change
> them more often than I change underwear.
>  [end rant].
>
> Can anyone recommend an IBM or Gateway server that you have used with
> Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has
> room for one or two PCI-express interface cards?
>

HP DL380 is my baby.

Thanks,
Steve Totaro

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[asterisk-users] Had it with Dell Garbage

2008-02-26 Thread Matt

I've had it with Dell server garbage.They seem to change RAID
controllers as much as I change socks, and then the controllers don't work
with Linux, unless you load a new driver.They sell servers with a PCI-e
slot in them, but then you get it and find out the RAID controller is using
the PCI-e slot!   Their sales folks are dumber than rocks, and they change
them more often than I change underwear.
[end rant].

Can anyone recommend an IBM or Gateway server that you have used with
Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has
room for one or two PCI-express interface cards?
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Re: [asterisk-users] Explain Cause of Error: manager.c:Accept returned -1: Too many open files

2008-02-26 Thread Matthew J. Roth

Dovid B wrote:
> Thanks. I like to know my errors and what cause them. Anyone available to 
> help me pick at their brain to see where its coming from or am I really 
> barking up the wrong tree ?
>   

Dovid,

The number of concurrent calls on the server is tightly related to the 
number of file handles Asterisk opens, so that's the first place that I 
would look.

Take a look at 'ls -l /proc/`cat /var/run/asterisk.pid`/fd/' if you 
really want to analyze the file handles Asterisk has open.  Doing this 
on a server that has experienced the problem and one that hasn't might 
be revealing.

You could also try running 'lsof | egrep "^asterisk "' periodically on 
one of the affected servers and capturing the output to timestamped 
files.  Comparing the data in the files leading up to a "Too many open 
files" error should be revealing.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] chan_ss7 0.10

2008-02-26 Thread Joel Solanki

Hi marek,

Thanks for the update.
I have Sangoma A104D and wanted to use ss7 signalling. I came accross
chan_ss7 but found sifira is not in active development.  But is this
chan_ss7 stable and can be used in production server implementation.
We are going to have 2 to 3 boxes with ss7 signalling using sangoma but I am

looking for open source ss7 implementation which is chan_ss7. so need to
know about stability and recommendation for using on production server.


Please provide your recommendation & suggestions.


Regards,
Joel
- Original Message -
From: "marek cervenka"
<[EMAIL 
PROTECTED]mailto:[EMAIL
 PROTECTED]>
>
To: 
mailto:asterisk-users@lists.digium.com>>;
<[EMAIL 
PROTECTED]mailto:[EMAIL
 PROTECTED]>
>
Sent: Saturday, November 17, 2007 8:49 PM
Subject: [asterisk-users] chan_ss7 0.10


> hi,
>
> i made tarball with some ss7 patches from 
> www.voip-info.orghttp://www.voip-info.org/>and
>  other
> places and put this at 
> http://www.freevoice.cz/chan_ss7-0.10.tgzhttp://www.freevoice.cz/chan_ss7-0.10.tgz>
>
> Sifira is not in active development anymore :( (but they made good
> work! thanks)
>
> from Changelog
> New in version 0.10 (community version)
> - port to asterisk 1.4.14 
> (http://br.geocities.com/bruno_agostinho/http://br.geocities.com/bruno_agostinho/>
)
> - added E prefix for emergency calls 
> (www.tvtrinec.czhttp://www.tvtrinec.cz/>
)
> - some stability fixes 
> (www.tvtrinec.czhttp://www.tvtrinec.cz/>
)
> - sangoma&zaptel example config
> - RBT (?)
> - autoPC+uptime+watermark+stats 
> (www.ss7.plhttp://www.ss7.pl/>
)
> - cic block/unblock fix (tomasz.paszkowski at ctinf.pl)
> - local/remote hangup info in NOTICE (cervajs at freevoice.cz)
>
> please test and report
> thanks
>
> ---
> Marek Cervenka
> ===
>
>
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Re: [asterisk-users] Sip trunk mystery

2008-02-26 Thread Dirk Enrique Seiffert

Hi Steve,

> Does it retransmit the invite six times and then hangup?  When I have
> seen this it was a firewall issue on the remote (provider) side.
>

Indeed it tries seven times. But I think this is the Asterisk default. The
same account configured in my Snom Phone works without problem, - from
same network to same network.

Thanks al lot

Enrique



Cartagena - Colombia
http://www.sipcolombia.com


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Re: [asterisk-users] How can I call cheap to UK cell phones

2008-02-26 Thread Erik Anderson

On Tue, Feb 26, 2008 at 1:19 PM, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:
> Greetings,
>
> How can I call cheap to UK cell phones. I am located in Toronto, Canada, but
> need to call UK cell phones both from Toronto and London.

I'd guess you could get an account with one of these providers:

http://voip-info.org/wiki/view/VOIP+Service+Providers+Business+Europe#UnitedKingdom

-erik

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Re: [asterisk-users] Problem with asterisk and aastra phones

2008-02-26 Thread Adrià Vidal

Ours have been running fine since pointing the aastra.cfg to the LAN NTP.
Don't know what can be happening with yours.

On Tue, Feb 26, 2008 at 12:52 AM, Marius Muja <[EMAIL PROTECTED]> wrote:

> There already is an ntp server on the LAN, but the phones still freeze.
>
> On Mon, Feb 25, 2008 at 2:18 PM, Adrià Vidal <[EMAIL PROTECTED]> wrote:
>
> > Aastra tech are a bit slow, be sure to put a ntp server into your LAN
> > and point Aastra's to it.
> > Your problems will be solved.
> >
> > Adrià Vidal
> >
> >
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> >
>
>
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-- 
--
Adrià Vidal
[EMAIL PROTECTED]
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[asterisk-users] How can I call cheap to UK cell phones

2008-02-26 Thread Zeeshan Zakaria

Greetings,

How can I call cheap to UK cell phones. I am located in Toronto, Canada, but
need to call UK cell phones both from Toronto and London.

-- 
Zeeshan A Zakaria
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Re: [asterisk-users] AMD on a SIP trunk...

2008-02-26 Thread BJ Weschke

 Add an answer() and a playback of 1 second of silence or something else 
to make sure the RTP is nailed up. AMD can/will hang if it has no media 
to analyze.

Carlos Chavez wrote:
>   We have an Asterisk server with a small outgoing call center.  We use
> AMD and it usually works very well on Zap channels (E1 PRI).  We added a
> couple of SIP trunks to reduce long distance costs but now AMD gets
> stuck when the call goes out through the SIP channels.  Here is an
> example call using a SIP line:
>
> -- Executing [EMAIL PROTECTED]:1]
> Set("Local/[EMAIL PROTECTED],2", "CIDTEMP="49875&calllogId=135514"
> <016566275538>") in new stack
> -- Executing [EMAIL PROTECTED]:2]
> Dial("Local/[EMAIL PROTECTED],2", "SIP/juarez-60/6275538|25|C") in
> new stack
> -- Called juarez-60/6275538
> -- SIP/juarez-60-0892f740 is making progress passing it to
> Local/[EMAIL PROTECTED],2
> -- SIP/juarez-60-0892f740 answered Local/[EMAIL PROTECTED],2
> -- Executing [EMAIL PROTECTED]:1] Answer("Local/[EMAIL PROTECTED],1", "")
> in new stack
> -- Executing [EMAIL PROTECTED]:2] AMD("Local/[EMAIL PROTECTED],1", "") in
> new stack
> -- AMD: Local/[EMAIL PROTECTED],1 016566275538 (null) (Fmt: 64)
> -- AMD: initialSilence [3500] greeting [2500] afterGreetingSilence
> [800] totalAnalysisTime [5000] minimumWordLength [100]
> betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold
> [256] 
>
>   AMD just stops and it takes over a minute until the line is dropped.
> The same number dialed through Zap works without a hitch.  What could be
> the reason?  If I dial the same number without AMD I can talk to the
> other person so I know the SIP line is fine.
>
>
>   
> 
>
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-- 
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/




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Re: [asterisk-users] Attatch monitor recording to a voicemail

2008-02-26 Thread Dovid B


- Original Message - 
From: "Jared Smith" <[EMAIL PROTECTED]>
To: "Dovid B" <[EMAIL PROTECTED]>
Cc: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Tuesday, February 26, 2008 8:08 PM
Subject: Re: [asterisk-users] Attatch monitor recording to a voicemail


> On Tue, 2008-02-26 at 19:48 +0200, Dovid B wrote:
>> Jared,
>> You mentioned that you have done it in the past.Can you post your code 
>> here
>
> No, unfortunately this was done under NDA, but the general gist goes
> like this:
>
> Dialplan pieces:
>
> A) Get automon working.  Don't forget to set the DYNAMIC_FEATURES
> variable (either as a global variable, or on a channel-by-channel
> basis).
>
> B) Set the MONITOR_EXEC variable to point to the program explained
> below.  (I suggest learning about how to prepend an underscore or two to
> the beginning of a variable name to ensure that it gets inherited by any
> created channels.)  For example, I set:
>
> exten => example,n,Set(_MONITOR_EXEC=/usr/local/bin/automon-to-vm)
>
> C) Set the MONITOR_EXEC_ARGS variable to tell the program below any
> information which it might need.  My example is a bit messy (and
> probably overengineered), but here it is.  (Just promise me you won't
> laugh at it.)
>
> exten => example,n,Set(ORIGDATE=${BASE64_ENCODE(${STRFTIME(${EPOCH},,%a
> %b %e %r %Z %Y)})})
> exten => example,n,Set(_MONITOR_EXEC_ARGS=${MACRO_EXTEN}^${CONTEXT}^
> ${MACRO_CONTEXT}^${EXTEN}^${PRIORITY}^${CHANNEL}^
> ${BASE64_ENCODE(${CALLERID(all)})}^${ORIGDATE}^${EPOCH})
>
> OK, now for the program itself to take the recorded audio and send it to
> voicemail:
>
> 1) Mix the inbound and outbound audio channels.  I use soxmix to do
> this, and put the inbound on the left channel and the outbound on the
> right channel.  For completeness' sake, convert into the various formats
> you have defined in voicemail.conf (typically wav, WAV, and gsm).
> 2) Iterate
> through /var/spool/asterisk/voicemail/${VM-CONTEXT}/${MAILBOX}/INBOX/,
> looking for msg.*, msg0001.*, etc. until you no longer find that
> message.  Move the audio file(s) created in step 2 to this location.
> 3) Create an appropriate .txt file that goes with the message.  If you
> look at a regular voicemail recording, you'll see there's a .txt file in
> addition to the audio file(s). This text file specifies when the
> voicemail recording was left, the duration, the CallerID information,
> etc.
>
> Anybody with reasonable programming skills should be able to do this in
> a few dozen lines of code -- it's nothing too spectacular.
>
> -- 
> Jared Smith
> Community Relations Manager
> Digium, Inc.
>

Thanks a lot. 



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Re: [asterisk-users] Explain Cause of Error: manager.c:Accept returned -1: Too many open files

2008-02-26 Thread Dovid B


- Original Message - 
From: "Jared Smith" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Tuesday, February 26, 2008 8:11 PM
Subject: Re: [asterisk-users] Explain Cause of Error: manager.c:Accept 
returned -1: Too many open files


> On Tue, 2008-02-26 at 19:53 +0200, Dovid B wrote:
>> While I know that "upping" ulimit will fix the issue I am trying to
>> understand what will cause it.
>
> There are *lots* of things in Asterisk that open file handles, and to
> try to track them down is probably a waste of time.  Just configure your
> system to allow Asterisk to open more files, and call it good!
>
> -- 
> Jared Smith
> Community Relations Manager
> Digium, Inc.
>
>
Jared,
Thanks. I like to know my errors and what cause them. Anyone available to 
help me pick at their brain to see where its coming from or am I really 
barking up the wrong tree ?

Dovid 



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[asterisk-users] AMD on a SIP trunk...

2008-02-26 Thread Carlos Chavez

We have an Asterisk server with a small outgoing call center.  We use
AMD and it usually works very well on Zap channels (E1 PRI).  We added a
couple of SIP trunks to reduce long distance costs but now AMD gets
stuck when the call goes out through the SIP channels.  Here is an
example call using a SIP line:

-- Executing [EMAIL PROTECTED]:1]
Set("Local/[EMAIL PROTECTED],2", "CIDTEMP="49875&calllogId=135514"
<016566275538>") in new stack
-- Executing [EMAIL PROTECTED]:2]
Dial("Local/[EMAIL PROTECTED],2", "SIP/juarez-60/6275538|25|C") in
new stack
-- Called juarez-60/6275538
-- SIP/juarez-60-0892f740 is making progress passing it to
Local/[EMAIL PROTECTED],2
-- SIP/juarez-60-0892f740 answered Local/[EMAIL PROTECTED],2
-- Executing [EMAIL PROTECTED]:1] Answer("Local/[EMAIL PROTECTED],1", "")
in new stack
-- Executing [EMAIL PROTECTED]:2] AMD("Local/[EMAIL PROTECTED],1", "") in
new stack
-- AMD: Local/[EMAIL PROTECTED],1 016566275538 (null) (Fmt: 64)
-- AMD: initialSilence [3500] greeting [2500] afterGreetingSilence
[800] totalAnalysisTime [5000] minimumWordLength [100]
betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold
[256] 

AMD just stops and it takes over a minute until the line is dropped.
The same number dialed through Zap works without a hitch.  What could be
the reason?  If I dial the same number without AMD I can talk to the
other person so I know the SIP line is fine.


-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Explain Cause of Error: manager.c: Accept returned -1: Too many open files

2008-02-26 Thread Jared Smith

On Tue, 2008-02-26 at 19:53 +0200, Dovid B wrote:
> While I know that "upping" ulimit will fix the issue I am trying to
> understand what will cause it. 

There are *lots* of things in Asterisk that open file handles, and to
try to track them down is probably a waste of time.  Just configure your
system to allow Asterisk to open more files, and call it good!

-- 
Jared Smith
Community Relations Manager
Digium, Inc.

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Re: [asterisk-users] Attatch monitor recording to a voicemail

2008-02-26 Thread Jared Smith

On Tue, 2008-02-26 at 19:48 +0200, Dovid B wrote:
> Jared,
> You mentioned that you have done it in the past.Can you post your code here 

No, unfortunately this was done under NDA, but the general gist goes
like this:

Dialplan pieces:

A) Get automon working.  Don't forget to set the DYNAMIC_FEATURES
variable (either as a global variable, or on a channel-by-channel
basis).

B) Set the MONITOR_EXEC variable to point to the program explained
below.  (I suggest learning about how to prepend an underscore or two to
the beginning of a variable name to ensure that it gets inherited by any
created channels.)  For example, I set:

exten => example,n,Set(_MONITOR_EXEC=/usr/local/bin/automon-to-vm)

C) Set the MONITOR_EXEC_ARGS variable to tell the program below any
information which it might need.  My example is a bit messy (and
probably overengineered), but here it is.  (Just promise me you won't
laugh at it.)

exten => example,n,Set(ORIGDATE=${BASE64_ENCODE(${STRFTIME(${EPOCH},,%a
%b %e %r %Z %Y)})})
exten => example,n,Set(_MONITOR_EXEC_ARGS=${MACRO_EXTEN}^${CONTEXT}^
${MACRO_CONTEXT}^${EXTEN}^${PRIORITY}^${CHANNEL}^
${BASE64_ENCODE(${CALLERID(all)})}^${ORIGDATE}^${EPOCH})

OK, now for the program itself to take the recorded audio and send it to
voicemail:

1) Mix the inbound and outbound audio channels.  I use soxmix to do
this, and put the inbound on the left channel and the outbound on the
right channel.  For completeness' sake, convert into the various formats
you have defined in voicemail.conf (typically wav, WAV, and gsm).
2) Iterate
through /var/spool/asterisk/voicemail/${VM-CONTEXT}/${MAILBOX}/INBOX/,
looking for msg.*, msg0001.*, etc. until you no longer find that
message.  Move the audio file(s) created in step 2 to this location.
3) Create an appropriate .txt file that goes with the message.  If you
look at a regular voicemail recording, you'll see there's a .txt file in
addition to the audio file(s). This text file specifies when the
voicemail recording was left, the duration, the CallerID information,
etc.

Anybody with reasonable programming skills should be able to do this in
a few dozen lines of code -- it's nothing too spectacular.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.

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Re: [asterisk-users] Cannot hear voice through SIP Phone from one side

2008-02-26 Thread Dovid B

This smells of NAT issues. Since you said that you do have the server set up 
as a DMZ did you set NAT=yes, externip= ? as well as the other NAT settings 
?

- Original Message - 
From: "Sanjoy Rath" <[EMAIL PROTECTED]>
To: 
Sent: Tuesday, February 05, 2008 10:32 PM
Subject: [asterisk-users] Cannot hear voice through SIP Phone from one side


>I have a asterisk server. Two SIP Soft XLites are connected to the server. 
>I am able to make
> calls from one SIP Phones to the other SIP Phones and landlines 
> successfully. The SIP Soft Phone on th eother side can hear my voice but I 
> cannot hear their voice.
>
> They can call my local cell phone as well. Samething, they can hears my 
> voice, I cannot hear their voice.
>
> The microphone and speakers are working on both sides because we are able 
> to use google talk and are able to talk successfully. But it would not 
> work on XLite over asterisk for some reason.
>
> The Asterisk server is a linux server. There is no firewall between the 
> servers. It is in a DMZ.
>
> Any suggestion how to get it to work :)
>
> Thanks,
> Sanjoy.
>
>
> 
> 
> Never miss a thing.  Make Yahoo your home page.
> http://www.yahoo.com/r/hs
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 



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[asterisk-users] Explain Cause of Error: manager.c: Accept returned -1: Too many open files

2008-02-26 Thread Dovid B

Hi List,
While I know that "upping" ulimit will fix the issue I am trying to understand 
what will cause it. I have a few set ups that are almost exactly the same yet 
some machines used to give this error often and others don't. I also noticed 
the error a lot more on my boxes running 1.4.X.

TIA.

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Re: [asterisk-users] Attatch monitor recording to a voicemail

2008-02-26 Thread Dovid B


- Original Message - 
From: "Jared Smith" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Tuesday, February 19, 2008 11:53 PM
Subject: Re: [asterisk-users] Attatch monitor recording to a voicemail


> On Tue, 2008-02-19 at 20:59 +, Ben Willcox wrote:
>> That gets me halfway there, but what I'm wondering about is the process
>> of moving of the recording to the correct place - i.e. should my
>> external program do the following:
>>
>> 1) Check the users voicemail directory for existing message filenames
>> 2) Copy the recording into the voicemail directory named msg.WAV
>> (incremented depending on number of existing messages)
>> 3) Create the msg.txt file in the correct format
>>
>> or is there another way that will sort this all out automatically? What
>> would happen if a real voicemail drops into that directory while my
>> external script is halfway through copying/creating for example...
>
> No, you're on the right track.
>
> In a nutshell, you'll have to figure out whether app_voicemail looks a
> the .txt files or the actually recordings first (when determining the
> next message number), and have your program create that file first.  (As
> I recall, it's the .txt files that app_voicemail looks at.)  Obviously
> if you're not careful, there can be a race condition there, but the idea
> is that if a caller were to leave a voicemail on the system, it would
> see the file already there and choose the next number.
>
> -- 
> Jared Smith
> Community Relations Manager
> Digium, Inc.
>

Jared,
You mentioned that you have done it in the past.Can you post your code here 
?

Thanks.

Dovid 



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Re: [asterisk-users] test

2008-02-26 Thread Joel Solanki

Thanks,
Joel

On Tue, Feb 26, 2008 at 10:47 PM, Erik Anderson <[EMAIL PROTECTED]> wrote:

> On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki <[EMAIL PROTECTED]>
> wrote:
> > checking wheather my mail goes to asterisk users mailling list or not
>
> ACK.
>
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Re: [asterisk-users] Sip trunk mystery

2008-02-26 Thread Steve Totaro

On Tue, Feb 26, 2008 at 12:31 PM, Dirk Enrique Seiffert
<[EMAIL PROTECTED]> wrote:
> Hello,
>
>  I am trying to add a sip-trunk to my Asterisk 1.4.15/Elastix 0.9.2 server.
>  The system is in production with local extensions, a zap trunk and a
>  working sip trunk with sipgate.de.
>
>  My asterisk server is behind a NAT/Firewall, anyhow it registers and works
>  well with sipgate.de on incoming and outgoing calls.
>
>  I aquired an account with a reseller net-voz.com: I did some testing with
>  the account directly from a Snom300 phone - works without a problem,
>  (behind the nat) I spent hours testing and adjusting the trunk
>  configuration for net-voz, maybe sombody on the list can give a helpful hint:
>
>  First of all: Registry works!
>
>  pbx*CLI> sip show registry
>  HostUsername   Refresh State
>   Reg.Time
>  sip.net-voz.com:5060xx6168 585 Registered
>   Tue, 26 Feb 2008 10:47:58
>  sipgate.de:5060 0823105 Registered
>   Tue, 26 Feb 2008 10:56:22
>
>  This is my config:
>
>  [ringtime]
>  username=5515816168
>  type=peer
>  type=friend
>  secret=118873
>  insecure=very
>  host=sip.net-voz.com
>  fromuser=5515816168
>  fromdomain=sip.net-voz.com
>  canreinvite=no
>  call-limit=50
>
>  I tried faking the user agent (without success)
>
>  useragent = Grandstream BT100 1.0.4.49
>  externip=xx.xx.116.229
>  localnet=192.168.8.0/255.255.255.0
>
>  On my gateway I can see the following with tcpdump:
>
>  listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes
>  11:05:57.386827 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length:
>  810
>  11:05:57.452414 IP 190.144.151.212.sip > pbx.lintec.sip: SIP, length: 442
>  11:05:57.453021 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length:
>  385
>  11:05:57.453587 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030
>  11:05:58.452868 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030
>  11:06:01.453814 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030
>
>  On the astersik CLI the logs show:
>
>  Audio is at 192.168.8.3 port 14800
>  Adding codec 0x4 (ulaw) to SDP
>  Adding codec 0x8 (alaw) to SDP
>  Adding non-codec 0x1 (telephone-event) to SDP
>  Reliably Transmitting (no NAT) to 190.144.151.212:5060:
>  INVITE sip:[EMAIL PROTECTED] SIP/2.0
>  Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport
>  From: "901" ;tag=as3c6dfee5
>  To: 
>  Contact: 
>  Call-ID: [EMAIL PROTECTED]
>  CSeq: 103 INVITE
>  User-Agent: Asterisk PBX
>  Max-Forwards: 70
>  Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch",
>  algorithm=MD5, uri="sip:[EMAIL PROTECTED]",
>  nonce="12040419552605702055508208",
>  response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque=""
>  Date: Tue, 26 Feb 2008 16:09:09 GMT
>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>  Supported: replaces
>  Content-Type: application/sdp
>  Content-Length: 260
>
>  v=0
>  o=root 2381 2382 IN IP4 192.168.8.3
>  s=session
>  c=IN IP4 192.168.8.3
>  t=0 0
>  m=audio 14800 RTP/AVP 0 8 101
>  a=rtpmap:0 PCMU/8000
>  a=rtpmap:8 PCMA/8000
>  a=rtpmap:101 telephone-event/8000
>  a=fmtp:101 0-16
>  a=silenceSupp:off - - - -
>  a=ptime:20
>  a=sendrecv
>
>  ---
>  Retransmitting #1 (no NAT) to 190.144.151.212:5060:
>  INVITE sip:[EMAIL PROTECTED] SIP/2.0
>  Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport
>  From: "901" ;tag=as3c6dfee5
>  To: 
>  Contact: 
>  Call-ID: [EMAIL PROTECTED]
>  CSeq: 103 INVITE
>  User-Agent: Asterisk PBX
>  Max-Forwards: 70
>  Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch",
>  algorithm=MD5, uri="sip:[EMAIL PROTECTED]",
>  nonce="12040419552605702055508208",
>  response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque=""
>  Date: Tue, 26 Feb 2008 16:09:09 GMT
>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>  Supported: replaces
>  Content-Type: application/sdp
>  Content-Length: 260
>
>
>  It looks like the comuunication starts but then gets lost.??
>
>  Any idea is appreciated.
>
>  Thanks
>
>  Enrique
>
>
>
>  Cartagena - Colombia
>  http://www.sipcolombia.com

Does it retransmit the invite six times and then hangup?  When I have
seen this it was a firewall issue on the remote (provider) side.

Thanks,
Steve Totaro

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[asterisk-users] Sip trunk mystery

2008-02-26 Thread Dirk Enrique Seiffert

Hello,

I am trying to add a sip-trunk to my Asterisk 1.4.15/Elastix 0.9.2 server.
The system is in production with local extensions, a zap trunk and a
working sip trunk with sipgate.de.

My asterisk server is behind a NAT/Firewall, anyhow it registers and works
well with sipgate.de on incoming and outgoing calls.

I aquired an account with a reseller net-voz.com: I did some testing with
the account directly from a Snom300 phone - works without a problem,
(behind the nat) I spent hours testing and adjusting the trunk
configuration for net-voz, maybe sombody on the list can give a helpful hint:

First of all: Registry works!

pbx*CLI> sip show registry
HostUsername   Refresh State
 Reg.Time
sip.net-voz.com:5060xx6168 585 Registered
 Tue, 26 Feb 2008 10:47:58
sipgate.de:5060 0823105 Registered
  Tue, 26 Feb 2008 10:56:22

This is my config:

[ringtime]
username=5515816168
type=peer
type=friend
secret=118873
insecure=very
host=sip.net-voz.com
fromuser=5515816168
fromdomain=sip.net-voz.com
canreinvite=no
call-limit=50

I tried faking the user agent (without success)

useragent = Grandstream BT100 1.0.4.49
externip=xx.xx.116.229
localnet=192.168.8.0/255.255.255.0

On my gateway I can see the following with tcpdump:

listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes
11:05:57.386827 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length:
810
11:05:57.452414 IP 190.144.151.212.sip > pbx.lintec.sip: SIP, length: 442
11:05:57.453021 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length:
385
11:05:57.453587 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030
11:05:58.452868 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030
11:06:01.453814 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030

On the astersik CLI the logs show:

Audio is at 192.168.8.3 port 14800
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 190.144.151.212:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport
From: "901" ;tag=as3c6dfee5
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch",
algorithm=MD5, uri="sip:[EMAIL PROTECTED]",
nonce="12040419552605702055508208",
response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque=""
Date: Tue, 26 Feb 2008 16:09:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 2381 2382 IN IP4 192.168.8.3
s=session
c=IN IP4 192.168.8.3
t=0 0
m=audio 14800 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #1 (no NAT) to 190.144.151.212:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport
From: "901" ;tag=as3c6dfee5
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch",
algorithm=MD5, uri="sip:[EMAIL PROTECTED]",
nonce="12040419552605702055508208",
response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque=""
Date: Tue, 26 Feb 2008 16:09:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260


It looks like the comuunication starts but then gets lost.??

Any idea is appreciated.

Thanks

Enrique



Cartagena - Colombia
http://www.sipcolombia.com





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Re: [asterisk-users] test

2008-02-26 Thread Erik Anderson

On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki <[EMAIL PROTECTED]> wrote:
> checking wheather my mail goes to asterisk users mailling list or not

ACK.

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[asterisk-users] test

2008-02-26 Thread Joel Solanki

checking wheather my mail goes to asterisk users mailling list or not
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Re: [asterisk-users] [URGENT] Zap channels fail to load

2008-02-26 Thread Andres Jimenez

On Tue, Feb 26, 2008 at 1:35 PM, Louwrens Benadé <[EMAIL PROTECTED]> wrote:
> Why does this look suspiciously like a T1 line? Are you sure this is a
>  fractional E1?

My provider names the line a PRA, but this is understood anywhere as a
PRI (no fractional).

From the Asterisk configuration point of view, is there any other
difference between the fractional and full PRI apart from the number
of channels?

Do you guys know of any particular setting our provider (Eircom,
Ireland) could require?

I have a problem with DTMF when the PRI is the one carrying the call:
"1s" and "2s" are not transmitted. If the call is internal or carryed
by and IAX trunk + SIP it works nicely.

Regards,

-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
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Re: [asterisk-users] Parked calls - can't pickup

2008-02-26 Thread Jared Smith

On Tue, 2008-02-26 at 10:01 -0500, OCG Technical Support wrote:
> It looks like I have a conflict!  (See results of diaplan show below).
> How
> can I force the parkedcalls context to be matched first?  (I include
> parkedcalls before I define the _X. priority).
> 
> pbx*CLI> dialplan show [EMAIL PROTECTED]
> [ Context 'entryocginternal' created by 'pbx_config' ]
>   '_X.' =>  1. Macro(dialexternal|${EXTEN}|${dialaccount})
> [pbx_config]
> 2. Goto(s|1)
> [pbx_config]
> [ Included context 'parkedcalls' created by 'res_features' ]
>   '701' =>  1. ParkedCall(701)
> [res_features]

You can always point your phones at a new context that looks like:

[some-other-context-name]
include => parkedcalls
include => entryocginternal

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] DDNS and host: updating when destination IP changes

2008-02-26 Thread Chris Mason (Lists)

bilal ghayyad wrote:
> OK that worked, but how can we resolve it without need
> to type the command manually, as the destination might
> change its IP address without our notice, so the
> question is:
>
> How can the host be updated periodically (like
> externrefresh settings), but need it for host, any
> help?
>   
If you set up that host as a dynamic peer, it should work automatically.
Also, use the dns name rather than the IP
Enable dns in dnsmgr.conf

These should do it.

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


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Re: [asterisk-users] problem transferring calls some of the times

2008-02-26 Thread Raúl Gómez C.

Ian,

I'm having *THE SAME PROBLEM* and I've noticed that when a transfer fail
(only happens when receptionist dial an external number) the call is marker
as "NO ANSWER" in the CDR, even when the call *HAS BEEN ANSWERED* by the
other party (the callee). See my previous post below.


http://lists.digium.com/pipermail/asterisk-users/2008-February/206228.html

http://lists.digium.com/pipermail/asterisk-users/2008-February/206533.html


So I think Asterisk doesn't want to transfer calls that hasn't been
answered, *OR* maybe are the phone itself (since I have the EXACT phones you
have GXP-2000), that is causing the problem.

BTW: I have the same Asterisk, Zaptel and Libpri versions as you have.

Please check your CDR and look if the calls that has failed to transfer are
marked as "NO ANSWER".


Thanks, I hope we can solve this anytime soon...

-- 
Nacho
Linux Counter #156439
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Re: [asterisk-users] Parked calls - can't pickup

2008-02-26 Thread OCG Technical Support

Another clue...I repeated the dialplan show command for the 700 extension
and it too is listed AFTER the _X. match.  However, forward a call to 700
works.  Why would calling 701 not pickup the call?  (Why is it matching the
_X. extension)

Thanks!

pbx*CLI> dialplan show [EMAIL PROTECTED]
[ Context 'entryocginternal' created by 'pbx_config' ]
  '_X.' =>  1. Macro(dialexternal|${EXTEN}|${dialaccount})
[pbx_config]
2. Goto(s|1)
[pbx_config]
[ Included context 'parkedcalls' created by 'res_features' ]
  '700' =>  1. Park()
[res_features]


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> OCG Technical Support
> Sent: Tuesday, February 26, 2008 10:02 AM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] Parked calls - can't pickup
> 
> It looks like I have a conflict!  (See results of diaplan 
> show below).  How can I force the parkedcalls context to be 
> matched first?  (I include parkedcalls before I define the 
> _X. priority).
> 
> pbx*CLI> dialplan show [EMAIL PROTECTED] [ Context 
> 'entryocginternal' created by 'pbx_config' ]
>   '_X.' =>  1. Macro(dialexternal|${EXTEN}|${dialaccount})
> [pbx_config]
> 2. Goto(s|1)
> [pbx_config]
> [ Included context 'parkedcalls' created by 'res_features' ]
>   '701' =>  1. ParkedCall(701)
> [res_features]
> 
> -= 2 extensions (3 priorities) in 2 contexts. =- pbx*CLI>
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Jared 
> > Smith
> > Sent: Tuesday, February 26, 2008 9:46 AM
> > To: Asterisk Users List
> > Subject: Re: [asterisk-users] Parked calls - can't pickup
> >
> > On Mon, 2008-02-25 at 21:03 -0500, Michelle Dupuis wrote:
> > > I have 2 contexts in my extensions.conf: internal and
> > external calls.
> > > I have included the parkedcalls context in both.
> > >
> > > Do I need to preface the include with a # symbol?
> >
> > No, you do not.  You simply need a like that says:
> >
> > include => parkedcalls
> >
> > A couple of things to check:
> >
> > 1) make sure you haven't changed the context name in features.conf 
> > (from "parkedcalls" to something else) and
> > 2) you can always type "dialplan show [EMAIL PROTECTED]" from the 
> > Asterisk CLI to see what would match if you dialed extension 701 in 
> > that context.
> >
> > --
> > Jared Smith
> > Community Relations Manager
> > Digium, Inc.
> >
> >
> > ___
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> http://www.api-digital.com --
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> > asterisk-users mailing list
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
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Re: [asterisk-users] TDM400P dialout problem

2008-02-26 Thread John covici

OK, here is the problem -- how do I compile 1.4.7.1 using kernel
2.6.24 -- when I try the make KBUILD_NOPEDANTIC=1 I get the no such
file or directory that I already mentioned -- when I take out that
KBUILD option, I get what I got before -- the error from the kernel
module build about the CFLAGS being changed in the Makefile.  So how
can I compile this thing till there is a fix for the bug or the
regression is removed?


on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
 > Slightly edited your message:
 > 
 > On Tue, Feb 26, 2008 at 08:48:13AM -0500, John covici wrote:
 > > on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
 > >  > On Tue, Feb 26, 2008 at 07:50:09AM -0500, John covici wrote:
 > >  > > I am getting this strange error:
 > >  > > 
 > >  > > make[1]: Entering directory `/usr/src/zaptel-1.4.7.1'
 > >  > > make -C /lib/modules/2.6.24-gentoo-r2/build 
 > > SUBDIRS=/usr/src/zaptel-1.4.7.1 HOTPLUG_FIRMWARE=yes modules
 > >  > > make[2]: Entering directory `/usr/src/linux-2.6.24-gentoo-r2'
 > >  > >   CC [M]  /usr/src/zaptel-1.4.7.1/wcfxo.o
 > >  > > /usr/src/zaptel-1.4.7.1/wcfxo.c:38:27: error: zaptel/zaptel.h: No 
 > > such file or directory
 > >  > > 
 > >  > > But the file is there.
 > >  > > At least its in the source directory.
 > >  > > I did the ./configure and the make menuselect to eliminate some
 > >  > > unnecessary modules.
 > >  > 
 > >  > -DSTANDALONE_ZAPATA not getting through to the EXTRA_CFLAGS?
 > >
 > > And why this weird error anyway?
 > 
 > Because STANDALONE_ZAPATA was not #define-d . 
 > 
 > >  > 
 > >  > Note: luckily you didn't have zaptel installed and this it didn't use a
 > >  > different version of zaptel.h from /usr/include/zaptel/zaptel.h .
 > >
 > > I do have the newer one installed -- is this the problem?
 > 
 > Hmmm I guess /ysr/include is not in the include path for kernel
 > modules building. A good thing in this case.
 > 
 > -- 
 >Tzafrir Cohen
 > icq#16849755  jabber:[EMAIL PROTECTED]
 > +972-50-7952406   mailto:[EMAIL PROTECTED]
 > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
 > 
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 >http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]

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Re: [asterisk-users] Parked calls - can't pickup

2008-02-26 Thread OCG Technical Support

It looks like I have a conflict!  (See results of diaplan show below).  How
can I force the parkedcalls context to be matched first?  (I include
parkedcalls before I define the _X. priority).

pbx*CLI> dialplan show [EMAIL PROTECTED]
[ Context 'entryocginternal' created by 'pbx_config' ]
  '_X.' =>  1. Macro(dialexternal|${EXTEN}|${dialaccount})
[pbx_config]
2. Goto(s|1)
[pbx_config]
[ Included context 'parkedcalls' created by 'res_features' ]
  '701' =>  1. ParkedCall(701)
[res_features]

-= 2 extensions (3 priorities) in 2 contexts. =-
pbx*CLI> 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jared Smith
> Sent: Tuesday, February 26, 2008 9:46 AM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] Parked calls - can't pickup
> 
> On Mon, 2008-02-25 at 21:03 -0500, Michelle Dupuis wrote:
> > I have 2 contexts in my extensions.conf: internal and 
> external calls.  
> > I have included the parkedcalls context in both.
> >
> > Do I need to preface the include with a # symbol?
> 
> No, you do not.  You simply need a like that says:
> 
> include => parkedcalls
> 
> A couple of things to check:
> 
> 1) make sure you haven't changed the context name in 
> features.conf (from "parkedcalls" to something else) and
> 2) you can always type "dialplan show [EMAIL PROTECTED]" from 
> the Asterisk CLI to see what would match if you dialed 
> extension 701 in that context.
> 
> --
> Jared Smith
> Community Relations Manager
> Digium, Inc.
> 
> 
> ___
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> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [asterisk-users] DDNS and host: updating when destination IP changes

2008-02-26 Thread bilal ghayyad

OK that worked, but how can we resolve it without need
to type the command manually, as the destination might
change its IP address without our notice, so the
question is:

How can the host be updated periodically (like
externrefresh settings), but need it for host, any
help?

Regards
Bilal



Just as a silly one, try 'asterisk -rx "iax2 reload"'.
I'm not sure if
 it'll
work or not, but it should force a recheck of the
hostname.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of bilal
 ghayyad
Sent: 25 February 2008 02:21 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DDNS and host: updating when
destination IP
changes

Hi All;

I am using IAX Trunk and I used ddns (dyndns.org) with
the host (host=xyz.dyndns.org), when I restart the
router who has the hostname xyz.dyndns.org then its IP
address change and updated, but at asterisk level
still it keeps sending for the old IP address and
sometimes this problem does not resolve until I
restart asterisk.

Any one faced this and has idea how to resolve it so
Asterisk can check the new IP address for the
host=xyz.dyndns.org each call?

Any help?
Regards
Bilal 




  

Never miss a thing.  Make Yahoo your home page. 
http://www.yahoo.com/r/hs

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Re: [asterisk-users] Parked calls - can't pickup

2008-02-26 Thread Jared Smith

On Mon, 2008-02-25 at 21:03 -0500, Michelle Dupuis wrote:
> I have 2 contexts in my extensions.conf: internal and external calls.  I
> have included the parkedcalls context in both.
> 
> Do I need to preface the include with a # symbol?  

No, you do not.  You simply need a like that says:

include => parkedcalls

A couple of things to check: 

1) make sure you haven't changed the context name in features.conf (from
"parkedcalls" to something else) and 
2) you can always type "dialplan show [EMAIL PROTECTED]" from the
Asterisk CLI to see what would match if you dialed extension 701 in that
context.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.

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[asterisk-users] How to transfer an unanswered call???

2008-02-26 Thread Raúl Gómez C.

Hi list,

I'm wondering if it's possible to transfer a call that is still ringing???

Actually, the problem is that my telco provider doesn't offer an uniform
method for answer/disconnection supervision, and by that I mean, some of
it's line (I think) offer a polarity reversal, but other lines (of the same
service provider) do not offer anything at all, so the answer of a call
can't be detected, and this is what is causing me a lot of trouble, because
when the attendant calls to one of this lines and want to transfer this call
to an internal extension the transfer fails, and the only common thing in
this situation is that the calls are marked in the CDR as "NO ANSWER".

Using the options "busydetect","busycount" and "callprogress" in the
zapata.conf file has worked fine for disconnection.

I've called the technical support of my telco provider and when I ask about
this trouble they has no clue at all of what I'm talking about.

That's why I'm looking for others method so solve this issues, any ideas???
Is my approach right?? Or there's something else I should try first???

*My setup:*
Asterisk 1.4.17
Zaptel 1.4.7.1
Sangoma Remora Card A400D with EC (2 FXS / 10 FXO)
Wanpipe 3.2.1
Grandstream GXP-2000 / GXP-2020 IP Phones


Thanks a lot for any help...


PD: This is a production system, so I can test some things but with less
flexibility than with a test system, maybe after hours...


*My config files:*

*zapata.conf:*
;autogenerated by /usr/local/sbin/config-zaptel  do not hand edit
;Zaptel Channels Configurations (zapata.conf)
;
;For detailed zapata options, view /etc/asterisk/zapata.conf.orig

[trunkgroups]

[channels]
context=default
;usecallerid=yes
;hidecallerid=no
callwaiting=yes
usecallingpres=yes
;callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.5
txgain=0.0
group=0
callgroup=0
pickupgroup=1

callerid="Llamada Externa"
busydetect=yes
busycount=4
callprogress=yes
progzone=us
hanguponpolarityswitch=yes

immediate=no

;Sangoma A400 [slot:4 bus:16 span:1]
;context=vigilancia
context=fax
group=1
signalling = fxo_ks
channel => 1

context=fax
group=1
signalling = fxo_ks
channel => 2

context=from-zaptel
group=0
signalling = fxs_ks
channel => 3

context=from-zaptel
group=0
signalling = fxs_ks
channel => 4

context=from-zaptel
group=0
signalling = fxs_ks
channel => 5

context=from-zaptel
group=0
signalling = fxs_ks
channel => 6

;context=from-zaptel
context=fax
group=0
signalling = fxs_ks
channel => 7

context=from-zaptel
group=2
signalling = fxs_ks
channel => 8

context=from-zaptel
group=2
signalling = fxs_ks
channel => 9

context=from-zaptel
group=3
signalling = fxs_ks
channel => 10

context=from-zaptel
group=4
signalling = fxs_ks
channel => 11

context=from-zaptel
group=5
signalling = fxs_ks
channel => 12


*zaptel.conf:*
# Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit
# Zaptel Channels Configurations (zaptel.conf)
#
loadzone=us
defaultzone=us

#Sangoma A400 [slot:4 bus:16 span:1]
fxoks=1
fxoks=2
fxsks=3
fxsks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
fxsks=9
fxsks=10
fxsks=11
fxsks=12


-- 
Nacho
Linux Counter #156439
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Re: [asterisk-users] Still can't pickup parked call

2008-02-26 Thread Jared Smith

On Tue, 2008-02-26 at 07:44 -0500, OCG Technical Support wrote:
> Well, I don't have a 701 extension defined but I do have _XXX which is where
> this call is jumping when I dial 701 to pickup.
> 
> I have the "include => parkedcalls" above the _XXX definition, so I assumed
> that parked calls would be matched first.  As well, since the 700 is
> matching to parked calls, I assumed 701 would as well.  
> 
> Still stuck 

If you want this to work without changing your pattern matching, change
your phones to point to a new context that looks like this:

[make-call-pickup-work]
include => parkedcalls
include => the-old-context-with-the-pattern-match

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] TDM400P dialout problem

2008-02-26 Thread Tzafrir Cohen

Slightly edited your message:

On Tue, Feb 26, 2008 at 08:48:13AM -0500, John covici wrote:
> on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
>  > On Tue, Feb 26, 2008 at 07:50:09AM -0500, John covici wrote:
>  > > I am getting this strange error:
>  > > 
>  > > make[1]: Entering directory `/usr/src/zaptel-1.4.7.1'
>  > > make -C /lib/modules/2.6.24-gentoo-r2/build 
> SUBDIRS=/usr/src/zaptel-1.4.7.1 HOTPLUG_FIRMWARE=yes modules
>  > > make[2]: Entering directory `/usr/src/linux-2.6.24-gentoo-r2'
>  > >   CC [M]  /usr/src/zaptel-1.4.7.1/wcfxo.o
>  > > /usr/src/zaptel-1.4.7.1/wcfxo.c:38:27: error: zaptel/zaptel.h: No such 
> file or directory
>  > > 
>  > > But the file is there.
>  > > At least its in the source directory.
>  > > I did the ./configure and the make menuselect to eliminate some
>  > > unnecessary modules.
>  > 
>  > -DSTANDALONE_ZAPATA not getting through to the EXTRA_CFLAGS?
>
> And why this weird error anyway?

Because STANDALONE_ZAPATA was not #define-d . 

>  > 
>  > Note: luckily you didn't have zaptel installed and this it didn't use a
>  > different version of zaptel.h from /usr/include/zaptel/zaptel.h .
>
> I do have the newer one installed -- is this the problem?

Hmmm I guess /ysr/include is not in the include path for kernel
modules building. A good thing in this case.

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

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[asterisk-users] Asterisk as useragent registered using 2 accounts

2008-02-26 Thread Rizwan Hisham

Hi all,
I am having a strange problem. I am using my asterisk server AST1 to
register with another asterisk server AST2 using 2 accounts (2 register
commands in sip.conf). I have made 2 local users in AST1, and configured my
dialplan in such a way that both local accounts on AST1 use different trunks
to send the call to AST2 server. These 2 different trunks are for 2 accounts
i have registered on AST1.


line1 ---> trunk1(ON AST1) ---> AST2
line2 ---> trunk2(ON AST1) ---> AST2

These 2 trunks are to differentiate that the call is coming from one of the
2 registered accounts on AST1.

The problem is, my AST2 server cannot differentiate between 2 accounts. It
always dumps the cdr at the end of every call against only one of the 2
registered accounts (acc2 even if im dialing from acc1) on AST1 i.e. the
call always goes out using account-2 even if i dial from accout-1. Here is
my sip.conf

TRUNKS

[acc1]
username=acc1
type=friend
secret=123
qualify=yes
port=9060
nat=yes
insecure=port,invite
host=ip-of-my-AST2
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm

[acc2]
username=acc2
type=friend
secret=123
qualify=yes
port=9060
nat=yes
insecure=port,invite
host=ip-of-my-AST2
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm


REGSITRATION

register => acc1:[EMAIL PROTECTED]:9060
register => acc2:[EMAIL PROTECTED]:9060

local lines on AST1 use trunk acc1 and acc2 to throw calls to my AST2.

But it seems AST2 does not recognise that calls are coming from 2 different
accounts.

How can i make asterisk realize it?

-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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Re: [asterisk-users] TDM400P dialout problem

2008-02-26 Thread John covici

I do have the newer one installed -- is this the problem?
And why this weird error anyway?

on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
 > On Tue, Feb 26, 2008 at 07:50:09AM -0500, John covici wrote:
 > > I am getting this strange error:
 > > 
 > > make[1]: Entering directory `/usr/src/zaptel-1.4.7.1'
 > > make -C /lib/modules/2.6.24-gentoo-r2/build 
 > > SUBDIRS=/usr/src/zaptel-1.4.7.1 HOTPLUG_FIRMWARE=yes modules
 > > make[2]: Entering directory `/usr/src/linux-2.6.24-gentoo-r2'
 > >   CC [M]  /usr/src/zaptel-1.4.7.1/wcfxo.o
 > > /usr/src/zaptel-1.4.7.1/wcfxo.c:38:27: error: zaptel/zaptel.h: No such 
 > > file or directory
 > > 
 > > But the file is there.
 > > At least its in the source directory.
 > > I did the ./configure and the make menuselect to eliminate some
 > > unnecessary modules.
 > 
 > -DSTANDALONE_ZAPATA not getting through to the EXTRA_CFLAGS?
 > 
 > Note: luckily you didn't have zaptel installed and this it didn't use a
 > different version of zaptel.h from /usr/include/zaptel/zaptel.h .
 > 
 > -- 
 >Tzafrir Cohen
 > icq#16849755  jabber:[EMAIL PROTECTED]
 > +972-50-7952406   mailto:[EMAIL PROTECTED]
 > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
 > 
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 > To UNSUBSCRIBE or update options visit:
 >http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]

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Re: [asterisk-users] [URGENT] Zap channels fail to load

2008-02-26 Thread Louwrens Benadé

Why does this look suspiciously like a T1 line? Are you sure this is a
fractional E1?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andres Jimenez
Sent: 26 February 2008 02:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [URGENT] Zap channels fail to load

On Tue, Feb 26, 2008 at 12:05 PM, Louwrens Benadé <[EMAIL PROTECTED]>
wrote:
> What's your output from 'ztcfg -vv'?

pbx:~# ztcfg -vv

Zaptel Version: 1.4.8
Echo Canceller: MG2
Configuration
==

SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 18)
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: Clear channel (Default) (Slaves: 24)

24 channels to configure.



-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
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Re: [asterisk-users] iax trunking problem

2008-02-26 Thread love U . all


thx guys , i think i discovered the problem , it seems that i had to put the 
host=192.168.0.x in iax.conf and not host=dynamic ,,otherwise had to register 
the clients 


From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 26 Feb 2008 15:03:25 
+0200Subject: [asterisk-users] iax trunking problem


i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX 
trunking between the 2 servers so that i dial -say from a sip extension 2000 on 
fedora server to a sip extension 3000 on CentOS server the call seems to be 
established but hangup automatically after very short time and here is the iax2 
set debug command result on centos server and also my iax.conf and 
extension.conf and sip.conf files 
:**;the
 iax2 debug output Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX 
Subclass: NEW   Timestamp: 4ms  SCall: 1  DCall: 0 
[192.168.0.25:4569]   VERSION : 2   CALLED NUMBER   : 3000   
CODEC_PREFS : ()   CALLING NUMBER  : 2000   CALLING PRESNTN : 0   CALLING 
TYPEOFN : 0   CALLING TRANSIT : 0   CALLING NAME: Gres   LANGUAGE: 
en   CALLED CONTEXT  : centos-context   USERNAME: iax1Centos   FORMAT   
   : 4   CAPABILITY  : 65535   ADSICPE : 2   DATE TIME   : 
2008-02-26  01:50:30Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX
 Subclass: AUTHREQ   Timestamp: 00010ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]   AUTHMETHODS : 3   CHALLENGE   : 667712244   
USERNAME: iax1CentosRx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 
Type: IAX Subclass: AUTHREP   Timestamp: 00100ms  SCall: 1  DCall: 
1 [192.168.0.25:4569]   MD5 RESULT  : 8cbf698b6ea75a404e8ae981fdec0a16  
  -- Accepting AUTHENTICATED call from 192.168.0.25:   > requested format = 
ulaw,   > requested prefs = (),   > actual format = ulaw,   > host 
prefs = (),   > priority = mine-- Executing [EMAIL PROTECTED]:1] 
Set('IAX2/iax1Centos-1', 'TIMEOUT(absolute)=1') in new stack-- Channel 
will hangup at 2008-02-26 19:01:22 UTC.-- Executing [EMAIL PROTECTED]:2] 
Dial('IAX2/iax1Centos-1', 'sip/3000/20') in new stackTx-Frame Retry[000] -- 
OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPTTimestamp: 00117ms  
SCall: 1  DCall: 1 [192.168.0.25:4569]   FORMAT  : 4[Feb 26 
18:14:42] WARNING[4581]: app_dial.c:1196 dial_exec_full: Unable to create 
channel of type 'sip' (cause 3 - No route to destination)  == Everyone is 
busy/congested at this time (1:0/0/1)-- Executing [EMAIL PROTECTED]:3] 
VoiceMail('IAX2/iax1Centos-1', '[EMAIL PROTECTED]') in new stack[Feb 26 
18:14:42] WARNING[4581]: app_voicemail.c:2850 leave_voicemail: No entry in 
voicemail config file for '3000'  == Auto fallthrough, channel 
'IAX2/iax1Centos-1' status is 'CHANUNAVAIL'Rx-Frame Retry[ No] -- OSeqno: 002 
ISeqno: 002 Type: IAX Subclass: ACK   Timestamp: 00117ms  SCall: 1  
DCall: 1 [192.168.0.25:4569]Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 
Type: CONTROL Subclass: ANSWERTimestamp: 00120ms  SCall: 1  DCall: 
1 [192.168.0.25:4569]Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 002 Type: 
CONTROL Subclass: CONGSTN   Timestamp: 00123ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: VOICE   
Subclass: 4   Timestamp: 00354ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX 
Subclass: ACK   Timestamp: 00354ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX 
Subclass: ACK   Timestamp: 00120ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 004 Type: IAX 
Subclass: ACK   Timestamp: 00123ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: IAX 
Subclass: HANGUPTimestamp: 00457ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]   CAUSE CODE  : 16Tx-Frame Retry[-01] -- OSeqno: 004 
ISeqno: 004 Type: IAX Subclass: ACK   Timestamp: 00457ms  SCall: 1  
DCall: 1 [192.168.0.25:4569]-- Hungup 'IAX2/iax1Centos-1'CentOS*CLI> 
*;CentOS
 server iax.conf[general]bindport=4569bindaddr=192.168.0.22 ; this is Centos 
server address 
[iax1Centos]type=friendsecret=volgausername=iax1Centoscontext=centos-contexthost=dynamictrunk=yes;Centos
 server extension.conf[centos-context]exten 
=>3000,1,Set(TIMEOUT(absolute)=1) exten =>3000,2,dial(sip/3000/20)exten 
=>3000,3,voicemail([EMAIL PROTECTED])exten=> _2XXX,1,Answer()exten=> 
_2XXX,2,Dial(IAX2/iax1Fedora:[EMAIL PROTECTED]/[EMAIL PROTECTED])exten=> 
_2XXX,3,Hangup()0--; centos sip.conf 
[general]por

Re: [asterisk-users] TDM400P dialout problem

2008-02-26 Thread Tzafrir Cohen

On Tue, Feb 26, 2008 at 07:50:09AM -0500, John covici wrote:
> I am getting this strange error:
> 
> make[1]: Entering directory `/usr/src/zaptel-1.4.7.1'
> make -C /lib/modules/2.6.24-gentoo-r2/build SUBDIRS=/usr/src/zaptel-1.4.7.1 
> HOTPLUG_FIRMWARE=yes modules
> make[2]: Entering directory `/usr/src/linux-2.6.24-gentoo-r2'
>   CC [M]  /usr/src/zaptel-1.4.7.1/wcfxo.o
> /usr/src/zaptel-1.4.7.1/wcfxo.c:38:27: error: zaptel/zaptel.h: No such file 
> or directory
> 
> But the file is there.
> At least its in the source directory.
> I did the ./configure and the make menuselect to eliminate some
> unnecessary modules.

-DSTANDALONE_ZAPATA not getting through to the EXTRA_CFLAGS?

Note: luckily you didn't have zaptel installed and this it didn't use a
different version of zaptel.h from /usr/include/zaptel/zaptel.h .

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

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Re: [asterisk-users] FXO Cards - T38

2008-02-26 Thread zoa

Fernando Berretta wrote:
> Tzafir,
>
> I'm sorry, my question wasn't clear.
>
> Apparently Asterisk 1.6.0b2 and b4 has support for t38 because of some 
> modifications on app_fax so the questions are:
>
> 1 - If I use Asterisk 1.6.0b2 o b4 and a fax is received from a FXO 
> Card and this FXO port is forwarded to other ATA/Gateway is asterisk 
> going to transmit this fax using t38 ?
> PSTN FAX MACHINEASTERISK(1.6.0b2) FXO 
> CARD---t38?ATA/Gateway-FAX 
> MACHINE
>
No this is not going to work with the code you find in add-ons (Steve 
Underwood was right, my last email was a bit vague).
The FAX -> ASTERISK -> t.38 part will not work.

2 - If the first answer is yes, if we compile app_fax with asterisk 1.4x 
same behavior could be achieved ?
>
> Regards,
> Fernando
>
> Tzafrir Cohen wrote:
>> On Mon, Feb 25, 2008 at 05:32:24PM -0300, Fernando Berretta wrote:
>>   
>>> Dear All,
>>>
>>> Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax etc. 
>>> and will be able to receive faxes and negotiate with voip CPE's like 
>>> ATA's to transmit faxes which comes from FXO cards to VoIP Devices using 
>>> T38 ? it is possible to compile this version of app_fax to work with 
>>> Asterisk 1.4x ? Someone has tried it ?
>>> 
>> You have rx_fax for 1.4 . You also have fax detection in chan_zap, and
>> thus can send faxes from the PSTN to rx_fax.
>>
>> Not exactly the same, but maybe this is actually what you're looking
>> for.
>>
>>   
>
> 
>
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Re: [asterisk-users] FXO Cards - T38

2008-02-26 Thread Fernando Berretta

Thanks for clarify.. so Asterisk will be able to receive faxes which 
comes from a Gateway using t38 but will not be able to relay faxes which 
comes from PSTN through a FXO card to other Gateway using t38


can this version of app_fax be used with Asterisk 1.4x ?


Steve Underwood wrote:

zoa wrote:
  

T.38 will not work with the fxo card.

Zoa
  

That statement is a bit vague. What has been put in add-ons so far is 
only support for T.38 termination. Not T.38 gateway operation.


Steve

  

Fernando Berretta wrote:
  


Dear All,

Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax 
etc. and will be able to receive faxes and negotiate with voip CPE's 
like ATA's to transmit faxes which comes from FXO cards to VoIP 
Devices using T38 ? it is possible to compile this version of app_fax 
to work with Asterisk 1.4x ? Someone has tried it ?


Best Regards,
Fernando

Thomas Kenyon wrote:

  

Steve Underwood wrote:
  
  

  
  
  

I thought * was still not capable for T.38 gateway operation. Doesn't 
beta 4 just added T.38 termination? And, I believe it misses out some 
key elements of doing that properly. Note that T.38 termination is an 
addon, so it can't be used with, say, G.729.


  
The only real option available at the moment is to keep one PSTN line on 
an ATA with an FXO port and T.38 support available and direct calls from 
the fax machines through to it.  However, I should point out that while 
I believe this should be possible, I haven't actually tried it myself.


  
  
  

The new asterisk T.38 functionality is from the Asterisk addons 1.6.0b2 
version of app_fax (and a few small changes in 1.6.0b4), which I thought 
someone would have mentioned to you, since it does use spandsp.


(Or at least the configure script checks for spandsp, I haven't actually 
looked at the code).


  




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[asterisk-users] iax trunking problem

2008-02-26 Thread love U . all


i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX 
trunking between the 2 servers so that i dial -say from a sip extension 2000 on 
fedora server to a sip extension 3000 on CentOS server the call seems to be 
established but hangup automatically after very short time and here is the iax2 
set debug command result on centos server and also my iax.conf and 
extension.conf and sip.conf files 
:**;the
 iax2 debug output Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX 
Subclass: NEW   Timestamp: 4ms  SCall: 1  DCall: 0 
[192.168.0.25:4569]   VERSION : 2   CALLED NUMBER   : 3000   
CODEC_PREFS : ()   CALLING NUMBER  : 2000   CALLING PRESNTN : 0   CALLING 
TYPEOFN : 0   CALLING TRANSIT : 0   CALLING NAME: Gres   LANGUAGE: 
en   CALLED CONTEXT  : centos-context   USERNAME: iax1Centos   FORMAT   
   : 4   CAPABILITY  : 65535   ADSICPE : 2   DATE TIME   : 
2008-02-26  01:50:30Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX
 Subclass: AUTHREQ   Timestamp: 00010ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]   AUTHMETHODS : 3   CHALLENGE   : 667712244   
USERNAME: iax1CentosRx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 
Type: IAX Subclass: AUTHREP   Timestamp: 00100ms  SCall: 1  DCall: 
1 [192.168.0.25:4569]   MD5 RESULT  : 8cbf698b6ea75a404e8ae981fdec0a16  
  -- Accepting AUTHENTICATED call from 192.168.0.25:   > requested format = 
ulaw,   > requested prefs = (),   > actual format = ulaw,   > host 
prefs = (),   > priority = mine-- Executing [EMAIL PROTECTED]:1] 
Set('IAX2/iax1Centos-1', 'TIMEOUT(absolute)=1') in new stack-- Channel 
will hangup at 2008-02-26 19:01:22 UTC.-- Executing [EMAIL PROTECTED]:2] 
Dial('IAX2/iax1Centos-1', 'sip/3000/20') in new stackTx-Frame Retry[000] -- 
OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPTTimestamp: 00117ms  
SCall: 1  DCall: 1 [192.168.0.25:4569]   FORMAT  : 4[Feb 26 
18:14:42] WARNING[4581]: app_dial.c:1196 dial_exec_full: Unable to create 
channel of type 'sip' (cause 3 - No route to destination)  == Everyone is 
busy/congested at this time (1:0/0/1)-- Executing [EMAIL PROTECTED]:3] 
VoiceMail('IAX2/iax1Centos-1', '[EMAIL PROTECTED]') in new stack[Feb 26 
18:14:42] WARNING[4581]: app_voicemail.c:2850 leave_voicemail: No entry in 
voicemail config file for '3000'  == Auto fallthrough, channel 
'IAX2/iax1Centos-1' status is 'CHANUNAVAIL'Rx-Frame Retry[ No] -- OSeqno: 002 
ISeqno: 002 Type: IAX Subclass: ACK   Timestamp: 00117ms  SCall: 1  
DCall: 1 [192.168.0.25:4569]Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 
Type: CONTROL Subclass: ANSWERTimestamp: 00120ms  SCall: 1  DCall: 
1 [192.168.0.25:4569]Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 002 Type: 
CONTROL Subclass: CONGSTN   Timestamp: 00123ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: VOICE   
Subclass: 4   Timestamp: 00354ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX 
Subclass: ACK   Timestamp: 00354ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX 
Subclass: ACK   Timestamp: 00120ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 004 Type: IAX 
Subclass: ACK   Timestamp: 00123ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: IAX 
Subclass: HANGUPTimestamp: 00457ms  SCall: 1  DCall: 1 
[192.168.0.25:4569]   CAUSE CODE  : 16Tx-Frame Retry[-01] -- OSeqno: 004 
ISeqno: 004 Type: IAX Subclass: ACK   Timestamp: 00457ms  SCall: 1  
DCall: 1 [192.168.0.25:4569]-- Hungup 'IAX2/iax1Centos-1'CentOS*CLI> 
*;CentOS
 server iax.conf[general]bindport=4569bindaddr=192.168.0.22 ; this is Centos 
server address 
[iax1Centos]type=friendsecret=volgausername=iax1Centoscontext=centos-contexthost=dynamictrunk=yes;Centos
 server extension.conf[centos-context]exten 
=>3000,1,Set(TIMEOUT(absolute)=1) exten =>3000,2,dial(sip/3000/20)exten 
=>3000,3,voicemail([EMAIL PROTECTED])exten=> _2XXX,1,Answer()exten=> 
_2XXX,2,Dial(IAX2/iax1Fedora:[EMAIL PROTECTED]/[EMAIL PROTECTED])exten=> 
_2XXX,3,Hangup()0--; centos sip.conf 
[general]port=5060bindaddr=0.0.0.0[3000]type=friendsecret=3000qualify=yescall-limit=1
   ;limit No of calls this exten can originat in same time;canreinvite=yes  
;can bypass asteris control in future 
calleshost=dynamiccontext=centos-contextmailbox=3000disallow=allallow=ulawallow=alawallow=gsm--

Re: [asterisk-users] TDM400P dialout problem

2008-02-26 Thread John covici

I am getting this strange error:

make[1]: Entering directory `/usr/src/zaptel-1.4.7.1'
make -C /lib/modules/2.6.24-gentoo-r2/build SUBDIRS=/usr/src/zaptel-1.4.7.1 
HOTPLUG_FIRMWARE=yes modules
make[2]: Entering directory `/usr/src/linux-2.6.24-gentoo-r2'
  CC [M]  /usr/src/zaptel-1.4.7.1/wcfxo.o
/usr/src/zaptel-1.4.7.1/wcfxo.c:38:27: error: zaptel/zaptel.h: No such file or 
directory

But the file is there.
At least its in the source directory.
I did the ./configure and the make menuselect to eliminate some
unnecessary modules.


on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
 > On Tue, Feb 26, 2008 at 06:45:15AM -0500, John covici wrote:
 > > Hi.  Since I am stuck with kernel 2.6.24, is there any way to compile
 > > zaptel 1.4.7.1 under kernel 2.6.24?  I tried using
 > > make KBUILD_NOPEDANTIC=1 -- however this does not compile.  Any other
 > > suggestions for this and can I still use the latest version of
 > > asterisk if I do this successfully?
 > 
 > What error(s) do you get? Later on I fixed a number of build problems 
 > with ztd-eth.c . But you can probably skip that module altogether if 
 > you don't need TDM over Ethernet.
 > 
 > -- 
 >Tzafrir Cohen
 > icq#16849755  jabber:[EMAIL PROTECTED]
 > +972-50-7952406   mailto:[EMAIL PROTECTED]
 > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
 > 
 > ___
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 > To UNSUBSCRIBE or update options visit:
 >http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]

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Re: [asterisk-users] FXO Cards - T38

2008-02-26 Thread Fernando Berretta


Tzafir,

I'm sorry, my question wasn't clear.

Apparently Asterisk 1.6.0b2 and b4 has support for t38 because of some 
modifications on app_fax so the questions are:


1 - If I use Asterisk 1.6.0b2 o b4 and a fax is received from a FXO Card 
and this FXO port is forwarded to other ATA/Gateway is asterisk going to 
transmit this fax using t38 ?
PSTN FAX MACHINEASTERISK(1.6.0b2) FXO 
CARD---t38?ATA/Gateway-FAX 
MACHINE


2 - If the first answer is yes, if we compile app_fax with asterisk 1.4x 
same behavior could be achieved ?


Regards,
Fernando

Tzafrir Cohen wrote:

On Mon, Feb 25, 2008 at 05:32:24PM -0300, Fernando Berretta wrote:
  

Dear All,

Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax etc. 
and will be able to receive faxes and negotiate with voip CPE's like 
ATA's to transmit faxes which comes from FXO cards to VoIP Devices using 
T38 ? it is possible to compile this version of app_fax to work with 
Asterisk 1.4x ? Someone has tried it ?



You have rx_fax for 1.4 . You also have fax detection in chan_zap, and
thus can send faxes from the PSTN to rx_fax.

Not exactly the same, but maybe this is actually what you're looking
for.

  


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Re: [asterisk-users] [URGENT] Zap channels fail to load

2008-02-26 Thread Andres Jimenez

On Tue, Feb 26, 2008 at 12:05 PM, Louwrens Benadé <[EMAIL PROTECTED]> wrote:
> What's your output from 'ztcfg -vv'?

pbx:~# ztcfg -vv

Zaptel Version: 1.4.8
Echo Canceller: MG2
Configuration
==

SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 18)
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: Clear channel (Default) (Slaves: 24)

24 channels to configure.



-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
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Re: [asterisk-users] Still can't pickup parked call

2008-02-26 Thread OCG Technical Support

Well, I don't have a 701 extension defined but I do have _XXX which is where
this call is jumping when I dial 701 to pickup.

I have the "include => parkedcalls" above the _XXX definition, so I assumed
that parked calls would be matched first.  As well, since the 700 is
matching to parked calls, I assumed 701 would as well.  

Still stuck 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Lacy Moore
> Sent: Tuesday, February 26, 2008 2:30 AM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] Still can't pickup parked call
> 
> I suspect there is something you are not telling us.  Try 
> posting this extension.conf file.  Looking at the logs you 
> have here leads me to believe you have an extension 701 
> defined to dial SIP/233.
> 
> In other words, somewhere in your context is:
> 
> exten => 701,1,Dial(SIP/233)
> 
> or something very similar.
> 
> An included context will never (ok, most probably won't ever) 
> overwrite the definitions in the current context.  For 
> example, if you define extension 100 in your main context and 
> then define extension 100 in an included context, the one in 
> the main context will most probably always prevail.
> 
> 
> 
> On Mon, Feb 25, 2008 at 9:41 PM, OCG Technical Support 
> <[EMAIL PROTECTED]> wrote:
> > I'm still struggling to pickup calls.  I now have a single context
> > (entryocginternal) where I have "include => parkedcalls".
> >
> > The log below shows me calling from one internal extension 
> to another, 
> > then picking up, then parking the call.
> >
> >-- SIP/239-0915d5c8 is ringing
> >-- SIP/239-0915d5c8 answered SIP/233-0915bf40
> >-- Packet2Packet bridging SIP/233-0915bf40 and SIP/239-0915d5c8
> >-- Started music on hold, class 'default', on 
> SIP/239-0915d5c8  == 
> > Spawn extension (macro-dialinternal, s, 7) exited non-zero on 
> > 'SIPPeer/SIP/233-0915bf40' in macro 'dialinternal'
> >  == Spawn extension (macro-dialinternal, s, 7) exited non-zero on 
> > 'SIPPeer/SIP/233-0915bf40'
> >-- Started music on hold, class 'default', on 
> SIP/239-0915d5c8  == 
> > Parked SIP/239-0915d5c8 on [EMAIL PROTECTED] Will timeout back to 
> > extension [entryocginternal] , 1 in 300 seconds
> >--  Playing 'digits/7' (language 'en')
> >--  Playing 'digits/0' (language 'en')
> >--  Playing 'digits/1' (language 'en')
> >-- Added extension '701' priority 1 to parkedcalls
> >
> > After parking the call, I then used that same phone to 
> pickup 701 by 
> > dialing 701.  As you can see, the 701 is being treated as a 
> regular extension - not
> > a parked call pickup.   What is going on?  Why is this nor working?
> >
> >-- Executing [EMAIL PROTECTED]:1] Macro("SIP/233-09152818",
> > "dialexternal|701|") in new stack
> >
> >
> > ___
> > -- Bandwidth and Colocation Provided by 
> http://www.api-digital.com --
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> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> 
> --
> Lacy Moore
> Somewhere I wish I wasn't
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] FXO Cards - T38

2008-02-26 Thread Steve Underwood

Benny Amorsen wrote:
> Steve Underwood <[EMAIL PROTECTED]> writes:
>
>   
>> Try reading the GPL and the FSF's interpretation of it. If things are 
>> running in the same address space as my code, they need to be GPL 
>> compatible, or I am likely to take action.
>> 
>
> The GPL is not an EULA. You don't have to agree to it to use the
> software, only to distribute it.
>   
This is the key drawback of GPL 2 for my purposes. You can indeed do 
whatever you want with my GPL code internally. Supply it to anyone as 
something mingled with non-GPL compatible code, though, and you are in 
violation of the licence. So, only in house use is OK.

Regards,
Steve


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Re: [asterisk-users] FXO Cards - T38

2008-02-26 Thread Steve Underwood

zoa wrote:
> T.38 will not work with the fxo card.
>
> Zoa
>   
That statement is a bit vague. What has been put in add-ons so far is 
only support for T.38 termination. Not T.38 gateway operation.

Steve

> Fernando Berretta wrote:
>   
>> Dear All,
>>
>> Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax 
>> etc. and will be able to receive faxes and negotiate with voip CPE's 
>> like ATA's to transmit faxes which comes from FXO cards to VoIP 
>> Devices using T38 ? it is possible to compile this version of app_fax 
>> to work with Asterisk 1.4x ? Someone has tried it ?
>>
>> Best Regards,
>> Fernando
>>
>> Thomas Kenyon wrote:
>> 
>>> Steve Underwood wrote:
>>>   
>>>   
>   
>   
>   
 I thought * was still not capable for T.38 gateway operation. Doesn't 
 beta 4 just added T.38 termination? And, I believe it misses out some 
 key elements of doing that properly. Note that T.38 termination is an 
 addon, so it can't be used with, say, G.729.
 
 
> The only real option available at the moment is to keep one PSTN line on 
> an ATA with an FXO port and T.38 support available and direct calls from 
> the fax machines through to it.  However, I should point out that while 
> I believe this should be possible, I haven't actually tried it myself.
>
>   
>   
>   
>>> The new asterisk T.38 functionality is from the Asterisk addons 1.6.0b2 
>>> version of app_fax (and a few small changes in 1.6.0b4), which I thought 
>>> someone would have mentioned to you, since it does use spandsp.
>>>
>>> (Or at least the configure script checks for spandsp, I haven't actually 
>>> looked at the code).
>>>
>>>   


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Re: [asterisk-users] [URGENT] Zap channels fail to load

2008-02-26 Thread Andres Jimenez

Comes from a previous message.

On Tue, Feb 26, 2008 at 12:25 PM, Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:

>  Here's my guess:
>
>  You built Asterisk vs. a newer Zaptel (that happened to have the
>  Astribank drivers).
>
>  Now you reverted to the old Zaptel drivers. And those are of a version
>  before 1.4.8 . Hence the new ZT_GET_PARAMS of 1.4.8 does not exist
>  there. The ZT_GET_PARAMS ioctl Asterisk sends is thus not understood by
>  Zaptel and fails.
>
>  Unrevert to the new Zaptel version (of the modules. Stick with the
>  original zaptel.conf). Does this help?

I did build asterisk using zaptel 1.4.8 about 10 days ago. We were
having issues with DTMF , so I downgraded zaptel (rebuilding Asterisk
and libpri, of course) to 1.4.7, but the problems remained.
Anything else worked just fine, so I kept researching and no other
changes were done till today.
Zaptel 1.4.7 was working perfectly for almost 2 weeks and has the
Astribanks drivers too, so I just tried to add the Astribank to my
configuration today.

I wasn't being able to make it work, so I reverted the changes made
earlier and disconnected the Astribank trying to go back to the
previous known working config.

I have rebuild everything against Zaptel 1.4.8 and it works now, but I
am a little bit concerned about why asterisk tried to use the
functions belonging to zaptel 1.4.8.

-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]

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Re: [asterisk-users] [URGENT] Zap channels fail to load

2008-02-26 Thread Andres Jimenez

On Tue, Feb 26, 2008 at 12:21 PM, Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:
> Note: [Urgent] is generally not a good way to escalate the issue on a
>  public mailing list. We're all here for the fun of it and demanding
>  prompt reply may actually serve the other way.

I am sorry about the "scalating", but I was panicking a little bit
after a couple of hours trying to fix the issue.

>  If you have paid to get support (e.g: by buying hardware), this may be a
>  good time to use it.

I am too used to get/give free advice that I forget I can use it.

>  Contact me privately :-)

Nice to meet an "Astribank guru" . I will contact you soon :-)


>  The symptom is expelained in the following report that I filed earlier
>  today (unrelated to this one)
>
>   http://bugs.digium.com/12071
>
>  Channel 1 was left open from a failed configuration attempt
>
>  So the real error hides earlier in your logs. Look for 'chan_zap' in the
>  logs from the startup of Asterisk. And sadly you must restart Asterisk
>  to fix the error.

I din't fix it for me.

>  Is this a fractional E1, indeed?

Not sure about how to call it. The provider (Eircom, Ireland) calls it PRA.

This messages follows in the other message reply.


-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]

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Re: [asterisk-users] [URGENT] Zap channels fail to load

2008-02-26 Thread Louwrens Benadé

What's your output from 'ztcfg -vv'?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andres Jimenez
Sent: 26 February 2008 01:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [URGENT] Zap channels fail to load

I forgot to mentio asterisk log this 2 errors:

[Feb 26 08:38:01] ERROR[30245] chan_zap.c: Unable to get parameters
[Feb 26 08:38:01] ERROR[30245] chan_zap.c: Unable to register channel '1-15'

Any hint?

Thanks in advance.
Andres

On Tue, Feb 26, 2008 at 10:44 AM, Andres Jimenez <[EMAIL PROTECTED]>
wrote:
> I have spent some time this morning trying to add an Astribank to our
>  current Asterisk, but it failed, so I just removed the hardware,
>  restore the config files to the original setup and started asterisk.;
>
>  I could see that no Zap channels are started so I did load chan_zap.so:
>  pbx*CLI> module load chan_zap.so
>  [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application:
>  Already have an application 'ZapSendKeypadFacility'
>   == Parsing '/etc/asterisk/zapata.conf': Found
>  [Feb 26 10:32:18] WARNING[3809]: chan_zap.c:904 zt_open: Unable to
>  specify channel 1: Device or resource busy
>  [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:7186 mkintf: Unable to open
>  channel 1: Device or resource busy
>  here = 0, tmp->channel = 1, channel = 1
>  [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:10527 build_channels: Unable
>  to register channel '1-15'
>
>  Fair enough. I did unloaded chan_zap.so (because of the first error)
>  and tried again:
>
>  pbx*CLI> module load chan_zap.so
>  [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application:
>  Already have an application 'ZapSendKeypadFacility'
>   == Parsing '/etc/asterisk/zapata.conf': Found
>  [Feb 26 10:32:18] WARNING[3809]: chan_zap.c:904 zt_open: Unable to
>  specify channel 1: Device or resource busy
>  [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:7186 mkintf: Unable to open
>  channel 1: Device or resource busy
>  here = 0, tmp->channel = 1, channel = 1
>  [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:10527 build_channels: Unable
>  to register channel '1-15'
>
>  It looks like the problem is in the zap card's first channel:
>
>  pbx:~# cat /proc/zaptel/1
>  Span 1: WCT1/0 "Wildcard TE12xP Card 0" HDB3/CCS
> IRQ misses: 36
>
>1 WCT1/0/1 Clear (In use)
>2 WCT1/0/2 Clear
>3 WCT1/0/3 Clear
>4 WCT1/0/4 Clear
>5 WCT1/0/5 Clear
>6 WCT1/0/6 Clear
>7 WCT1/0/7 Clear
>8 WCT1/0/8 Clear
>9 WCT1/0/9 Clear
>   10 WCT1/0/10 Clear
>   11 WCT1/0/11 Clear
>   12 WCT1/0/12 Clear
>   13 WCT1/0/13 Clear
>   14 WCT1/0/14 Clear
>   15 WCT1/0/15 Clear
>   16 WCT1/0/16 HDLCFCS
>   17 WCT1/0/17 Clear
>   18 WCT1/0/18 Clear
>   19 WCT1/0/19 Clear
>   20 WCT1/0/20 Clear
>   21 WCT1/0/21 Clear
>   22 WCT1/0/22 Clear
>   23 WCT1/0/23 Clear
>   24 WCT1/0/24 Clear
>   25 WCT1/0/25
>   26 WCT1/0/26
>   27 WCT1/0/27
>   28 WCT1/0/28
>   29 WCT1/0/29
>   30 WCT1/0/30
>   31 WCT1/0/31
>
>
>  Is there any change any Astribank related stuff can be causing this?
>
>  I have ensured that no Astribanks modules are loaded and even rebooted
>  the box, but no success.
>
>
>  pbx:~# cat /etc/zaptel.conf
>  # CRC off
>  #
>  loadzone = uk
>  defaultzone = uk
>
>  span=1,1,0,ccs,hdb3
>  bchan=1-15
>  dchan=16
>  bchan=17-24
>
>  #or this with crc on
>  #
>  #loadzone = uk
>  #defaultzone = uk
>
>  #span=1,1,0,ccs,hdb3,crc4
>  #bchan=1-15
>  #dchan=16
>  #bchan=17-24
>
>  pbx:~# cat /etc/asterisk/zapata.conf
>  language=en
>  internationalprefix = 00
>  nationalprefix = 0
>  switchtype = euroisdn
>  pridialplan = local
>  priindication = outofband
>  usecallerid = yes
>  hidecallerid = no
>  callwaiting = yes
>  usecallingpres = yes
>  callwaitingcallerid = yes
>  threewaycalling = yes
>  transfer = yes
>  cancallforward = yes
>  callreturn = yes
>  group = 1
>  callgroup = 0
>  pickupgroup = 0
>  immediate = no
>  echotraining = yes
>  echocancel = yes
>  echocancelwhenbridged = no
>  facilityenable = yes
>  musiconhold = default
>  ;overlapdial = yes
>  overlapdial = no
>  immediate = no
>  txgain = -4.0
>  rxgain = -4.0
>  signalling = pri_cpe
>  channel => 1-15
>  ;channel => 17-32
>  channel => 17-24
>  ;toneduration=100
>  toneduration=300
>  ;relaxdtmf=yes
>
>
>
>  Thanks,
>
>  --
>  Andres Jimenez
>
>  GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
>



-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]

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Re: [asterisk-users] TDM400P dialout problem

2008-02-26 Thread Tzafrir Cohen

On Tue, Feb 26, 2008 at 06:45:15AM -0500, John covici wrote:
> Hi.  Since I am stuck with kernel 2.6.24, is there any way to compile
> zaptel 1.4.7.1 under kernel 2.6.24?  I tried using
> make KBUILD_NOPEDANTIC=1 -- however this does not compile.  Any other
> suggestions for this and can I still use the latest version of
> asterisk if I do this successfully?

What error(s) do you get? Later on I fixed a number of build problems 
with ztd-eth.c . But you can probably skip that module altogether if 
you don't need TDM over Ethernet.

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

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Re: [asterisk-users] TDM400P dialout problem

2008-02-26 Thread John covici

Hi.  Since I am stuck with kernel 2.6.24, is there any way to compile
zaptel 1.4.7.1 under kernel 2.6.24?  I tried using
make KBUILD_NOPEDANTIC=1 -- however this does not compile.  Any other
suggestions for this and can I still use the latest version of
asterisk if I do this successfully?

Thanks.

on Monday 02/25/2008 sean darcy([EMAIL PROTECTED]) wrote
 > On Mon, Feb 25, 2008 at 3:42 AM, Anthony Messina <[EMAIL PROTECTED]> wrote:
 > > Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble 
 > > dialing
 > >  out to the pstn. The call is initiated at Zap/1-1 and should exit via 
 > > Zap/3.
 > >  I get the following:
 > >
 > >  -- Starting simple switch on 'Zap/1-1'
 > >  -- Executing [EMAIL PROTECTED]:1] Dial("Zap/1-1", "Zap/3/8801234") in new 
 > > stack
 > >  [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing '8801234'
 > >  [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:2030 zt_call: Deferring 
 > > dialing...
 > >  -- Called 3/8801234
 > >  [Feb 25 02:37:00] WARNING[7194]: chan_zap.c:3835 zt_handle_event: Detected
 > >  alarm on channel 3: No Alarm
 > >  -- Hungup 'Zap/3-1'
 > >   == Everyone is busy/congested at this time (1:0/0/1)
 > >  [Feb 25 02:37:00] NOTICE[7082]: chan_zap.c:6678 handle_init_event: Alarm
 > >  cleared on channel 3
 > >
 > >  So the call fails and if I weren't using a test extension:
 > >  exten => 2111,1,Dial(Zap/3/8801234)
 > >
 > >  it would proceed in the dialplan.
 > >
 > >  asterisk]# cat /proc/zaptel/1
 > >  Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER)
 > >
 > >1 WCTDM/0/0 FXOKS (In use)
 > >2 WCTDM/0/1
 > >3 WCTDM/0/2 FXSKS (In use)
 > >4 WCTDM/0/3
 > >
 > >
 > >  Where do I go with this?
 > >
 > >  --
 > >  Anthony -  http://messinet.com - http://messinet.com/~amessina/gallery
 > >  8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
 > >
 > > ___
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 > >  asterisk-users mailing list
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 > >http://lists.digium.com/mailman/listinfo/asterisk-users
 > >
 > 
 > Look at http://bugs.digium.com/view.php?id=11855.
 > 
 > sean
 > 
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 >http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]

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[asterisk-users] CLIR missing in MySQL CDR records

2008-02-26 Thread Christian Victor

Hello!

I just encountered a strange thing in my mysql cdr records. From a
certain date on Asterisk (1.4.6) stopped to populate the CLIR and SCR
flieds in the cdr table. As far as I know no changes happened to the
system on that date and until then CLIR are recorded properly.

The CLIR is still transmitted by the PRI and is shown in the console
when a call comes in. But no traces of it in the CDR.

Did anybody of you ever experienced this?

Christian
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Re: [asterisk-users] [URGENT] Zap channels fail to load

2008-02-26 Thread Tzafrir Cohen

On Tue, Feb 26, 2008 at 11:06:29AM +, Andres Jimenez wrote:
> I forgot to mentio asterisk log this 2 errors:
> 
> [Feb 26 08:38:01] ERROR[30245] chan_zap.c: Unable to get parameters
> [Feb 26 08:38:01] ERROR[30245] chan_zap.c: Unable to register channel '1-15'
> 
> Any hint?

Here's my guess:

You built Asterisk vs. a newer Zaptel (that happened to have the
Astribank drivers).

Now you reverted to the old Zaptel drivers. And those are of a version
before 1.4.8 . Hence the new ZT_GET_PARAMS of 1.4.8 does not exist
there. The ZT_GET_PARAMS ioctl Asterisk sends is thus not understood by
Zaptel and fails.

Unrevert to the new Zaptel version (of the modules. Stick with the
original zaptel.conf). Does this help?

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

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Re: [asterisk-users] [URGENT] Zap channels fail to load

2008-02-26 Thread Tzafrir Cohen

Note: [Urgent] is generally not a good way to escalate the issue on a
public mailing list. We're all here for the fun of it and demanding
prompt reply may actually serve the other way.

If you have paid to get support (e.g: by buying hardware), this may be a
good time to use it.

That said, your message looks interesting. So see my reply inline :-)

On Tue, Feb 26, 2008 at 10:44:27AM +, Andres Jimenez wrote:
> I have spent some time this morning trying to add an Astribank to our
> current Asterisk, but it failed, 

Contact me privately :-)

> so I just removed the hardware,
> restore the config files to the original setup and started asterisk.;
> 
> I could see that no Zap channels are started so I did load chan_zap.so:
> pbx*CLI> module load chan_zap.so

You probably needed 'module unload chan_zap.so' before that. The module
was loaded, but it has bailed out before registering the CLI commands.

> [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application:
> Already have an application 'ZapSendKeypadFacility'
>   == Parsing '/etc/asterisk/zapata.conf': Found
> [Feb 26 10:32:18] WARNING[3809]: chan_zap.c:904 zt_open: Unable to
> specify channel 1: Device or resource busy
> [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:7186 mkintf: Unable to open
> channel 1: Device or resource busy
> here = 0, tmp->channel = 1, channel = 1
> [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:10527 build_channels: Unable
> to register channel '1-15'
> 
> Fair enough. I did unloaded chan_zap.so (because of the first error)
> and tried again:
> 
> pbx*CLI> module load chan_zap.so
> [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application:
> Already have an application 'ZapSendKeypadFacility'
>   == Parsing '/etc/asterisk/zapata.conf': Found
> [Feb 26 10:32:18] WARNING[3809]: chan_zap.c:904 zt_open: Unable to
> specify channel 1: Device or resource busy
> [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:7186 mkintf: Unable to open
> channel 1: Device or resource busy
> here = 0, tmp->channel = 1, channel = 1
> [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:10527 build_channels: Unable
> to register channel '1-15'
> 
> It looks like the problem is in the zap card's first channel:
> 
> pbx:~# cat /proc/zaptel/1
> Span 1: WCT1/0 "Wildcard TE12xP Card 0" HDB3/CCS
> IRQ misses: 36
> 
>1 WCT1/0/1 Clear (In use)
>2 WCT1/0/2 Clear

The symptom is expelained in the following report that I filed earlier
today (unrelated to this one)

  http://bugs.digium.com/12071

Channel 1 was left open from a failed configuration attempt

So the real error hides earlier in your logs. Look for 'chan_zap' in the
logs from the startup of Asterisk. And sadly you must restart Asterisk
to fix the error.

>3 WCT1/0/3 Clear
>4 WCT1/0/4 Clear
>5 WCT1/0/5 Clear
>6 WCT1/0/6 Clear
>7 WCT1/0/7 Clear
>8 WCT1/0/8 Clear
>9 WCT1/0/9 Clear
>   10 WCT1/0/10 Clear
>   11 WCT1/0/11 Clear
>   12 WCT1/0/12 Clear
>   13 WCT1/0/13 Clear
>   14 WCT1/0/14 Clear
>   15 WCT1/0/15 Clear
>   16 WCT1/0/16 HDLCFCS
>   17 WCT1/0/17 Clear
>   18 WCT1/0/18 Clear
>   19 WCT1/0/19 Clear
>   20 WCT1/0/20 Clear
>   21 WCT1/0/21 Clear
>   22 WCT1/0/22 Clear
>   23 WCT1/0/23 Clear
>   24 WCT1/0/24 Clear

Is this a fractional E1, indeed?

Or is the range in zaptel.conf incorrect? I noticed in your config file
that you changed it. Was this change made after Asterisk was started?

>   25 WCT1/0/25
>   26 WCT1/0/26
>   27 WCT1/0/27
>   28 WCT1/0/28
>   29 WCT1/0/29
>   30 WCT1/0/30
>   31 WCT1/0/31
> 
> 
> Is there any change any Astribank related stuff can be causing this?
> 
> I have ensured that no Astribanks modules are loaded and even rebooted
> the box, but no success.

I'm not sure how an Astribank module would have affected this one. The
Astribank spans tend to be the last ones, hence it should not have
misplaced your hardware.

If the module is loaded but no hardware is present, your system should
not be affected.

> 
> 
> pbx:~# cat /etc/zaptel.conf
> # CRC off
> #
> loadzone = uk
> defaultzone = uk
> 
> span=1,1,0,ccs,hdb3
> bchan=1-15
> dchan=16
> bchan=17-24
> 
> #or this with crc on
> #
> #loadzone = uk
> #defaultzone = uk
> 
> #span=1,1,0,ccs,hdb3,crc4
> #bchan=1-15
> #dchan=16
> #bchan=17-24
> 
> pbx:~# cat /etc/asterisk/zapata.conf
> language=en
> internationalprefix = 00
> nationalprefix = 0
> switchtype = euroisdn
> pridialplan = local
> priindication = outofband
> usecallerid = yes
> hidecallerid = no
> callwaiting = yes
> usecallingpres = yes
> callwaitingcallerid = yes
> threewaycalling = yes
> transfer = yes
> cancallforward = yes
> callreturn = yes
> group = 1
> callgroup = 0
> pickupgroup = 0
> immediate = no
> echotraining = yes
> echocancel = yes
> echocancelwhenbridged = no
> faci

Re: [asterisk-users] mfcr2 stuck

2008-02-26 Thread Jakub "Arkon" Syrek

Now it takes about 25 seconds after dialing number to make asterisk ready to 
answer call with Unicall. I think that this stuck is because of timeout in 
ANI request.
Here is my  log.
[EMAIL PROTECTED] ~]# cat /var/log/asterisk/full | grep unicall

[Feb 26 11:18:40] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 
0001  [1/IDLE/Idle  /Idle ]
[Feb 26 11:18:40] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 Detected
[Feb 26 11:18:40] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 Creating a 
new call with CRN 32770
[Feb 26 11:18:40] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 1101  -> 
[2/DETECTED/Seize ack /Seize ack]
[Feb 26 11:18:40] NOTICE[2968] chan_unicall.c: Unicall/20 event Detected
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 4 
on  [2/DETECTED/Seize ack /Seize ack]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 on  -> 
[2/DETECTED/Group A   /Category req ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 4 
off [2/DETECTED/Group A   /Category req ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 off -> 
[2/DETECTED/Group A   /Category req ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 1 
on  [2/DETECTED/Group A   /Category req ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 on  -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 1 
off [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 off -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 3 
on  [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 on  -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 3 
off [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 off -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 2 
on  [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 on  -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 2 
off [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 off -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 2 
on  [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 on  -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 2 
off [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 off -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 5 
on  [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 on  -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 5 
off [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 off -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 2 
on  [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 on  -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 2 
off [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 off -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 1 
on  [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 on  -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 1 
off [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:53] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 off -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:54] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 1 
on  [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:54] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20 5 on  -> 
[2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:54] WARNING[2968] chan_unicall.c: MFC/R2 UniCall/20  <- 1 
off [2/DETECTED/Group A   /ANI request  ]
[Feb 26 11:18:54] WARNING[2968] chan_un

Re: [asterisk-users] [URGENT] Zap channels fail to load

2008-02-26 Thread Andres Jimenez

I forgot to mentio asterisk log this 2 errors:

[Feb 26 08:38:01] ERROR[30245] chan_zap.c: Unable to get parameters
[Feb 26 08:38:01] ERROR[30245] chan_zap.c: Unable to register channel '1-15'

Any hint?

Thanks in advance.
Andres

On Tue, Feb 26, 2008 at 10:44 AM, Andres Jimenez <[EMAIL PROTECTED]> wrote:
> I have spent some time this morning trying to add an Astribank to our
>  current Asterisk, but it failed, so I just removed the hardware,
>  restore the config files to the original setup and started asterisk.;
>
>  I could see that no Zap channels are started so I did load chan_zap.so:
>  pbx*CLI> module load chan_zap.so
>  [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application:
>  Already have an application 'ZapSendKeypadFacility'
>   == Parsing '/etc/asterisk/zapata.conf': Found
>  [Feb 26 10:32:18] WARNING[3809]: chan_zap.c:904 zt_open: Unable to
>  specify channel 1: Device or resource busy
>  [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:7186 mkintf: Unable to open
>  channel 1: Device or resource busy
>  here = 0, tmp->channel = 1, channel = 1
>  [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:10527 build_channels: Unable
>  to register channel '1-15'
>
>  Fair enough. I did unloaded chan_zap.so (because of the first error)
>  and tried again:
>
>  pbx*CLI> module load chan_zap.so
>  [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application:
>  Already have an application 'ZapSendKeypadFacility'
>   == Parsing '/etc/asterisk/zapata.conf': Found
>  [Feb 26 10:32:18] WARNING[3809]: chan_zap.c:904 zt_open: Unable to
>  specify channel 1: Device or resource busy
>  [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:7186 mkintf: Unable to open
>  channel 1: Device or resource busy
>  here = 0, tmp->channel = 1, channel = 1
>  [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:10527 build_channels: Unable
>  to register channel '1-15'
>
>  It looks like the problem is in the zap card's first channel:
>
>  pbx:~# cat /proc/zaptel/1
>  Span 1: WCT1/0 "Wildcard TE12xP Card 0" HDB3/CCS
> IRQ misses: 36
>
>1 WCT1/0/1 Clear (In use)
>2 WCT1/0/2 Clear
>3 WCT1/0/3 Clear
>4 WCT1/0/4 Clear
>5 WCT1/0/5 Clear
>6 WCT1/0/6 Clear
>7 WCT1/0/7 Clear
>8 WCT1/0/8 Clear
>9 WCT1/0/9 Clear
>   10 WCT1/0/10 Clear
>   11 WCT1/0/11 Clear
>   12 WCT1/0/12 Clear
>   13 WCT1/0/13 Clear
>   14 WCT1/0/14 Clear
>   15 WCT1/0/15 Clear
>   16 WCT1/0/16 HDLCFCS
>   17 WCT1/0/17 Clear
>   18 WCT1/0/18 Clear
>   19 WCT1/0/19 Clear
>   20 WCT1/0/20 Clear
>   21 WCT1/0/21 Clear
>   22 WCT1/0/22 Clear
>   23 WCT1/0/23 Clear
>   24 WCT1/0/24 Clear
>   25 WCT1/0/25
>   26 WCT1/0/26
>   27 WCT1/0/27
>   28 WCT1/0/28
>   29 WCT1/0/29
>   30 WCT1/0/30
>   31 WCT1/0/31
>
>
>  Is there any change any Astribank related stuff can be causing this?
>
>  I have ensured that no Astribanks modules are loaded and even rebooted
>  the box, but no success.
>
>
>  pbx:~# cat /etc/zaptel.conf
>  # CRC off
>  #
>  loadzone = uk
>  defaultzone = uk
>
>  span=1,1,0,ccs,hdb3
>  bchan=1-15
>  dchan=16
>  bchan=17-24
>
>  #or this with crc on
>  #
>  #loadzone = uk
>  #defaultzone = uk
>
>  #span=1,1,0,ccs,hdb3,crc4
>  #bchan=1-15
>  #dchan=16
>  #bchan=17-24
>
>  pbx:~# cat /etc/asterisk/zapata.conf
>  language=en
>  internationalprefix = 00
>  nationalprefix = 0
>  switchtype = euroisdn
>  pridialplan = local
>  priindication = outofband
>  usecallerid = yes
>  hidecallerid = no
>  callwaiting = yes
>  usecallingpres = yes
>  callwaitingcallerid = yes
>  threewaycalling = yes
>  transfer = yes
>  cancallforward = yes
>  callreturn = yes
>  group = 1
>  callgroup = 0
>  pickupgroup = 0
>  immediate = no
>  echotraining = yes
>  echocancel = yes
>  echocancelwhenbridged = no
>  facilityenable = yes
>  musiconhold = default
>  ;overlapdial = yes
>  overlapdial = no
>  immediate = no
>  txgain = -4.0
>  rxgain = -4.0
>  signalling = pri_cpe
>  channel => 1-15
>  ;channel => 17-32
>  channel => 17-24
>  ;toneduration=100
>  toneduration=300
>  ;relaxdtmf=yes
>
>
>
>  Thanks,
>
>  --
>  Andres Jimenez
>
>  GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
>



-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]

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[asterisk-users] [URGENT] Zap channels fail to load

2008-02-26 Thread Andres Jimenez

I have spent some time this morning trying to add an Astribank to our
current Asterisk, but it failed, so I just removed the hardware,
restore the config files to the original setup and started asterisk.;

I could see that no Zap channels are started so I did load chan_zap.so:
pbx*CLI> module load chan_zap.so
[Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application:
Already have an application 'ZapSendKeypadFacility'
  == Parsing '/etc/asterisk/zapata.conf': Found
[Feb 26 10:32:18] WARNING[3809]: chan_zap.c:904 zt_open: Unable to
specify channel 1: Device or resource busy
[Feb 26 10:32:18] ERROR[3809]: chan_zap.c:7186 mkintf: Unable to open
channel 1: Device or resource busy
here = 0, tmp->channel = 1, channel = 1
[Feb 26 10:32:18] ERROR[3809]: chan_zap.c:10527 build_channels: Unable
to register channel '1-15'

Fair enough. I did unloaded chan_zap.so (because of the first error)
and tried again:

pbx*CLI> module load chan_zap.so
[Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application:
Already have an application 'ZapSendKeypadFacility'
  == Parsing '/etc/asterisk/zapata.conf': Found
[Feb 26 10:32:18] WARNING[3809]: chan_zap.c:904 zt_open: Unable to
specify channel 1: Device or resource busy
[Feb 26 10:32:18] ERROR[3809]: chan_zap.c:7186 mkintf: Unable to open
channel 1: Device or resource busy
here = 0, tmp->channel = 1, channel = 1
[Feb 26 10:32:18] ERROR[3809]: chan_zap.c:10527 build_channels: Unable
to register channel '1-15'

It looks like the problem is in the zap card's first channel:

pbx:~# cat /proc/zaptel/1
Span 1: WCT1/0 "Wildcard TE12xP Card 0" HDB3/CCS
IRQ misses: 36

   1 WCT1/0/1 Clear (In use)
   2 WCT1/0/2 Clear
   3 WCT1/0/3 Clear
   4 WCT1/0/4 Clear
   5 WCT1/0/5 Clear
   6 WCT1/0/6 Clear
   7 WCT1/0/7 Clear
   8 WCT1/0/8 Clear
   9 WCT1/0/9 Clear
  10 WCT1/0/10 Clear
  11 WCT1/0/11 Clear
  12 WCT1/0/12 Clear
  13 WCT1/0/13 Clear
  14 WCT1/0/14 Clear
  15 WCT1/0/15 Clear
  16 WCT1/0/16 HDLCFCS
  17 WCT1/0/17 Clear
  18 WCT1/0/18 Clear
  19 WCT1/0/19 Clear
  20 WCT1/0/20 Clear
  21 WCT1/0/21 Clear
  22 WCT1/0/22 Clear
  23 WCT1/0/23 Clear
  24 WCT1/0/24 Clear
  25 WCT1/0/25
  26 WCT1/0/26
  27 WCT1/0/27
  28 WCT1/0/28
  29 WCT1/0/29
  30 WCT1/0/30
  31 WCT1/0/31


Is there any change any Astribank related stuff can be causing this?

I have ensured that no Astribanks modules are loaded and even rebooted
the box, but no success.


pbx:~# cat /etc/zaptel.conf
# CRC off
#
loadzone = uk
defaultzone = uk

span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-24

#or this with crc on
#
#loadzone = uk
#defaultzone = uk

#span=1,1,0,ccs,hdb3,crc4
#bchan=1-15
#dchan=16
#bchan=17-24

pbx:~# cat /etc/asterisk/zapata.conf
language=en
internationalprefix = 00
nationalprefix = 0
switchtype = euroisdn
pridialplan = local
priindication = outofband
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
cancallforward = yes
callreturn = yes
group = 1
callgroup = 0
pickupgroup = 0
immediate = no
echotraining = yes
echocancel = yes
echocancelwhenbridged = no
facilityenable = yes
musiconhold = default
;overlapdial = yes
overlapdial = no
immediate = no
txgain = -4.0
rxgain = -4.0
signalling = pri_cpe
channel => 1-15
;channel => 17-32
channel => 17-24
;toneduration=100
toneduration=300
;relaxdtmf=yes



Thanks,

-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]

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Re: [asterisk-users] FW: jabber

2008-02-26 Thread Philippe Sultan

Hi Clive,

> Hi all,
> Do some one experiencing running jabber applications (jabberstatus...) in
> asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I
> got such result.
> IBM*CLI> help jabber
> No such command 'jabber'.
> IBM*CLI> help jabberstatus
> No such command 'jabberstatus'.
>
>
> Any one can help me on this, or may be I miss out somethings that cause
> jabber applications did'nt install.

It looks like res_jabber is not installed on your system. If
res_jabber is loaded, the output of the 'module show like jabber'
command should match the following :
*CLI> module show like jabber
Module Description
 Use Count
res_jabber.so  AJI - Asterisk Jabber Interface
 0
1 modules loaded

The jabber/XMPP related modules depend on the iksemel library, which
depends on GnuTLS. Note that starting from the 1.6 series, GnuTLS is
not used anymore by res_jabber, as we moved to OpenSSL.

You should check your module installation configuration via 'make
menuselect', and make sure your system supports the libraries required
by res_jabber.

Philippe

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Re: [asterisk-users] FXO Cards - T38

2008-02-26 Thread zoa


T.38 will not work with the fxo card.

Zoa

Fernando Berretta wrote:
> Dear All,
>
> Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax 
> etc. and will be able to receive faxes and negotiate with voip CPE's 
> like ATA's to transmit faxes which comes from FXO cards to VoIP 
> Devices using T38 ? it is possible to compile this version of app_fax 
> to work with Asterisk 1.4x ? Someone has tried it ?
>
> Best Regards,
> Fernando
>
> Thomas Kenyon wrote:
>> Steve Underwood wrote:
>>   
   
   
>>> I thought * was still not capable for T.38 gateway operation. Doesn't 
>>> beta 4 just added T.38 termination? And, I believe it misses out some 
>>> key elements of doing that properly. Note that T.38 termination is an 
>>> addon, so it can't be used with, say, G.729.
>>> 
 The only real option available at the moment is to keep one PSTN line on 
 an ATA with an FXO port and T.38 support available and direct calls from 
 the fax machines through to it.  However, I should point out that while 
 I believe this should be possible, I haven't actually tried it myself.

   
   
>> The new asterisk T.38 functionality is from the Asterisk addons 1.6.0b2 
>> version of app_fax (and a few small changes in 1.6.0b4), which I thought 
>> someone would have mentioned to you, since it does use spandsp.
>>
>> (Or at least the configure script checks for spandsp, I haven't actually 
>> looked at the code).
>>
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>
> 
>
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