Re: [asterisk-users] Agents outbound calls to be recorded

[Alejandro Kauffmann]

On 7/6/2011 4:36 AM, bilal ghayyad wrote:

Hi All;

I know that incoming calls for the agent can be recorded, but how I can let the 
outbound calls for the agents to be recorded? I can determine the directory to 
store the outbound calls of the agents to be other than the directory to store 
the incoming calls of the agents?

Regards
Bilal


This is an example of what we do.

MixMonitor(crm/${STRFTIME(${EPOCH},,%B)}/${STRFTIME(${EPOCH},,%d-%m-%Y)}/${STRFTIME(${EPOCH},,%Y%m%d)}-${EXTEN:3}N-${UNIQUEID}-${CALLERID(NUM)}.wav,v(-1)V(2)b,)


What this does is save the recording in:

/var/spool/asterisk/monitor/crm/July/06-07-2011/ (Date in Euro format)

with name:

mmdd-dialednumber-uniqueid-extensionthatdialed.wav

Warning:

I've seen 1.8 create the directory if it does not exist.  Asterisk 1.4 
will NOT create it.  Don't know what 1.6 does with it.


Alex

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Re: [asterisk-users] Huawei K3765 + Internet + SMS + Telephone

[Alejandro Kauffmann]

On 3/31/2011 3:05 PM, Michelle Konzack wrote:

Hello Hans Witvliet,

Am 2011-03-31 22:24:50, hacktest Du folgendes herunter:

Hi Michelle,

Perhaps i'm not understanding your question correctly.
> From what i read, i seems that you got your huawei working correctly as
an umts/hspa-modem, But now you want to use sms/voip directly?

There are some devices created by "udev" and it seems I have to tty  and
a sound port or something like this...


afaict, you can only use the voice/text-services from asterisk over the
IP-layer offered by your modem.

Do you mean with the "tty" and the sound port?


If you want to use the GSM-chip directly, you need (parts of) another
project: not asterisk, but openbsc. But i don't think that they are yet
capable of communicating to Huawei-hardware (i have one myself)

I think not

I know with FreeSWITCH it is possibel, but FreeSWITCH is not  in  Debian
nor is it stabel enough.

(I have Asterisk and FreeSWITCH installed to do testing)


You got a working IP-connecion ontop of the underlying gsm-stuff, and
have access only on anything on the ip-level, not to the protocols
underneath, i think

Hmmm...

Thanks, Greetings and nice Day/Evening
 Michelle Konzack



Look into chan_datacard.

http://forge.asterisk.org/gf/project/chan_datacard/

Alex

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Re: [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)

[Alejandro Kauffmann]

James Lamanna wrote:
> Hi,
> Does anyone have any good empirical data suggesting what the maximum
> number of PRI calls (incoming and outgoing)
> without hardware echo cancellation can be handled on a single box is?
> I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of
> D-Channels going down and then coming back up (See below).
>
> I've looked at the number of simultaneous calls at each of these
> points, and each time the span seems to
> have around 21-23 calls, and the total number of calls ranges between 47 and 
> 53.
> I'm trying to figure out if this is a load issue or an issue on the
> provider side, though my provider says they
> do not see any errors on any of the T1s.
> Could this be some sort of hardware interrupt problem? If so, how can I check?
>
> The specs of the machine are, Dual Xeon 2.80Ghz (both single core but w/HT)
> 4GB memory.
> Running asterisk 1.4.26.3 (32-bit)
> with libpri-1.4.7 and zaptel-1.4.12.9
>
> Thanks.
>
> -- James
>
> Please CC me on responses.
>
>
> [Mar 22 09:45:00] VERBOSE[8887] logger.c:   == Primary D-Channel on span 2 
> down
> [Mar 22 09:45:00] WARNING[8887] chan_dahdi.c: No D-channels available!
>  Using Primary channel 48 as D-channel anyway!
> [Mar 22 09:45:00] VERBOSE[8887] logger.c:   == Primary D-Channel on span 2 up
> [Mar 22 09:59:23] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 
> down
> [Mar 22 09:59:23] WARNING[8886] chan_dahdi.c: No D-channels available!
>  Using Primary channel 24 as D-channel anyway!
> [Mar 22 09:59:23] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 up
> [Mar 22 09:59:23] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 
> down
> [Mar 22 09:59:23] WARNING[8886] chan_dahdi.c: No D-channels available!
>  Using Primary channel 24 as D-channel anyway!
> [Mar 22 09:59:23] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 up
> [Mar 22 10:36:11] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 
> down
> [Mar 22 10:36:11] WARNING[8886] chan_dahdi.c: No D-channels available!
>  Using Primary channel 24 as D-channel anyway!
> [Mar 22 10:36:11] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 up
> [Mar 22 10:36:11] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 
> down
> [Mar 22 10:36:11] WARNING[8886] chan_dahdi.c: No D-channels available!
>  Using Primary channel 24 as D-channel anyway!
> [Mar 22 10:36:11] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 up
> [Mar 22 10:44:36] NOTICE[] chan_dahdi.c: PRI got event: HDLC Bad
> FCS (8) on Primary D-channel of span 3
> [Mar 22 10:45:44] NOTICE[8886] chan_dahdi.c: PRI got event: HDLC Bad
> FCS (8) on Primary D-channel of span 1
> [Mar 22 10:59:33] NOTICE[8887] chan_dahdi.c: PRI got event: HDLC Abort
> (6) on Primary D-channel of span 2
> [Mar 22 11:30:53] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 
> down
> [Mar 22 11:30:53] WARNING[8886] chan_dahdi.c: No D-channels available!
>  Using Primary channel 24 as D-channel anyway!
> [Mar 22 11:30:53] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 up
> [Mar 22 15:34:28] VERBOSE[8887] logger.c:   == Primary D-Channel on span 2 
> down
> [Mar 22 15:34:28] WARNING[8887] chan_dahdi.c: No D-channels available!
>  Using Primary channel 48 as D-channel anyway!
> [Mar 22 15:34:28] VERBOSE[8887] logger.c:   == Primary D-Channel on span 2 up
>
>   
I believe your problem is IRQ sharing which is why you are getting the 
"...HDLC Bad FCS..." notices in your debug.  Make
 sure the card is not sharing IRQs with the network card(s), or the hard 
drive(s).  You MIGHT get by with it sharing with other
 devices.  Upgrade to a newer  Digium card that is not as picky with IRQ 
sharing or switch brands.  I've had several TE410
 cards and they don't like to share.

Alex

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Re: [asterisk-users] Error Dialplan ?

[Alejandro Kauffmann]

Phibee Network Operation Center wrote:
> Hi
>
> I have a problems with a new Asterisk Server,
>
> when i want call, i have:
>
> [Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160 
> handle_request_invite: Call from 'PHISIP01' to extension 
> '00420225352184' rejected because extension not found.
>
> but into my extensions.conf:
>
> exten => _00420X.,1,Set(CDR(CodeTier)=CZE)
> exten => _00420X.,2,Dial(SIP/${ext...@as5350,180,rt)
> exten => _00420X.,3,Hangup
>
> and a dialplan show:
>
> exten => _00420X.,1,Set(CDR(CodeTier)=CZE)
> exten => _00420X.,2,Dial(SIP/${ext...@as5300,180,rt)
> exten => _00420X.,3,Hangup
>
>
> I have a error, it's sure because i am new user, but ?
>
>
> context of the user are good
>
> thanks
> Jerome
Shot in the dark since you don't include enough of your dialplan.

[context where incoming call is landing]
include => context where your extension is defined

[context where your extension is defined]
exten => _00420X.,1,Set(CDR(CodeTier)=CZE)
exten => _00420X.,2,Dial(SIP/${ext...@as5350,180,rt)
exten => _00420X.,3,Hangup


Alex

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Re: [asterisk-users] Echo and static on PRI with errors

[Alejandro Kauffmann]

Tom O'Connor wrote:
>
>
> On Tue, Jun 30, 2009 at 3:12 PM, Tilghman Lesher 
>  > wrote:
>
> On Tuesday 30 June 2009 08:24:29 Tom O'Connor wrote:
> >  I'm currently
> > pointing fingers at either the hardware (someone on #asterisk
> said it could
> > be a cruddy chipset, but it's an HP Server.. so should be
> kosher.. ), I
>
> Is it an HP server from the HP server line, or is it an HP server
> from the old
> Compaq line?  Don't assume that because of the HP name, it's
> actually reliable
> with 3rd party hardware.
>
> It's a HP DL145 G2.  more than that, i can't say.
>
>
>
> -- 
> Tom O'Connor
>
> http://www.twinhelix.org
> t...@twinhelix.org 
> 
>
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The card is TE110P compatible and as such probably suffers from the same 
interrupt sharing problem.  The "...HDLC Bad FCS.." messages tend to be 
related to interrupt sharing.
What does lspci -vb show?  Anything sharing interrupts with the card?

Alex

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[asterisk-users] Queue moh problem with 1.4.23.1

[Alejandro Kauffmann]

I just installed 1.4.23.1 with the queue realtime logger backport.  Here 
are my configs:

musiconhold.conf

[default]
mode=files
directory=/var/lib/asterisk/moh-native
random=yes

queues.conf

[7703]
wrapuptime=0
timeout=15
strategy=rrmemory
retry=5
queue-youarenext=queue-youarenext
queue-thereare=queue-thereare
queue-thankyou=queue-thankyou
queue-callswaiting=queue-callswaiting
queue-holdtime=queue-holdtime
queue-minutes=queue-minutes
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=no
joinempty=yes
context=
announce-holdtime=no
announce-frequency=0
periodic-announce-frequency=20
periodic-announce=/var/lib/asterisk/moh-native/fpm-calm-river
;periodic-announce=/var/lib/asterisk/sounds/queue-youarenext
;periodic-announce=/var/lib/asterisk/sounds/queue-thereare
;periodic-announce=/var/lib/asterisk/sounds/queue-thankyou
;periodic-announce=/var/lib/asterisk/sounds/queue-callswaiting
;periodic-announce=/var/lib/asterisk/sounds/queue-holtime
;periodic-announce=/var/lib/asterisk/sounds/queue-minutes
servicelevel=20

log:

[Mar  6 05:19:20] DEBUG[21589] res_config_mysql.c: MySQL RealTime: 
Everything is fine.
[Mar  6 05:19:20] DEBUG[21589] res_config_mysql.c: MySQL RealTime: 
Insert SQL: INSERT INTO queue_log SET time = '1236338360', callid = 
'1236338360.9', queuename = '7703', agent = 'NONE', event = 
'ENTERQUEUE', data = '|3030'
[Mar  6 05:19:20] DEBUG[21589] res_config_mysql.c: MySQL RealTime: row 
inserted on table: queue_log, id: 69
[Mar  6 05:19:20] VERBOSE[21589] logger.c: -- Started music on hold, 
class 'default', on IAX2/3030-2790
[Mar  6 05:19:20] DEBUG[21589] app_queue.c: Everyone is busy at this time
[Mar  6 05:19:25] DEBUG[21589] app_queue.c: Everyone is busy at this time
[Mar  6 05:19:30] DEBUG[21589] app_queue.c: Everyone is busy at this time
[Mar  6 05:19:35] DEBUG[21589] app_queue.c: Everyone is busy at this time
[Mar  6 05:19:40] VERBOSE[21589] logger.c: -- Stopped music on hold 
on IAX2/3030-2790
[Mar  6 05:19:40] VERBOSE[21589] logger.c: -- Playing periodic 
announcement
[Mar  6 05:19:40] VERBOSE[21589] logger.c: --  
Playing '/var/lib/asterisk/moh-native/fpm-calm-river' (language 'en')

Good thing is backport is working just fine.  The problem is that I hear 
no moh even though log says it was started.  I also do not hear any of 
the queue announcements.  Whatever I set for periodic announcement I 
hear just fine (as shown in queues.conf above)and they are the same 
files I do not otherwise hear in the proper settings.

It's early morning and I might be missing something, but anyone have any 
clues?

Alex

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Re: [asterisk-users] Problems getting 1.6 to run with user asterisk and group asterisk

[Alejandro Kauffmann]

Tzafrir Cohen wrote:
> On Tue, Jan 06, 2009 at 09:39:36PM -0600, Alejandro Kauffmann wrote:
>> Tzafrir Cohen wrote:
>>> On Tue, Jan 06, 2009 at 02:28:53AM -0600, Alejandro Kauffmann wrote:
>>>> I've built SVN-trunk-r167180 and try to start it with:
>>>>
>>>> asterisk -f -C /etc/asterisk/asterisk.conf
>>>>
>>>> which results in:
>>>>
>>>> Unable to open pid file '/var/run/asterisk.pid': Permission denied
>>>> Unable to bind socket to /var/run/asterisk.ctl: Permission denied
>>>>
>>>> However, /etc/asterisk/asterisk.conf has:
>>>>
>>>> astrundir => /var/run/asterisk
>>>> runuser = asterisk
>>>> rungroup = asterisk
>>> Could you please post the complete file? (maybe grep -v '^;')
>>>
>> Tzafrir here is the output you requested:
>>
>> [directories](!) ; remove the (!) to enable this
> 
> With the '(!)' this section has no effect.
> 
>> astetcdir => /etc/asterisk
>> astmoddir => /usr/lib/asterisk/modules
>> astvarlibdir => /var/lib/asterisk
>> astdbdir => /var/lib/asterisk
>> astkeydir => /var/lib/asterisk
>> astdatadir => /var/lib/asterisk
>> astagidir => /var/lib/asterisk/agi-bin
>> astspooldir => /var/spool/asterisk
>> astrundir => /var/run/asterisk
>> astlogdir => /var/log/asterisk
> 
Tsafrir:

Thank you for that.  My reading skills must not be up to par.  I guess 
the use of a template section in asterisk.conf was not something I 
expected and thus ignored.

Alex

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Re: [asterisk-users] Problems getting 1.6 to run with user asterisk and group asterisk

[Alejandro Kauffmann]

Tzafrir Cohen wrote:
> On Tue, Jan 06, 2009 at 02:28:53AM -0600, Alejandro Kauffmann wrote:
>> I've built SVN-trunk-r167180 and try to start it with:
>>
>> asterisk -f -C /etc/asterisk/asterisk.conf
>>
>> which results in:
>>
>> Unable to open pid file '/var/run/asterisk.pid': Permission denied
>> Unable to bind socket to /var/run/asterisk.ctl: Permission denied
>>
>> However, /etc/asterisk/asterisk.conf has:
>>
>> astrundir => /var/run/asterisk
>> runuser = asterisk
>> rungroup = asterisk
> 
> Could you please post the complete file? (maybe grep -v '^;')
> 

Tzafrir here is the output you requested:

[directories](!) ; remove the (!) to enable this
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk

[options]
runuser = asterisk ; The user to run as
rungroup = asterisk ; The group to run as
documentation_language = en_US ; Set the Language you want Documentation 
displayed in. Value is in the same format as locale names


[compat]
pbx_realtime=1.6
res_agi=1.6
app_set=1.6


Alex

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[asterisk-users] Problems getting 1.6 to run with user asterisk and group asterisk

[Alejandro Kauffmann]

I've built SVN-trunk-r167180 and try to start it with:

asterisk -f -C /etc/asterisk/asterisk.conf

which results in:

Unable to open pid file '/var/run/asterisk.pid': Permission denied
Unable to bind socket to /var/run/asterisk.ctl: Permission denied

However, /etc/asterisk/asterisk.conf has:

astrundir => /var/run/asterisk
runuser = asterisk
rungroup = asterisk

The directory, user, and group exist.  Permissions are fine.  The only 
way I can get it to run is to edit defaults.h and change:

#define DEFAULT_RUN_DIR"/var/run/asterisk"
#define DEFAULT_SOCKET "/var/run/asterisk/asterisk.ctl"
#define DEFAULT_PID"/var/run/asterisk/asterisk.pid"

After these changes and recompiling all is fine.

It seems that at startup asterisk.conf is being parsed, runuser and 
rungroup are being set, but astrundir is not overriding the defaults.

Am I missing something?

Alex

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Re: [asterisk-users] set monitor_filename

[Alejandro Kauffmann]

Ralf Träskman wrote:
> Hi
> 
> Hm is this function for recording? The thing I want to do is to be able to 
> see how many calls there is waiting in a queue, maybe im looking in the wrong 
> direction?
> 
> /ralf
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro 
> Kauffmann
> Sent: den 5 december 2008 05:03
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] set monitor_filename
> 
> Ralf Träskman wrote:
>> Hi
>>
>>  
>>
>> I have this in my queue extension and I see this in asterisk when I call 
>> to the queue, but no file is created in the directory any ideas?
>>
>>  
>>
>> exten => 
>> s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID})
>>
>>  
>>
>> -- Executing [EMAIL PROTECTED]:1] Set("SIP/0850001175-b7942770", 
>> "MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12")
>>
>>  
>>
>> Regards
>>
>> /ralf
> 
> The basics.  Does the queuecalls directory exist?  Does the user that * 
> runs under have write permission in that directory?
> 
> The monitor-format parameter MUST be set in queues.conf to enable 
> recording and to select the format of the recording.  In addition, I 
> believe this is still true in 1.4, if you don't set the monitor-join 
> parameter to yes you will end up with two files (in & out) instead of a 
> single file with both legs of the call.
> 
> Alex
> 

Yes, this is for recording.  Take a look at 
http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue and check some of 
the links in the "See also" section at the bottom of the page.  Google 
for asterisk+queue+stats and you will find others.  Check the list 
archives for some more.

Alex


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Re: [asterisk-users] set monitor_filename

[Alejandro Kauffmann]

Ralf Träskman wrote:
> Hi
> 
>  
> 
> I have this in my queue extension and I see this in asterisk when I call 
> to the queue, but no file is created in the directory any ideas?
> 
>  
> 
> exten => 
> s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID})
> 
>  
> 
> -- Executing [EMAIL PROTECTED]:1] Set("SIP/0850001175-b7942770", 
> "MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12")
> 
>  
> 
> Regards
> 
> /ralf

The basics.  Does the queuecalls directory exist?  Does the user that * 
runs under have write permission in that directory?

The monitor-format parameter MUST be set in queues.conf to enable 
recording and to select the format of the recording.  In addition, I 
believe this is still true in 1.4, if you don't set the monitor-join 
parameter to yes you will end up with two files (in & out) instead of a 
single file with both legs of the call.

Alex

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Re: [asterisk-users] 2 Asterisks to one PBX - E1 conection

[Alejandro Kauffmann]

dubravko caric wrote:
> Hi all,
> 
> I have a question regarding connection of two Asterisk servers to our 
> PBX. Each Asterisk server has one PCI E1 card, and they are in failover 
> mode with Linux HA. On our PBX we have only one E1 card towards Asterisk 
> servers.
> 
> My question is how to connect these two Asterisks to one E1 card on PBX, 
> and when primary Asterisk server fails not to have to manually pull out 
> E1 cable from primary server and plug it in secondary server in order to 
> have active connection to E1 card on PBX.
> 
> Is there some kind of splitter which, on one side can accept two E1 
> connections from Asterisks and on the other side one E1 link from PBX. 
> This splitter must also recognize towards which one of two E1 links on 
> Asterisk side it should send signals to. eg. when primary Asterisk fails 
> this splitter should send signals to its eg. port 2 (connection towards 
> secondary Asterisk).
> 
> I would be most grateful if someone could provide me with a link to such 
> products.
> 
> Thanks
> 
> Dubravko
> 
Don't know how well it works, but we've been looking at these:

http://www.rhinoequipment.com/1portfail.html

Alex

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Re: [asterisk-users] Update (IAX Trunking Help)

[Alejandro Kauffmann]

Steve Anness wrote:
> Thanks for the all the help, I have been pulling my hair out
> 
> I now have the trunk working in both directions.  However, how do I add
> voicemail capability?
> 
> 
> exten => _11XXX,1,Dial(iax2/colo/${EXTEN:2},20,Ttr)
> exten => _11XXX,n,Voicemail(${EXTEN:2:3}|su)
> 
> Thinking that if I dialed 11127 and after 20 if there was no answer it would
> dial 327 (but maybe am mistaken about that).  327 is the voicemail box for
> extension 127. 
> 
> Steve Anness 


${EXTEN:2:3} says:

Start with an offset of 2 and dial the next 3 digits.  Offset 0 is the 
first 1, offset 1 is the second 1, and offset 2 is the first X.  This 
will simply dial XXX not substitute the first X for a 3.  You would need 
something like:

exten => _11XXX,n,Voicemail(3${EXTEN:3:2}|su)

Not sure why the mailbox for 127 isn't 127, but you probably have your 
reasons.

Alex

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Re: [asterisk-users] Record name for conference...

[Alejandro Kauffmann]

Carlos Chavez wrote:
> I have a customer that wants to use meetme but they want to have the users
> record their name so it is played to the other people on the conference. Is
> there an easy way to do this?
> 
> --
> Carlos Chavez
> Director de Tecnología
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Tel: +52-55-91169161 Ext 2001
> 
> 
I believe the "i" option is what you are looking for.  The caller will 
be asked to record, review, and approve the recording before they join. 
That recording will be played announcing the person joining/leaving the 
conference.

Alex

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Re: [asterisk-users] Update (IAX Trunking Help)

[Alejandro Kauffmann]

Steve Anness wrote:
> I posted earlier in the day about needed help with IAX trunking.  I did 
> some more reading and made some more changes.
> 
> Here is what I have thus far:
> 
> Iax.conf on one server
> 
> [general]
> bindport = 4569   
> bindaddr = 0.0.0.0  
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> mailboxdetail=yes
> 
> [vvfarm]
> type=friend
> username=colo
> secret=testpassword
> auth=plaintext
> host=64.194.211.170
> context=iax-incoming
> peercontext=vvfarm-extensions
> qualify=yes
> trunk=yes
> 
> Extensions.conf on the same server
> 
> [iax-incoming]
> exten => _###,1,Dial(SIP/17${EXTEN}-1,20)
> 
> [remote-extensions]
> 
> exten => _1,1,Dial(SIP/17${EXTEN}-1,20)
> exten => _1,n,Voicemail(${EXTEN:0:3}|su)
> exten => _1,n,Dial(SIP/${EXTEN}-1)
> 
> exten => _11XXX,1,Dial(iax2/vvfarm/${EXTEN:2}-1,20)
> 
> Iax.conf on server B
> 
> [general]
> bindport = 4569
> bindaddr = 0.0.0.0  
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> mailboxdetail=yes
> 
> [colo]
> type=friend
> username=vvfarm
> secret=testpassword
> auth=plaintext
> host=72.249.129.91
> context=iax-incoming
> peercontext=remote-extensions
> qualify=yes
> trunk=yes
> 
> Extensions.conf on server B
> 
> [vvfarm-extensions]
> exten => _1XX,1,Dial(SIP/${EXTEN}-1,20)
> exten => _1XX,n,Voicemail(${EXTEN:0:3}|su)
> exten => _1XX,n,Dial(SIP/${EXTEN}-1)
> 
> exten => _17XXX,1,Dial(iax2/colo/${EXTEN}-1,20)
> 
> [iax-incoming]
> 
> exten => _XXX,1,Dial(SIP/${EXTEN}-1,20)
> 
> The error I am getting when trying to call from Server A to Server B is
> 
> [Oct  8 17:13:00] NOTICE[3616]: chan_iax2.c:7367 socket_process: 
> Rejected connect attempt from 72.249.129.91, who was trying to reach 
> '[EMAIL PROTECTED]'
> 
> The error I am getting when trying to call from server B to Server A is
> 
> [Oct  8 17:26:46] NOTICE[3115]: chan_iax2.c:7332 socket_process: 
> Rejected connect attempt from 64.194.211.170, who was trying to reach 
> '[EMAIL PROTECTED]'
> 
> What have I done wrong?  Why won’t it dial 17119-1 and 127-1, respectfully.
> 
> Steve Anness

Your patterns don't match.  You are sending [EMAIL PROTECTED], but 
vvfarm-extensions has no pattern xxx-1.  Same problem in the other 
direction.  Try changing the dial statement in server A from:

exten => _11XXX,1,Dial(iax2/vvfarm/${EXTEN:2}-1,20)

to:

exten => _11XXX,1,Dial(iax2/vvfarm/${EXTEN:2},20)

and in server B from:

exten => _17XXX,1,Dial(iax2/colo/${EXTEN}-1,20)

to:

exten => _17XXX,1,Dial(iax2/colo/${EXTEN},20)


Alex

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Re: [asterisk-users] Asterisk Queue's

[Alejandro Kauffmann]

Tobias Ahlander wrote:
>  >Date: Tue, 02 Sep 2008 18:08:52 +1200
>  >From: Paul Crane <[EMAIL PROTECTED] >
>  >Subject: Re: [asterisk-users] Asterisk Queue's
>  >To: Asterisk Users Mailing List - Non-Commercial Discussion
>  > >
>  >Message-ID: <[EMAIL PROTECTED] 
> >
>  >Content-Type: text/plain; charset=ISO-8859-1
>  >
>  >-BEGIN PGP SIGNED MESSAGE-
>  >Hash: SHA1
>  >
>  >Philipp Kempgen wrote:
>  >> Tobias Ahlander schrieb:
>  >>
>   From: Mark Michelson <[EMAIL PROTECTED] 
> >
>  >>
>   Tobias Ahlander wrote:
>  >>
>  > Yes, I have autofill set in queues.conf. I suspect that this 
> behaviour
>  > is because the Polycom phones I use have 2 lines. Has anyone used 
> this
>  > function with polycom phones before? Also, my agents are Dynamic,
>  > perhaps this works better with Static agents?
>  >
>  > Here's my queues.conf (with commented lines deleted for easier 
> reading):
>  >
>  > [general]
>  > autofill = yes
>  > monitor-type = MixMonitor
>  >
>  > [sales]
>  > strategy = rrmemory
>  > wrapuptime=15
>  >
>   Depending on which Asterisk version you are using, there was a bug 
> in the
>  >>> queue
>   application for some 1.4 releases where the autofill option would 
> only be
>  >>> set
>   properly if it were placed inside a queue. In other words, you may 
> want to
>  >>> try
>   putting autofill=yes inside the [sales] queue in your configuration.
>  
>   Also, if you're using a version of Asterisk 1.2, autofill is not a 
> valid
>  >>> option
>   and you'll be stuck with the behavior you're seeing.
>  >>
>  >>> Unfortunately this didn't help at all... Anyone else has any tips? 
> Is there
>  >>> a way to limit the polycom phones to only take one call from the 
> Queue at
>  >>> the same time? Asterisk version running is 1.4.13
>  >>
>  >> Maybe the phones have call-waiting enabled?
>  >> Does it work if you remove the second line?
>  >>
>  >>
>  >>Philipp Kempgen
>  >>
>  >
>  >Try setting the call-limit to 1 in sip.conf as well as limitonpeer to yes.
>  >
>  >- --
>  >Paul Crane
>  >
>  >Technical Support Officer
>  >VentureVoIP Ltd
>  >John Wickliffe House
>  >265 Princes Street
>  >Dunedin
> 
> Paul,
> 
> This option doesn't help me that much. When I have it enabled, I can't 
> put a call on hold and transfer it since Asterisk rejects usage limit to 1.
> 
> Philipp,
> 
> I'm using Polycom phones. When I set the "Calls Per Line" (which I'm 
> told is Call Waiting) I seem to be able to transfer calls etc, but I'm 
> still noticing the same behaviour with the queues as before.
> 
> 
> Any more tricks I can try?
> 

Have you tried ringinuse=no in the queue definition in queues.conf and 
call-limit=2 in sip.conf?

Alex

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Re: [asterisk-users] Number portability in other parts of the world.

[Alejandro Kauffmann]

Alexander Lopez wrote:
> I think it would be a good idea to start an item in the Wiki about this.
>
> Can anyone else chime in for their countries??
>
> Others in the EU, Eastern, Far East?
>
> So Far I have:
>
> Australia:PSTN to PSTN and Cell to Cell are OK , but Cell to PSTN and 
> PSTN to Cell are NOT OK.Dean Collins
>
> Poland:   Not Today but possibly in 2009  Daniel  
>
> UK:   Portable if Telco has a porting agreement. Not all Telco have 
> agreements in place.  Steve Kennedy
>
> France: Porting from France Telcom to another provider not an issue, however 
> if porting between other Telco's, Telco's must have porting agreement between 
> them.  Randulo
>
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Administrator TOOTAI
>> Sent: Thursday, June 26, 2008 8:48 AM
>> To: asterisk-users@lists.digium.com
>> Subject: Re: [asterisk-users] Number portability in other parts of the
>> world.
>>
>> Steve Kennedy a écrit :
>> 
>>> [...]
>>>   
Are the same rules and conditions that exist here in the States
mirrored elsewhere?
How does a person in Europe go fully VoIP and still keep the main
number?

 
>>> In the UK numbers are portable, though the telco wanting the number must
>>> have a porting agreement with the telco that has the number. Not all
>>> telcos have porting agreements.
>>>
>>>   
>> Same in France. If the number is an original France Telecom one, no
>> problem. If the number was _already_ ported, can be a problem. In all
>> other cases, I would suggest you to check if there is an agreement
>> between telco.
>>
>> In Poland, not possible today, should be end 2008/begining 2009.
>>
>> --
>> Daniel
>>
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>
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>   

Mexico will have portability starting July 5th of this year.  It is 
limited to PSTN to PSTN and Cell to Cell
within the same geographic region.  In this case, PSTN includes the new 
phone service provided by the
Cable TV companies and other players.  The final rules are still in 
flux, but it seems once you switch you will have to wait 6
months before you can switch again.  The process, no surprise here, will 
be encumbered by strict documentation
requirements.

How well it will work is anyones guess.  Telmex still owns the bulk of 
the "last mile" to homes, and I have yet to
see anything that says they will have to provide access to their 
competitors.  Cable companies will have an advantage here
as they already have homes wired to their network.  The electric 
companies (state owned) are joining the game by leasing
bandwidth on their grid, but it's currently limited to other carriers.  
As good as their electric service is, I wouldn't be surprised
to be "shocked" by the quality of the calls on their grid.

The real battle will be in Cell service.  While Telmex has a 70%+ market 
share, 70%+ of that is prepaid with no
contractual obligations.  If the authorities oversee this as well as 
they did changing long distance carriers a few years back, I expect
that i'll be back with Telmex within weeks of switching without having 
requested the change.  Still, it's progress.

Alex

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Re: [asterisk-users] Multi-SPAN (4xE1) Zap Group (Outbound)

[Alejandro Kauffmann]

Mike Trest - Personal wrote:
> Hi,
> Can someone point me to a zapata.conf example that will create a 
> single DIAL OUT
> group including all 4 spans on a TE4XXP?
>
> One friend says to change the group number all to "1" on all 4 spans.
> Another suggestions says it is possible to have these unique groups (1-4)
> and to combine all 4 into a single group "5".
>
> I like the second suggestion best.
>
> Can you guide me to the correct changes for my current zapata.conf?
> The 4 spans are stand alone E1/PRI trunks (Not NFAS).
>
> The CURRENT channel and group statements are:
> ;Span  1  group=1 channel  => 1-15,17-31
> ;Span  2  group=2 channel  => 32-46,48-62
> ;Span  3  group=3 channel  => 63-77,79-93
> ;Span  4  group=4 channel  => 94-103,110-124
>
>
>
>
> Thanks,  ..mike..
>
>
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>
>   
Try:

group=0,1
channel  => 1-15,17-31
group=0,2
channel  => 32-46,48-62
group=0,3
channel  => 63-77,79-93
group=0,4
channel  => 94-103,110-124

This allows you to use group 0 to dial out over all 4 spans, but each span 
still has it's own
group that you can use to troubleshoot.  You can break this down even further 
if you need.


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Re: [asterisk-users] RANT (was Re: Which IP Phone is really the best?)

[Alejandro Kauffmann]

RE Kushner List Account wrote:
> Bill Hackensack wrote:
>   
>> I realize my messages may seem rude and obnoxious, but let's face it, 
>> I'm just saying what the rest of you are thinking.  I learned by 
>> reading, reading, and reading.  
>> 
> 
> Nobody wants to search, nobody wants to read - HOW DARE YOU!
>
> Only geeks and nerds read, the rest of us are spoon fed! You should know 
> better.
> 
>
> I understand your rant, I can't tell you how many people call me and I 
> type their question into Google and poof, magic, I have their answer. 
> And they all think I'm a God or something.
>
> -Ron
> http://www.m102.com
>
>
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>
>   
As long as this is an official rant thread

Good to know no new phones have hit the market since the last time this 
question was asked and answered.  It's also good to know
opinions about specific products don't change over time.  It's great to 
know that no bugs have been found in any of the "best" phone's
firmware that might drop them in the ranking since the last time this 
question was asked.

I wonder if this rant thread would exist if the OP had not mentioned he 
was preparing a quote for a customer?

/sarcasm off

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Re: [asterisk-users] PRI dialout problem with some numbers...

[Alejandro Kauffmann]

Carlos Chavez wrote:
>   I have an Asterisk server (1.4.13) using PRI in Monterrey, Mexico.
> This is really the first server I have used with PRI in Mexico as we
> normally use MFC/R2.  Everything seems to be working except that some
> numbers always seem to be busy when you dial them.  All these numbers
> belong to different phone companies.  I know that with R2 this problem
> is present if you have a "#define DEFAULT_T1" value under 15000 in
> mfcr2.c (the default used to be 5000).  Is there an equivalent value for
> PRI?  The company we are using is Alestra.  Here is what I get when we
> dial a number that belongs to a company called Protel:
>
> -- Executing [EMAIL PROTECTED]:1] Set("SIP/199-08be6c00",
> "TIMEOUT(absolute)=3600") in new stack
> -- Channel will hangup at 2007-11-05 22:03:34 UTC.
> -- Executing [EMAIL PROTECTED]:2] Dial("SIP/199-08be6c00",
> "Zap/g1/11070665||Ww") in new stack
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called g1/11070665
> -- Zap/1-1 is proceeding passing it to SIP/199-08be6c00
> -- Channel 0/1, span 1 got hangup request, cause 31
> -- Hungup 'Zap/1-1'
> [Nov  5 15:03:34] NOTICE[22300]: cdr.c:434 ast_cdr_free: CDR on channel
> 'Zap/1-1' not posted
>   == Everyone is busy/congested at this time (1:0/0/1)
>
>
>
>   
> 
>
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>
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.503 / Virus Database: 269.15.22/1112 - Release Date: 11/5/2007 
> 7:11 PM
>   

We have several sites setup with Alestra.  Contact me off list if you 
like and I'll see what I can do to help you out.

Alex

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Re: [asterisk-users] PRI span configuration - span remains down

[Alejandro Kauffmann]

David Kennedy wrote:
> Hi
>
> While I have fixed the problem from this post, I do have another
> problem, and you have asked for a debug output here, so I'll go
> against my better instinct and reply here :)
>
> -- Making new call for cr 32774
> -- Requested transfer capability: 0x00 - SPEECH
>
>   
>> [ 00 01 0e 06 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 
>> 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ]
>> 
>
>   
>> Informational frame:
>> SAPI: 00  C/R: 0 EA: 0
>>  TEI: 000EA: 1
>> N(S): 007   0: 0
>> N(R): 003   P: 0
>> 44 bytes of data
>> 
> -- Restarting T203 counter
> Stopping T_203 timer
> Starting T_200 timer
>   
>> Protocol Discriminator: Q.931 (8)  len=44
>> Call Ref: len= 2 (reference 6/0x6) (Originator)
>> Message type: SETUP (5)
>> [04 03 80 90 a3]
>> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: 
>> Speech (0)
>>  Ext: 1  Trans mode/rate: 64kbps, circuit-mode 
>> (16)
>>  Ext: 1  User information layer 1: A-Law (35)
>> [18 03 a9 83 86]
>> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  
>> Dchan: 0
>>ChanSel: Reserved
>>   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>>   Ext: 1  Channel: 6 ]
>> [6c 0c 21 83 38 34 35 38 39 39 31 30 30 31]
>> Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI: 
>> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>>   Presentation: Presentation allowed of network 
>> provided number (3)  '8458991001' ]
>> [70 0c 80 30 32 30 38 36 35 39 32 32 39 31]
>> Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown 
>> Number Plan (0)  '' ]
>> [a1]
>> Sending Complete (len= 1)
>> 
> q931.c:2881 q931_setup: call 32774 on channel 6 enters state 1 (Call 
> Initiated)
> -- Called g0/
> -- T200 counter expired, What to do...
> -- Retransmitting 48 bytes
> voip1*CLI>
>   
>> [ 00 01 0e 07 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 
>> 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ]
>> 
> voip1*CLI>
>   
>> Informational frame:
>> SAPI: 00  C/R: 0 EA: 0
>>  TEI: 000EA: 1
>> N(S): 007   0: 0
>> N(R): 003   P: 1
>> 44 bytes of data
>> 
> -- Rescheduling retransmission (1)
> voip1*CLI>
> < [ 00 01 01 11 ]
> voip1*CLI>
> < Supervisory frame:
> < SAPI: 00  C/R: 0 EA: 0
> <  TEI: 000EA: 1
> < Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> < N(R): 008 P/F: 1
> < 0 bytes of data
> -- ACKing all packets from 6 to (but not including) 8
> -- ACKing packet 7, new txqueue is -1 (-1 means empty)
> -- Since there was nothing left, stopping T200 counter
> -- Nothing left, starting T203 counter
> -- Got RR response to our frame
> -- Restarting T203 counter
> voip1*CLI>
> < [ 02 01 06 10 08 02 80 06 5a 08 03 82 ac 18 ]
> voip1*CLI>
> < Informational frame:
> < SAPI: 00  C/R: 1 EA: 0
> <  TEI: 000EA: 1
> < N(S): 003   0: 0
> < N(R): 008   P: 0
> < 10 bytes of data
> -- ACKing all packets from 7 to (but not including) 8
> -- Since there was nothing left, stopping T200 counter
> -- Stopping T203 counter since we got an ACK
> -- Nothing left, starting T203 counter
> < Protocol Discriminator: Q.931 (8)  len=10
> < Call Ref: len= 2 (reference 6/0x6) (Terminator)
> < Message type: RELEASE COMPLETE (90)
> < [08 03 82 ac 18]
> < Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
> Location: Public network serving the local user (2)
> <  Ext: 1  Cause: Requested channel not available
> (44), class = Network Congestion (resource unavailable) (2) ]
> <  Cause data 1: 18 (24)
> -- Processing IE 8 (cs0, Cause)
> q931.c:3503 q931_receive: call 32774 on channel 6 enters state 0 (Null)
> Sending Receiver Ready (4)
> voip1*CLI>
>   
>> [ 02 01 01 08 ]
>> 
> voip1*CLI>
>   
>> Supervisory frame:
>> SAPI: 00  C/R: 1 EA: 0
>>  TEI: 000EA: 1
>> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
>> N(R): 004 P/F: 0
>> 0 bytes of data
>> 
> -- Restarting T203 counter
> -- Restarting T203 counter
> -- Channel 0/6, span 1 got hangup, cause 44
> -- Forcing restart of channel 0/6 on span 1 since channel reported in use
> voip1*CLI>
>   
>> [ 00 01 10 08 08 02 00 00 46 18 03 a9 83 86 79 01 80 ]
>> 
> voip1*CLI>
>   
>> Informational frame:
>> SAPI: 00  C/R: 0 EA: 0
>>  TEI: 000EA: 1
>> N(S): 008   0: 0
>> N(R): 004   P: 0
>> 13 bytes of data
>> 
> -- Restarting T203 counter
> Stopping T_203 timer
> Starting T_200 timer
>   
>> Protocol Discriminator: Q.931 (8)  len=13
>> Call Ref: len= 2 (reference 0/0x0) (Originator)
>> Message type: RESTART (70)
>> [18 03 a9 83 86]
>> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  
>> Dchan: 0
>>ChanSel: Reserved
>>   Ext: 1  Coding: 0  Number Specifie

Re: [asterisk-users] TE210P issues

[Alejandro Kauffmann]

Jerry Geis wrote:
> I have a box with a TE210P. Things work for a while then stop when 
> making call files.
> I get NOANSWER as the return code (right away).
>
> I am running asterisk 1.2.12.1, libpri 1.2.3 and zap 1.2.9.1
>
> When I try to update to newer zaptel the machine locks when loading the 
> zaptel drivers.
>
> I tried to manually load the wct1xxp module (I think that is the one for 
> the dual T1 card???)
> and the machine locks. I am in a remote location so I cannot see if 
> anything is on the console.
>
> I tried jumping to 1.4 and the same thing happens.
> I have updated quite a few asterisk boxes remotely and never had this 
> issue before.
>
> Last thing I tried was "chkconfig zaptel off", reboot, then try loading 
> in new version and the same thing happened.
> It locked up.
>
> After rebooting I put back the old zaptel and it works again for  awhile.
>
> What shall I try?
>
>
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>
>   
The driver for both the TE205P and TE210P is wct4xxp.  That might be why 
your machine locks up.

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Re: [asterisk-users] Stable-Stable Asterisk

[Alejandro Kauffmann]

Doug Lytle wrote:
> Tzafrir Cohen wrote:
>> On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote:
>>   
>> stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and
>> now swear (by?) 1.2 or 1.4.
>>
>>   
> My decision based on what I've been reading in the bug tracker and 
> people commenting on how they've had to roll back to 1.2 to regain a 
> stable system.  We are not having issues with our 1.2.x installs, but 
> I've been 'encouraged' by the development team to upgrade to 1.4.
> 
> Doug
> 
I would tell your development team that this is a mission critical 
system and not a desktop PC.  Unless you must have a feature only 
available in 1.4, leave your mission critical systems alone.  Patch when 
necessary, and upgrade when needed.

My 2 cents

Alex

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RE: [asterisk-users] Registrations, how many is too many?

[Alejandro Kauffmann]

> That is interesting.. Not sure though how getting rid of IAX 
> could have fixed your SIP issues, seems odd.
> 
> We can't really get rid of IAX, our customers would flip their lids.
> 
> The big difference we have is that this has happened on more 
> than one occasion when there was little to no call volume.

Might want to try not mixing the two.  Provide only IAX on one server and
setup another for SIP?  I can't tell you what the relationship is, but we've
tested it more than once and any time IAX call volume goes up, SIP crashes.
When we go back to pure SIP, no problems.

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RE: [asterisk-users] Registrations, how many is too many?

[Alejandro Kauffmann]

> Anyone have any idea if there is some sort of limitation to 
> the number of SIP or IAX end points which can register to an 
> Asterisk system (2.8Ghz dual processor, 2GB ram) while also 
> handling 30-50 simultaneous calls without getting into trouble?
> 
> Of course the 30-50 simultaneous calls end up being 60-100 
> channels of mostly G711 VoIP.
> 
> We have seen issues where our Asterisk just gets all crazy 
> and SIP quits working all together, to the point sometimes 
> where we can't even fix it with a restart.
> 
> At one point we were using all realtime for IAX and SIP 
> clients, then we went to text files (more or less), still we 
> are seeing this issue.
> 
> When this happens we can't even do simple things with SIP 
> like "sip show peers" etc. because Asterisk just says that 
> the application doesn't exist.
> 
> This has been a battle for a few months and we can't put our 
> finger on it. Can't seem to figure out when it's going to 
> happen either which is VERY tough on the nerves to say the least.
> 
> This happens during peak times but also in the middle of the 
> night when call volume is slow to non existent.
> 
> The only thing that's constant during both peak and non peak 
> times is the amount of registrations the system deals with.
> 
> We have approx 1500-1800 end points registering to this 
> particular system at any one time. This is a split between 
> IAX and SIP not sure what the percentage of each is at the moment.
> 
> It's been a long time since a problem has beat me/us and this 
> one has won so far.
> 
> Any help in getting my sanity back would be REALLY 
> appreciated. ___

We've seen the same behavior since the 1.0.x version and have been unable to
track it down.  What I can tell you is that we used to peak around 40 calls
(SIP/IAX to zap over E1 PRI all using alaw) and SIP would crash.  I saw a
peak of 97 calls today (216 channels with chanspy accounting for most of the
difference) and this is on a single core 3.2Ghz with 1Gb ram.  What changed?
We got rid of IAX.  I can't tell you if this will work for you or not, but
my nerves are doing better now that I don't get calls at random hours of the
day telling me that "we have no phones".  As a side note, inbound ZAP(PRI)
would still work, but only to IAX endpoints.

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RE: [asterisk-users] Help Needed: Can't make "local" calls on abrandnewPRI

[Alejandro Kauffmann]

> 
> So the questions:  Is there anyway to further verify that 
> asterisk is  
> not sending any extra digits or filler digits to the telco on 
> the PRI? If the problem is not in asterisk or zaptel, what do 
> I say to the  
> Telco to get them to believe the problem is on their end?
> 

At the console type pri intense debug span 1 (or whatever your span number
is) then place the call and check the output.  Play with the pridialplan
parameter in zapata.conf.  The default is national, try unknown first.  Once
you're done, pri no debug span 1 to turn debugging off.

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RE: [asterisk-users] Simple question

[Alejandro Kauffmann]

> Whats the difference between the following statements in extensions.conf
> include=>inbound
> AND
> #include inbound/*.conf 
 
The first one includes a context the second one includes a file(s). 

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RE: [asterisk-users] IBM Server / USB Ports

[Alejandro Kauffmann]

>
> I see that the digium card doesn't share the IRQ however
> Digium has recommended diabled USB still... additionally the
> Digium card is on 169 which isn't a valid IRQ.. how can I
> find out what it is sharing with?
>

lspci -vb will give you the irq as seen by the cards on the PCI bus

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RE: [asterisk-users] how to indicate an non-existent number?

[Alejandro Kauffmann]

> How do you guys manage that issue? Do you record a message ("sorry, the 
> number dialed can't be completed") and play it when the PRI or BRI 
> returns a specific code? And what code is that?

We check HANGUPCAUSE and playback messages we recorded depending on value.
Check http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php or
google for q.931 cause codes or isdn cause codes.

Alex

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RE: [asterisk-users] Multiple TE110P cards in one chassis

[Alejandro Kauffmann]

Title: Message



 >  Does anyone know if you can have multiple 
TE110P cards in one chassis? 
 
As 
long as you make sure they don't share interrupts, sure.  We have several 
boxes running just fine with anywhere from 1 to 4 
TE110P. 
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RE: [asterisk-users] How big is *your* dialplan??

[Alejandro Kauffmann]

> Hello!

> In my relentless quest for knowledge, I pose this question: who's got the
biggest dialplans, and how big are these monsters?


Our "small" contribution...

1175 extensions (2580 priorities) in 303 contexts

Alex

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RE: [asterisk-users] Asterisk autoloading of card modules

[Alejandro Kauffmann]

No rpms, all compiled from source.  I cheat though, AMP/freepbx for the init
script.  GUI is a godsend when you just need to add a simple phone on a box
that sits across the country and all you have on your mom´s old computer is
a browser.  https looks real nice right about then. :P

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Devraj
Mukherjee
Sent: Monday, July 24, 2006 7:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk autoloading of card modules


Hi Alejandro,

Thanks for  your suggestions. Where did you fetch your rpms?

I had to fix up the init scripts for everything to work

On 7/24/06, Alejandro Kauffmann <[EMAIL PROTECTED]> wrote:
>
> > My /etc/sysconfig/zaptel configuration has only one MODULES 
> > directive
> enabled MODULES="$MODULES wctdm"
>
> > However when I start asterisk it loads the wct1xxp module. Which
> configuration file controls the loading of card > modules?
>
> Check /etc/modprobe.conf  I clear that out and just leave the module I 
> want enabled in /etc/sysconfig/zaptel.
>
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RE: [asterisk-users] Asterisk autoloading of card modules

[Alejandro Kauffmann]

> My /etc/sysconfig/zaptel configuration has only one MODULES directive
enabled MODULES="$MODULES wctdm"

> However when I start asterisk it loads the wct1xxp module. Which
configuration file controls the loading of card > modules?

Check /etc/modprobe.conf  I clear that out and just leave the module I want
enabled in /etc/sysconfig/zaptel.

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RE: RE: Re: [asterisk-users] setting call-limits

[Alejandro Kauffmann]

> can you explain? I don't find any information on it... is this a tool or a
library?

Look at http://www.voip-info.org/wiki-Asterisk+standard+extensions

There's an example on how to setup hints under standard priorities.

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RE: Re: [asterisk-users] setting call-limits

[Alejandro Kauffmann]

>> on 1.2.4 and 1.2.7, we have to set the 'type=peer' for call-limits to 
>> work effectively.
>> 
>> type=friend doesn't seem to enforce call limits at all.
>> 
>> if you haven't tried type=peer, try that first.

>No, this doesn't work. 


I believe you need to setup hints for call-limit to work.
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RE: [Asterisk-Users] HP Proliant server?

[Alejandro Kauffmann]

> Has anyone had any experience running asterisk on a dual-xeon HP Proliant
server. Have you had any experience setting up digium cards on this?

We have asterisk running on a DL140 dual-xeon (only 1 proc atm) with 1GB of
ram and a TE410P.  Results are mixed.  The bios has 3-4 options you can
change none of which help when trying to get the card it's own interrupt.
We start to get dropped calls once we hook up the 4th E1, but we believe it
has more to do with our users generating 10-13GB/day worth of recordings
using both automon and a web application we developed for custom recording.
We hit the manager extensively and do too much transcoding.  

If we ever get around to reducing transcoding, rationalizing the insane
amount of calls that get recorded, add a proxy for the manager, and a few
other tweaks, we believe we can get it to run all 4 E1.  Current daily call
volume tops out at 35k calls.  Peak we cap our 3 E1 which is 90 simultaneous
calls with a couple dozen more via IAX trunks to our servers around the
country.

All in all, a good bang for the buck if you can live on the edge a bit.

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RE: [Asterisk-Users] Zaptel not compiling on lastest Centos 4.2 kernel.

[Alejandro Kauffmann]

RHEL 4 and therefore CentOS 4 had a bug introduced in the latest kernel.  

https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568

This bug report has a typo as well.  It should read:

#define DEFINE_RWLOCK(x) rwlock_t x = RW__LOCK_UNLOCKED

Fix the line and recompile zaptel.  All should be well.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl O. Pinc
Sent: Monday, March 13, 2006 10:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zaptel not compiling on lastest Centos 4.2
kernel.



On 03/13/2006 11:33:18 AM, Chuck Bunn wrote:
> Hi,
> 
> I made a big mistake on a Centos 4.2 box - I forgot to exclude the
> kernel from updating. Now zaptel will not do a "make linux26" see  
> below. Is there a way to roll this back or is there a patch to get  
> Zaptel to compile? I have a link to the modules using 'ln -s  
> /lib/modules/uname -r/build linux-2.6" so that I did not have to  
> specifiy the kernel version directly.

I successfly compiled zaptel 1.2.1 on Linux 2.6.  Perhaps when you upgraded
the kernel you did not install the corresponding kernel-source rpm?

Karl <[EMAIL PROTECTED]>
Free Software:  "You don't pay back, you pay forward."
  -- Robert A. Heinlein

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RE: [Asterisk-Users] H323 compilation Help needed

[Alejandro Kauffmann]

You need to run make from /usr/src/asterisk and not
/usr/src/asterisk/channels/h323.  Just make then make install.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hussain Umair
Sent: Wednesday, January 04, 2006 5:35 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] H323 compilation Help needed


hi all im trying to compile h323 i have got the pwlib and openh323 working 
that is simph323 is running properly but when i try to compile h323 in the 
channels directory it gives me the following error can anybody please help 
me with

[EMAIL PROTECTED] src]# cd /usr/src/asterisk/channels/h323/
[EMAIL PROTECTED] h323]# make opt
g++ -DNDEBUG   -I../../include -Wmissing-prototypes -fPIC  
-DP_LINUX=2.6.5-1.358 -ffunction-sections -fdata-sections -D_REENTRANT -Wall

  -fPIC -DP_USE_PRAGMA -DPHAS_TEMPLATES -I/root/pwlib/include/ptlib/unix 
-I/usr/include/pwlib -I/root/pwlib/include -DPTRACING 
-I/root/openh323/include -DHAS_IXJ -DHAS_OSS -fPIC -DP_USE_PRAGMA -Os 
-DNDEBUG -pipe -x c++ -c ast_h323.cxx -o ast_h323.o
ast_h323.cxx:1:1: warning: "_GNU_SOURCE" redefined
:4:1: warning: this is the location of the previous definition
In file included from ast_h323.cxx:51:
ast_h323.h:159: error: type specifier omitted for parameter `RTP_QOS'
ast_h323.h:159: error: syntax error before `*' token
ast_h323.cxx:957: error: type specifier omitted for parameter `RTP_QOS'
ast_h323.cxx:957: error: syntax error before `*' token
ast_h323.cxx: In member function `H323Channel*
   MyH323Connection::CreateRealTimeLogicalChannel(...)':
ast_h323.cxx:959: error: `capability' undeclared (first use this function)
ast_h323.cxx:959: error: (Each undeclared identifier is reported only once 
for
   each function it appears in.)
ast_h323.cxx:959: error: `dir' undeclared (first use this function)
ast_h323.cxx:959: error: `sessionID' undeclared (first use this function)
make: *** [ast_h323.o] Error 1



Thanks alot in advance...

_
Express yourself instantly with MSN Messenger! Download today it's FREE! 
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[Asterisk-Users] IAX2 and Queues Problem?

[Alejandro Kauffmann]

Hey everyone here's my problem.

Have a queue configured, it plays the desired recording, checks to see if
agents are logged in via agentcallback, forwards the call according to
distribution method, times out according to timeout settings, logs out the
agent that did not answer, hunts for next agent, logs the rest of the agents
out one by one when they don't answer, and drops call into voicemail as
defined when queue has no more agents.

This all works when i'm using Firefly as a SIP client.  When I setup Firefly
as an IAX2 client it rings the first agent and will not move on or log the
agent out if they don't answer within the timeout period.  It just continues
to ring the first extension.  Every SIP softphone i've tried works just fine
as does Firefly in SIP mode.  I have not tested any other IAX2 clients yet.

Quick and dirty solution is use SIP, but we switched to IAX2 when we started
seeing random SIP client crashes.  All our SIP clients would just freeze
every so often and have not seen the problem since we switched to IAX2.

Anyone seen this before with Firefly in IAX2 mode or any IAX2 client for
that matter?

Running Feb 27 HEAD.

Thanks in advance,

Alex


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RE: [Asterisk-Users] IAX2 attended transfer on 1-0-6 Stable

[Alejandro Kauffmann]

Atxfer is only available in HEAD not stable.

Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Redstone
Sent: Tuesday, May 03, 2005 12:48 PM
To: Asterisk User
Subject: [Asterisk-Users] IAX2 attended transfer on 1-0-6 Stable


Hi Guys

I'm still wrestling with trying to make IAX2 softphones do attended
transfer.

My iax.conf section has:

[XXX]
secret=mysecret
context=mycontext
host=dynamic
type=friend
callerid="Paul Redstone"
mailbox=NNN   ; notifies if mailbox has something in
qualify=no   ; checks for connection every 10 seconds
notransfer=yes 

All my dial statements in extensions.conf are like:

Dial(ZAP/g1/${ARG1},20,tTr)  with t and T so that I should be able to
transfer.

Unattended transfer works OK using # but attended does not.

Features.conf has

[featuremap] 
blindxfer => #1; Blind transfer key - this simple let syou 
transfer a call and immediately hangup
disconnect => *0   ; Disconnect 
automon => *1  ; One Touch Record 
atxfer => *2   ; Attended transfer which lets you transfer
and 
talk to someone else then hangup to transfer

Not sure if this matters as I running 1-0-6 stable.

Tried every iax2 softphone around. I'm missing something here but cannot see

what it is. Google comes up with statements about attended transfers being 
difficult but that is all.

Anyone definitive experience of softphones doing attended transfers?

TIA

Paul
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RE: [Asterisk-Users] Asterisk@Home bug

[Alejandro Kauffmann]

We installed AAH .05, tweaked it and learned more about dialplans, Queues
(not included in that version of AMP), "upgrading" to CVS Head (needed
atxfer/automon) and anything else we needed to scale AAH to our needs (75
agents 15k+ calls/day)than I believe we would have learned by simply trying
to start with asterisk.  AAH does a great job of showing people new to *
what it can do once it's all put together.  

While I agree that AAH questions tend to be simple in nature and in some
cases, as this one, not related to * mainstream installations, most of the
questions deal with AMP, zaptel, libpri, spandsp, .conf files, etc.  Please
try not to immediately dismiss AAH questions to the forums simply because
they are prefixed by "need help AAH".

Now as for this question in particular, come on guys do a bit of leg
work

>From install_addon.sh (added in .9)

echo " ---"
echo "|Installing RS-232 console on COM1  |"
echo " ---"
echo ""
echo ""
echo ""
echo ""
if ! grep ttyS0 /etc/securetty >/dev/null 2>&1; then
echo "s0:12345:respawn:/sbin/agetty -i -h -L 9600 ttyS0 vt100" >>
/etc/inittab
echo ttyS0 >> /etc/securetty
fi

Try downloading the tarball and looking at what it is that you are blindly
installing on your system everytime you download the iso and burn it.

/rant off

Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Saturday, April 30, 2005 6:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] bug


This is not that AAH mailing list, check out the fourms.



On Sat, 30 Apr 2005, Manny A. Wise wrote:

> After installation of [EMAIL PROTECTED] v1, I have an annoying message in 
> the screen, anyone know how to fix it
>
>
>
> INIT: Id "s0" respawning too fast: disable for 5 minutes
>
>
>
> Thanks
>
>
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RE: [Asterisk-Users] SUSE 9.2 and Zaptel channels

[Alejandro Kauffmann]

>Of course I am not a kernel expert, so .. please be patient.

>I am investigating on my zaptel/zapata problem.

>As the main error message asterisk quits on mentions <'/dev/zap/channel':
No such file or directory> I went
>peeking over there.

>[Asterisk Verbose Error
>Mar 13 20:43:35 WARNING[5779]: chan_zap.c:763 zt_open: Unable to open '/
>dev/zap/channel': No such file or directory
>Mar 13 20:43:35 ERROR[5779]: chan_zap.c:6208 mkintf: Unable to open channel
1: No such file or directory here = 0, >tmp->channel = 1, channel = 1 Mar 13
20:43:35 ERROR[5779]: chan_zap.c:9155 setup_zap: Unable to register channel
>'1' Mar 13 20:43:35 WARNING[5779]: loader.c:345 ast_load_resource:
>chan_zap.so: load_module failed, returning -1
>]

>As a matter of fact this is what is listed in /dev/:

>[EMAIL PROTECTED]:~/sources/voip/zaptel-1.0.6> ls /dev [] z2ram
zap1 zap2 zap3 zap4 zapchannel zapctl
>zappseudo zaptimer zero zkshim zqft0


>I edited the list to avoid a huge message. As said I am not a Linux low
level expert ... still it's striking that >Asterisk does not find a pesudo
file /zap/channel and there is something similar, ie zapchannel.

>Anyways the zaptel modules are there:

>[EMAIL PROTECTED]:~/sources/voip/zaptel-1.0.6> cat /proc/modules
wcfxo 11808 0 - Live 0xdf2a8000 wcfxs 
>27680 0 - Live 0xdf2b1000 zaptel 176772 2 wcfxo,wcfxs, Live 0xdf2ba000

>Does anybody have compiled the whole asterisk set 1.0.6 on SUSE 9.2?

>Thanks
>Aldo

Suse 9.2 uses udev.  Look for README.udev in you zaptel source directory and
follow the instructions.

Regards,
Alex


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