Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[Tzafrir Cohen]

On Fri, Oct 10, 2008 at 12:19:09AM -0500, Anthony Messina wrote:

> still, there are some concerning things that have been lingering, namely for 
> me: http://bugs.digium.com/view.php?id=13443

This is the result of an incorrect sample file in asterisk 1.6.0, that
wrongly uses a feature from 1.6.1 . The fixed sample file is the sole
change of asterisk 1.6.0.1 .

-- 
   Tzafrir Cohen
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[asterisk-users] Multicore process

[Setta Punpeng]

Hi all,

I have asterisk 1.4 installed on 2 Dual Core processors (total = 4  
cores) + CentOS5.0 x64 + 3 E1 PRI and 80 sip extension. Sometime the  
system seems to slow down during peak hours. I check with 'sar' and  
see that CPU0 has highest load while the other CPUs has very low  
loads. Since asterisk uses thread, CPU load should be shared among  
CPU? or do I need to have special setting?



11:30:01 AM   CPU %user %nice   %system   %iowait 
%steal %idle
11:40:01 AM   all  1.13  0.00 10.96  0.01   
0.00 87.90
11:40:01 AM 0  1.86  0.00 41.18  0.00   
0.00 56.95
11:40:01 AM 1  0.47  0.00  0.18  0.00   
0.00 99.36
11:40:01 AM 2  1.41  0.00  1.97  0.02   
0.00 96.59
11:40:01 AM 3  0.78  0.00  0.55  0.01   
0.00 98.67
11:50:01 AM   all  0.99  0.00 10.02  0.01   
0.00 88.98
11:50:01 AM 0  1.92  0.00 38.14  0.00   
0.00 59.94
11:50:01 AM 1  0.34  0.00  0.15  0.00   
0.00 99.52
11:50:01 AM 2  1.06  0.00  1.47  0.02   
0.00 97.46
11:50:01 AM 3  0.66  0.00  0.41  0.01   
0.00 98.93
12:00:01 PM   all  0.83  0.00  8.46  0.01   
0.00 90.70
12:00:01 PM 0  1.91  0.00 32.68  0.00   
0.00 65.41
12:00:01 PM 1  0.30  0.00  0.14  0.00   
0.00 99.55
12:00:01 PM 2  0.63  0.00  0.75  0.01   
0.00 98.60
12:00:01 PM 3  0.50  0.00  0.34  0.01   
0.00 99.15
12:10:01 PM   all  1.04  0.00  7.48  0.02   
0.00 91.46
12:10:01 PM 0  1.88  0.00 29.15  0.00   
0.00 68.97
12:10:01 PM 1  0.77  0.00  0.10  0.00   
0.00 99.12
12:10:01 PM 2  0.39  0.00  0.44  0.05   
0.00 99.13
12:10:01 PM 3  1.13  0.00  0.32  0.01   
0.00 98.54
Average:  all  0.40  0.00  4.34  0.00   
0.00 95.26



Best regards,

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Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[Anthony Messina]

On Thursday 09 October 2008 09:57:30 pm Steve Totaro wrote:
> Now I have not touched any of that code, but to me, it would have been much
> simpler to change names, then change functionality later.  Make DAHDI a
> drop in replacement for Zaptel, in fact, if memory serves me correctly that
> is what someone at Digium explained, it was merely a find and replace
> operation.

i agree with the idea that a drop in should have been created, and 
functionality built from there. i'm not sure i feel as strongly as the OP 
suggested in the subject subject line he created.

for those users of 1.6, you're now in a corner: go any higher than 1.6 beta 9 
and you need dahdi. no overlap with zaptel was created here. perhaps zaptel 
could have been kept in until 1.6.1, giving the 0.0.1 overldap :)

i've been trying to follow the devel of dahdi closely, even to the point of 
building kernel modules for fedora that *should* work with jeffrey ollie's 
tools packages, just approved by the package reviewers.

still, there are some concerning things that have been lingering, namely for 
me: http://bugs.digium.com/view.php?id=13443

well, anyway, just my two cents.

-- 
Anthony -  http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[Brendan Martens]

I assume you guys are using 1.6.0, yeah? Looks like there was some  
sort of confusion about dahdi in 1.6.0... I just saw this because of  
Sean Darcy's question about 1.6.0.1 in a different thread. This is  
from the 1.6.0.1 changelog:


2008-10-08  Russell Bryant <[EMAIL PROTECTED]>

* Asterisk 1.6.0.1 released.

* configs/chan_dahdi.conf.sample: Remove mention of configuration
  sections for defining channels in chan_dahdi.conf.  This code
  is in 1.6.1, and was not merged into 1.6.0.

Maybe that explains why dahdi won't work right? I don't know for sure,  
just noticed this and thought it may be applicable.


Brendan Martens

On Oct 9, 2008, at 11:00 PM, Tzafrir Cohen wrote:


On Thu, Oct 09, 2008 at 08:12:46PM -0400, Alex Balashov wrote:

Good point.

I have a T100P that will not be seen by DAHDI for anything, but works
fabulously with Zaptel.


Which driver does it use?

Is it shown by dahdi_hardware  / zaptel_hardware ? (for  
zaptel_hardware:

try in zaptel 1.4.12 and above) .

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

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Re: [asterisk-users] 1.6.0.1 ??

[Brendan Martens]

You can find the changelog in the the downloads area: 
http://downloads.digium.com/pub/telephony/asterisk/

Excerpted from http://downloads.digium.com/pub/telephony/asterisk/ChangeLog-1.6.0.1 
:


2008-10-08  Russell Bryant <[EMAIL PROTECTED]>

* Asterisk 1.6.0.1 released.

* configs/chan_dahdi.conf.sample: Remove mention of configuration
  sections for defining channels in chan_dahdi.conf.  This code
  is in 1.6.1, and was not merged into 1.6.0.

Brendan Martens

On Oct 9, 2008, at 11:18 PM, sean darcy wrote:


In download dated 10/9.

Bug fix? Mistake?

sean


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[asterisk-users] 1.6.0.1 ??

[sean darcy]

In download dated 10/9.

Bug fix? Mistake?

sean


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Re: [asterisk-users] Asterisk CDR Analyser

[Klaverstyn, David C]

Brilliant, many thanks.  It is now working once I changed it to the
correct table name.  Line 232 is correct as well, column 103

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Bright
Sent: Friday, 10 October 2008 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk CDR Analyser

That query appears in call-log.php around line 232.

On Thu, Oct 9, 2008 at 10:20 PM, Klaverstyn, David C
<[EMAIL PROTECTED]> wrote:
> Hi All,
>
>
>
> I'm stuck and need some help.  I have installed the Asterisk CDR
Analyser
> Version 2.0.1.  It mostly works except for the CDR Report.  I get the
> following error even though it lists the CDR details.
>
>
>
> Database error: Invalid SQL: SELECT substring(calldate,1,10) AS day,
> sum(duration) AS calltime, count(*) as nbcall FROM cdr WHERE
> UNIX_TIMESTAMP(calldate) >= UNIX_TIMESTAMP('2008-10-01') GROUP BY
> substring(calldate,1,10)
> MySQL Error: 1146 (Table 'asterisk.cdr' doesn't exist)
>
>
>
> From memory there is a file that needs to be modified that is hard
coded to
> use the table asterisk.cdr but I can't find it anywhere.
>
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Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[Tzafrir Cohen]

On Thu, Oct 09, 2008 at 08:12:46PM -0400, Alex Balashov wrote:
> Good point.
> 
> I have a T100P that will not be seen by DAHDI for anything, but works 
> fabulously with Zaptel.

Which driver does it use?

Is it shown by dahdi_hardware  / zaptel_hardware ? (for zaptel_hardware:
try in zaptel 1.4.12 and above) . 

-- 
   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] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[Steve Totaro]

On Thu, Oct 9, 2008 at 10:32 PM, sean darcy <[EMAIL PROTECTED]> wrote:

> Remco Barendse wrote:
> > The information (or lack of it) on upgrading from zaptel to that
> > @&*^QW%&^%!!!  dahdi is very frustrating.
> >
> > I cannot find anything on how to uninstall zaptel, i found an earlier
> post
> > to this list which suggested make uninstall and make remove in the zaptel
> > directory which just generates errors and does nothing (on zaptel 12.1).
> >
> > Then i install dahdi-linux and dahdi-tools and i want to start
> configuring
> > it, so i am trying dahdi_genconf like the docs suggested which generates
> > this really helpful error message :
> > /usr/sbin/dahdi_genconf: Cannot read '/etc/dahdi/genconf_parameters': No
> > such file or directory
> >
> > Also the config files and everything are much more complicated
> > for dahdi than they were for zaptel
> >
> > There was some nice documentation and examples on how to get started with
> > configuring certain devices with zaptel on the digium page, for my TDM11B
> > they only mention zaptel.
> >
> > Did anyone even try this?
> >
>
> It'll work. But it's not easy. I didn't find dahdi_genconf helpful.
>
> Post your /etc/dahdi/system.conf ( the analogue of zaptel.conf ) and
> /etc/asterisk/chan_dahdi.conf ( analogue of zapata.conf ).
>
> With some help, you'll fix this.
>
> sean
>
>
Total hindsight and thinking as a user, but the initial explanation of DAHDI
came out because someone put something out there premature and someone
noticed that Zaptel was being replaced by DAHDI.

The party line explanation from Digium was that someone owned the rights to
the zaptel name.  A calling card dealer who had been very nice to allow
Digium to continue using the Zaptel name but was at his end, so hence the
name change.

Not sure I totally buy that but whatever, my thought was it was to remove
any rights or credits from the Zapata Telephony Project and Jim Dixon.
Digium could control DAHDI exactly the way it controls Asterisk, such as the
adwords debacle, and the following thread entitled "Open Letter to Digium"
to the biz list.

Now I have not touched any of that code, but to me, it would have been much
simpler to change names, then change functionality later.  Make DAHDI a drop
in replacement for Zaptel, in fact, if memory serves me correctly that is
what someone at Digium explained, it was merely a find and replace
operation.

Again, I have not touched the code but from reading a few posts, it sounds
like significant changes have been made and significant changes need to be
made to get DAHDI working and it is far from trivial.

Zaptel was not trivial to a new user, the genconfs helped but I never much
bothered with them since I was beyond that when they were introduced.  I
already had custom zap files for pretty much any setup with a little
modification.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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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] Asterisk CDR Analyser

[Sean Bright]

That query appears in call-log.php around line 232.

On Thu, Oct 9, 2008 at 10:20 PM, Klaverstyn, David C
<[EMAIL PROTECTED]> wrote:
> Hi All,
>
>
>
> I'm stuck and need some help.  I have installed the Asterisk CDR Analyser
> Version 2.0.1.  It mostly works except for the CDR Report.  I get the
> following error even though it lists the CDR details.
>
>
>
> Database error: Invalid SQL: SELECT substring(calldate,1,10) AS day,
> sum(duration) AS calltime, count(*) as nbcall FROM cdr WHERE
> UNIX_TIMESTAMP(calldate) >= UNIX_TIMESTAMP('2008-10-01') GROUP BY
> substring(calldate,1,10)
> MySQL Error: 1146 (Table 'asterisk.cdr' doesn't exist)
>
>
>
> From memory there is a file that needs to be modified that is hard coded to
> use the table asterisk.cdr but I can't find it anywhere.
>
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Polycom 330 not dialing 4 digit extensions beginning with 11xx

[Luis Morales]

check digimap into polycom web interface and check the digmap rules
for your voip system

On Fri, Oct 10, 2008 at 9:07 PM, Ed DeHart <[EMAIL PROTECTED]> wrote:
> I have four Polycom 330 phones connected to an asterisk system.  There are
> other VoIP phones connected too.  All of the extensions are four digits
> beginning with 11.  From any of the phones, except the Polycom, picking up
> the handset to call extension 1103 for example works fine.  With the Polycom
> 330, as I press the second 1 of 1103 it stops taking input and gives me an
> error.
> I tried creating four digit extensions on another asterisk system where I
> have Polycom 501 phones connected and they too will not let me dial 1103.  I
> can dial 1203 or any other combinations of number, just not an extension
> that begins with 11.
> Any suggestions?
> -Ed
>
>
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-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible"

Leonardo Da'Vinci
-

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Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[sean darcy]

Remco Barendse wrote:
> The information (or lack of it) on upgrading from zaptel to that 
> @&*^QW%&^%!!!  dahdi is very frustrating.
> 
> I cannot find anything on how to uninstall zaptel, i found an earlier post 
> to this list which suggested make uninstall and make remove in the zaptel 
> directory which just generates errors and does nothing (on zaptel 12.1).
> 
> Then i install dahdi-linux and dahdi-tools and i want to start configuring 
> it, so i am trying dahdi_genconf like the docs suggested which generates 
> this really helpful error message :
> /usr/sbin/dahdi_genconf: Cannot read '/etc/dahdi/genconf_parameters': No 
> such file or directory
> 
> Also the config files and everything are much more complicated 
> for dahdi than they were for zaptel
> 
> There was some nice documentation and examples on how to get started with 
> configuring certain devices with zaptel on the digium page, for my TDM11B 
> they only mention zaptel.
> 
> Did anyone even try this?
> 

It'll work. But it's not easy. I didn't find dahdi_genconf helpful.

Post your /etc/dahdi/system.conf ( the analogue of zaptel.conf ) and 
/etc/asterisk/chan_dahdi.conf ( analogue of zapata.conf ).

With some help, you'll fix this.

sean


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Re: [asterisk-users] Asterisk-Panasonic TDA 600 error

[C F]

On Thu, Oct 9, 2008 at 10:45 AM, Javier Prieto Gomez
<[EMAIL PROTECTED]> wrote:
> Hi I have a Panasonic TDA600 conected with one E1 to the pstn and one PRI
> with my Asterisk using a  Digium TE220B card.
> The Panasonic is master clock and Asterisk is slave, additionally  the
> Panasonic takes the clock from the PSTN E1. This scheme works and I´m able
> to make and receive calls the problem is that some times the  Asterisk lose
> the synchronism with the Panasonic and the log shows this error:
>
>
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 32:
> Yellow Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 32
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 33:
> Yellow Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 33
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 34:
> Yellow Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 34
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 35:
> Yellow Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 35
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 36:
> Yellow Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 36
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 37:
> Yellow Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 37
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 38:
> Yellow Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 38
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 39:
> Yellow Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 39
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 40: No
> Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 40
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 41: No
> Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 41
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 42: No
> Alarm
> [Oct  1 16:22:14] NOTICE[6129] chan_zap.c: PRI got event: Alarm (4) on
> Primary D-channel of span 2
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 42
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 43: No
> Alarm
> [Oct  1 16:22:14] WARNING[6129] chan_zap.c: No D-channels available!  Using
> Primary channel 47 as D-channel anyway!
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 43
> [Oct  1 16:22:14] NOTICE[6129] chan_zap.c: PRI got event: No more alarm (5)
> on Primary D-channel of span 2
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 44: No
> Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 44
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 45: No
> Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 45
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 46: No
> Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 46
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 48: No
> Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 48
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 49: No
> Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 49
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 50: No
> Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 50
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 51: No
> Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 51
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 52: No
> Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 52
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 53: No
> Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 53
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 54: No
> Alarm
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo
> cancellation on channel 54
> [Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm

Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[Sean Bright]

On Thu, Oct 9, 2008 at 7:31 PM, Remco Barendse <[EMAIL PROTECTED]> wrote:
> The information (or lack of it) on upgrading from zaptel to that
> @&*^QW%&^%!!!  dahdi is very frustrating.
>
> I cannot find anything on how to uninstall zaptel, i found an earlier post
> to this list which suggested make uninstall and make remove in the zaptel
> directory which just generates errors and does nothing (on zaptel 12.1).

What types of errors do you encounter running 'make uninstall'?
You'll need to make sure both asterisk and zaptel are shutdown before
running make install:

 # service asterisk stop
 # service zaptel stop

> Then i install dahdi-linux and dahdi-tools and i want to start configuring
> it, so i am trying dahdi_genconf like the docs suggested which generates
> this really helpful error message :
> /usr/sbin/dahdi_genconf: Cannot read '/etc/dahdi/genconf_parameters': No
> such file or directory

Unfortunately there was a bug in the initial 2.0.0 release.  This has
since been resolved in Subversion (see more details here
http://bugs.digium.com/view.php?id=13615).

If you'd like, you can grab the latest from Subversion of both the
DAHDI Linux an DAHDI Tools packages, using the following commands:

 $ svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux
 $ svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools

> Also the config files and everything are much more complicated
> for dahdi than they were for zaptel

As far as I am aware, the format of the configuration files
(/etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf) are
basically the same as their predecessors, /etc/zaptel.conf and
/etc/asterisk/zapata.conf.  Feel free to post here with any questions
and we'll try to help out.

Sean

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[asterisk-users] Asterisk CDR Analyser

[Klaverstyn, David C]

Hi All,

 

I'm stuck and need some help.  I have installed the Asterisk CDR
Analyser Version 2.0.1.  It mostly works except for the CDR Report.  I
get the following error even though it lists the CDR details.

 

Database error: Invalid SQL: SELECT substring(calldate,1,10) AS day,
sum(duration) AS calltime, count(*) as nbcall FROM cdr WHERE
UNIX_TIMESTAMP(calldate) >= UNIX_TIMESTAMP('2008-10-01') GROUP BY
substring(calldate,1,10)
MySQL Error: 1146 (Table 'asterisk.cdr' doesn't exist)

 

>From memory there is a file that needs to be modified that is hard coded
to use the table asterisk.cdr but I can't find it anywhere.

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Re: [asterisk-users] Polycom 330 not dialing 4 digit extensions beginning with 11xx

[hin lee]

Ed,

Sounds like the digitmap on the Polycom phones is the issue.  You can read more 
about the digitmap from:

http://www.voip-info.org/wiki/view/Polycom+Phones

Digitmap reference
Example: [2-9]11|0T|011xxx.T|91[2-9]x|[1-8]xx

It means the following:

* [2-9]11: 911 rule: x11 are dialled immediately (111 is covered below by 
[1-8]xx
* 0T: Local operator rule: After dialing "0" the phone waits T seconds and 
then completes the call automatically
* 011xxx.T: International rule without prefix
* 91[2-9]x: LD rule with prefix
* 9,1[2-9]x: LD rule with prefix, gives second dialtone after 
dialing 9
* [1-8]xx: A regular 3 digit extension is dialed immediately ("9" excluded 
as a prefix) 


--- On Thu, 10/9/08, Ed DeHart <[EMAIL PROTECTED]> wrote:

> From: Ed DeHart <[EMAIL PROTECTED]>
> Subject: [asterisk-users] Polycom 330 not dialing 4 digit extensions 
> beginning with 11xx
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Date: Thursday, October 9, 2008, 6:37 PM
> I have four Polycom 330 phones connected to an asterisk
> system.  There  
> are other VoIP phones connected too.  All of the extensions
> are four  
> digits beginning with 11.  From any of the phones, except
> the Polycom,  
> picking up the handset to call extension 1103 for example
> works fine.   
> With the Polycom 330, as I press the second 1 of 1103 it
> stops taking  
> input and gives me an error.
> 
> I tried creating four digit extensions on another asterisk
> system  
> where I have Polycom 501 phones connected and they too will
> not let me  
> dial 1103.  I can dial 1203 or any other combinations of
> number, just  
> not an extension that begins with 11.
> 
> Any suggestions?
> 
>   -Ed
> 
> 
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Re: [asterisk-users] Polycom 330 not dialing 4 digit extensions beginning with 11xx

[Philipp Kempgen]

Ed DeHart schrieb:
> I have four Polycom 330 phones connected to an asterisk system.  There  
> are other VoIP phones connected too.  All of the extensions are four  
> digits beginning with 11.  From any of the phones, except the Polycom,  
> picking up the handset to call extension 1103 for example works fine.   
> With the Polycom 330, as I press the second 1 of 1103 it stops taking  
> input and gives me an error.
> 
> I tried creating four digit extensions on another asterisk system  
> where I have Polycom 501 phones connected and they too will not let me  
> dial 1103.  I can dial 1203 or any other combinations of number, just  
> not an extension that begins with 11.
> 
> Any suggestions?

Maybe your dialplan ("digit map") on the Polycom is flawed?


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Polycom 330 not dialing 4 digit extensions beginning with 11xx

[Michael Graves]

You need to investigate the digit map in the phones configuration. This
determines what dialing patterns the phone will accept. See deails
here:

http://sipx-wiki.calivia.com/index.php/Digit_Maps_used_to_Define_the_Dia
l_Plan

Michael

--Original Message Text---
From: Ed DeHart
Date: Thu, 9 Oct 2008 21:37:27 -0400

I have four Polycom 330 phones connected to an asterisk system.  There
are other VoIP phones connected too.  All of the extensions are four
digits beginning with 11.  From any of the phones, except the Polycom,
picking up the handset to call extension 1103 for example works fine. 
With the Polycom 330, as I press the second 1 of 1103 it stops taking
input and gives me an error.

I tried creating four digit extensions on another asterisk system where
I have Polycom 501 phones connected and they too will not let me dial
1103.  I can dial 1203 or any other combinations of number, just not an
extension that begins with 11.


Any suggestions?


-Ed








--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245


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[asterisk-users] Polycom 330 not dialing 4 digit extensions beginning with 11xx

[Ed DeHart]

I have four Polycom 330 phones connected to an asterisk system.  There  
are other VoIP phones connected too.  All of the extensions are four  
digits beginning with 11.  From any of the phones, except the Polycom,  
picking up the handset to call extension 1103 for example works fine.   
With the Polycom 330, as I press the second 1 of 1103 it stops taking  
input and gives me an error.


I tried creating four digit extensions on another asterisk system  
where I have Polycom 501 phones connected and they too will not let me  
dial 1103.  I can dial 1203 or any other combinations of number, just  
not an extension that begins with 11.


Any suggestions?

-Ed


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Re: [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

[Luis Morales]

Well,

Your case it's not easy. Try  with ringtimeout=8000 (zapata.conf)
option. So it's very strange no signal on hang up. I have an question
for your test. Did you hang up first in your mobile and listen on the
phone for the signal ?

There are other option to play such as:
- hanguppolarityswitch = yes (zapata.conf)


Another question, to connect analog line to digium card you make the
rj-11 connect ? there is an posibility that you have incorrect pin
out.


Regards,

Luis Morales

On Fri, Oct 10, 2008 at 5:10 PM, Mike <[EMAIL PROTECTED]> wrote:
> On Fri, Oct 10, 2008 at 08:10:39AM +1930, Luis Morales wrote:
>> Mike,
>>
>> Can you tell us :
>>
>> - asterisk version
>> - zaptel version
>>
>> When you call over this line, when you hangup did you hear an busy
>> tone ? or any class tone ? To do this test connect your lines to
>> analog phone and make a call. Let's us know the results.
>>
>> Regards,
>>
>> Luis Morales
>
> Zaptel Version: 1.2.11
>
> Asterisk 1.2.13
>
> I called my mobile from the line and hung up.  The line just went
> silent.  There were no tones.  I also watched the lamp on the phone, it
> didn't got out.  I guess this could be because the line current isn't
> dropped or maybe because of capacitance in the phone?
>
> I tried this on my BT line and when I clear down, the lamp on the phone
> goes off momentarilly and then I get a single, continuous tone.
>
> Gordon, would you mind doing this test on your line to see what happens?
>
> If not, I'll try to get hold of someone with a Telewest phone and get
> them to try it.
>
> I'm trying to work out what to expect from the line and see if that is
> consistent with what I am seeing.  Once I know what the phone line is
> meant to do, then I can work out if it is doing and what I can do with
> Asterisk to accomodate it.
>
> Mike.
>
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-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible"

Leonardo Da'Vinci
-

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Re: [asterisk-users] Transfer/Park Question.

[Daniel Hazelbaker]

On Oct 9, 2008, at 2:59 PM, Brent Davidson wrote:

> I've got a situation where I need to use a transfer to the parking lot
> as hold, but am not going to use BLF indicators on the phone to pick  
> up
> the parked calls so I need to hear the 3-digit extension after the
> transfer.  I'm using Snom 300 phones and have tried setting a
> programmable button to Key Event F_TRANSFER 700, which successfully  
> does
> the transfer but cuts off audio so you don't hear the extension to
> dial.   Same with setting a Park Orbit.  I can use the DTMF button  
> type
> to send the transfer command and then the extension but then the  
> person
> doing the parking hears all of the tones, which is annoying.
>
> Is there any way to set up the transfer silently and still get the
> parking slot extension back?

Short answer: currently no.

Medium answer: I just rolled out 60+ Snom phones (300s and 320s) and  
we do call parking with DTMF.  People were used to just hitting PARK  
and their phone displaying the park extension (old NEC system).  I  
didn't tell anybody anything except "it will speak the extension back  
to you" and nobody has complained about hearing the DTMF digits.  We  
chose a 3 digit code (#92 I believe) to try an alleviate the  
possibility of somebody accidently parking a call  while filling out a  
DTMF based form/menu system, but in theory you could assign just * to  
park and only deal with 1 tone.  Just be aware that if the user needs  
to hit * for anything else, they won't be able to use it.

Long answer: Snom phones support text messages to the phone that  
automatically display.  I am looking for a way to use that in  
conjunction with Snom's ParkOrbit feature (which does work, you just  
don't hear the extension).  Basically Asterisk would do a normal park  
and then trigger a SIP NOTIFY message to the parkING phone that says  
"Parked: 701".  The message can be cleared by the user by pressing X,  
or ideally Asterisk would auto-clear the message after 10 seconds (or  
whatever).

In theory I can do the "long answer" now with a Manager application,  
but I don't like the idea of relying on an external application.  If  
it crashes or locks up for whatever reason then suddenly people get  
parked and nobody knows where.

Also be aware that in 1.2.x and 1.4.x, if you park a call and then  
pick it up, you can't park it again.  At least not with the DTMF  
method.  I borrowed a patch from the 1.6 branch that fixes this and  
made it applicable to 1.4.20.1, well I borrowed part of it.  The  
entire patch let you configure who could park etc., I wanted both  
sides to always park so I just took the 2 or 3 lines that were needed  
for that.  If you are interested I can e-mail it to you directly.

Regards,
Daniel

> Thanks,
> Brent Davidson
>
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Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[Alex Balashov]

Good point.

I have a T100P that will not be seen by DAHDI for anything, but works 
fabulously with Zaptel.

Steve Totaro wrote:

> 
> 
> On Thu, Oct 9, 2008 at 7:31 PM, Remco Barendse <[EMAIL PROTECTED] 
> > wrote:
> 
> The information (or lack of it) on upgrading from zaptel to that
> @&*^QW%&^%!!!  dahdi is very frustrating.
> 
> I cannot find anything on how to uninstall zaptel, i found an
> earlier post
> to this list which suggested make uninstall and make remove in the
> zaptel
> directory which just generates errors and does nothing (on zaptel 12.1).
> 
> Then i install dahdi-linux and dahdi-tools and i want to start
> configuring
> it, so i am trying dahdi_genconf like the docs suggested which generates
> this really helpful error message :
> /usr/sbin/dahdi_genconf: Cannot read '/etc/dahdi/genconf_parameters': No
> such file or directory
> 
> Also the config files and everything are much more complicated
> for dahdi than they were for zaptel
> 
> There was some nice documentation and examples on how to get started
> with
> configuring certain devices with zaptel on the digium page, for my
> TDM11B
> they only mention zaptel.
> 
> Did anyone even try this?
> 
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> 
> 
> I don't have answers just a question.
> 
> DAHDI is alpha or beta code, what motivates you to upgrade so badly that 
> you are frustrating yourself so much?
> 
> -- 
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
> 
> 
> 
> 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[Steve Totaro]

On Thu, Oct 9, 2008 at 7:31 PM, Remco Barendse <[EMAIL PROTECTED]> wrote:

> The information (or lack of it) on upgrading from zaptel to that
> @&*^QW%&^%!!!  dahdi is very frustrating.
>
> I cannot find anything on how to uninstall zaptel, i found an earlier post
> to this list which suggested make uninstall and make remove in the zaptel
> directory which just generates errors and does nothing (on zaptel 12.1).
>
> Then i install dahdi-linux and dahdi-tools and i want to start configuring
> it, so i am trying dahdi_genconf like the docs suggested which generates
> this really helpful error message :
> /usr/sbin/dahdi_genconf: Cannot read '/etc/dahdi/genconf_parameters': No
> such file or directory
>
> Also the config files and everything are much more complicated
> for dahdi than they were for zaptel
>
> There was some nice documentation and examples on how to get started with
> configuring certain devices with zaptel on the digium page, for my TDM11B
> they only mention zaptel.
>
> Did anyone even try this?
>
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I don't have answers just a question.

DAHDI is alpha or beta code, what motivates you to upgrade so badly that you
are frustrating yourself so much?

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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[asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[Remco Barendse]

The information (or lack of it) on upgrading from zaptel to that 
@&*^QW%&^%!!!  dahdi is very frustrating.

I cannot find anything on how to uninstall zaptel, i found an earlier post 
to this list which suggested make uninstall and make remove in the zaptel 
directory which just generates errors and does nothing (on zaptel 12.1).

Then i install dahdi-linux and dahdi-tools and i want to start configuring 
it, so i am trying dahdi_genconf like the docs suggested which generates 
this really helpful error message :
/usr/sbin/dahdi_genconf: Cannot read '/etc/dahdi/genconf_parameters': No 
such file or directory

Also the config files and everything are much more complicated 
for dahdi than they were for zaptel

There was some nice documentation and examples on how to get started with 
configuring certain devices with zaptel on the digium page, for my TDM11B 
they only mention zaptel.

Did anyone even try this?

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Re: [asterisk-users] Asterisk 1.6.0 CDR billsec and duration not working from h extension

[Steve Murphy]

On Thu, 2008-10-09 at 14:09 -0700, Eric Chamberlain wrote:
> Can someone tell me what I am doing wrong?  Why doesn't CDR(duration)  
> or CDR(billsec) return the correct values?
> 
> cdr.conf
> 
> endbeforehexten=yes
> 
> 

Eric--

To fix a problem, I had to reshuffle things around in the
asterisk core; one involves posting the CDR just after a bridge
ends; and therefore, before the hangup exten is run. To fix this,
I inserted a loop to run the hangup exten just after the 
bridge ends; This all introduced a crash and some hopefully
short-lived problems with asterisk while we find and work out the
kinks.

I believe that Russell has held back the fixes from 1.6.0 until
the dust settles. But in 1.4 svn, the new code is in place;
I do have a crash bug report filed against it, and am waiting for
the reporter to give enough info to debug it, but I get positive
reports from most.

So, my advice is to back up to a version where the h exten works with
CDR calls;
and then skip forward when the fixes are included. You'll have to suffer
with the bugs that prompted the fixes, but if you haven't noticed them
before, you might not for a while longer.

Sorry to disturb the user community wrt to CDR's, but the bugs are
tough to kill and getting one usually has unintended, negative
side-affects. If
I knew of a better way, I'd be doing it that way... gee, I thought I 
was doing it the "better way"!

The current fixes did go to patch on bug reports, and the user community
is always invited to test these out and report. If no-one does that,
then
we stick it in and folks get problems... sorry!

murf


> extensions.conf
> 
> [macro-Dial]
> ; ${ARG1} - Dial String
> 
> exten => s,1,Dial(${ARG1},,M(post-dial))
> 
> exten => h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long,  
> billed for ${CDR(billsec)} seconds)
> 
> 
> The log shows:
> 
>  -- Executing [EMAIL PROTECTED]:1] NoOp("SIP/10.10.10.170-b7d94f78",  
> "Call was hung up - 0 seconds long, billed for 0 seconds") in new stack
> 
> 
> But cdr-csv/Master.csv has logged time values for duration and billsec:
> 
> "","510555","+410001","pop-inbound","""1510555""  
> <510555>","SIP/10.10.10.170-b7d94f78","SIP/ 
> voipprovider.com-089ae8a0","Dial","SIP/1510555:password::[EMAIL 
> PROTECTED] 
> ,,M(post-dial)","2008-10-09 20:59:00","2008-10-09  
> 20:59:03","2008-10-09 20:59:08", 
> 8,5,"ANSWERED","DOCUMENTATION","1223585940.35"
> 
> 
> --
> Eric Chamberlain

-- 
Steve Murphy
Software Developer
Digium


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[asterisk-users] Cisco 7960 sccp, Skinny and 1.4

[Wayne]

Hi All,
I'm thinking of creating a new asterisk server using the latest 1.4 
stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its 
been a while!).

My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 
loaded) and to support them better, I remember compiling in a skinny(?) 
driver to replace the (from what I could tell) basic in built sccp 
support. After digging around a little it would appear that the original 
creator of the skinny driver has not done any development for ages.

Simple question, has 1.4 got better native support for sccp now without 
having to add in anything extra to make everything work ok?, if not, is 
there a version that someone may have carried forward of the skinny 
driver that will work with 1.4?


Thank you,
Wayne.


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[asterisk-users] Transfer/Park Question.

[Brent Davidson]

I've got a situation where I need to use a transfer to the parking lot 
as hold, but am not going to use BLF indicators on the phone to pick up 
the parked calls so I need to hear the 3-digit extension after the 
transfer.  I'm using Snom 300 phones and have tried setting a 
programmable button to Key Event F_TRANSFER 700, which successfully does 
the transfer but cuts off audio so you don't hear the extension to 
dial.   Same with setting a Park Orbit.  I can use the DTMF button type 
to send the transfer command and then the extension but then the person 
doing the parking hears all of the tones, which is annoying.

Is there any way to set up the transfer silently and still get the 
parking slot extension back?

Thanks,
Brent Davidson

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Re: [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

[Mike]

On Fri, Oct 10, 2008 at 08:10:39AM +1930, Luis Morales wrote:
> Mike,
> 
> Can you tell us :
> 
> - asterisk version
> - zaptel version
> 
> When you call over this line, when you hangup did you hear an busy
> tone ? or any class tone ? To do this test connect your lines to
> analog phone and make a call. Let's us know the results.
> 
> Regards,
> 
> Luis Morales

Zaptel Version: 1.2.11

Asterisk 1.2.13

I called my mobile from the line and hung up.  The line just went
silent.  There were no tones.  I also watched the lamp on the phone, it
didn't got out.  I guess this could be because the line current isn't
dropped or maybe because of capacitance in the phone?

I tried this on my BT line and when I clear down, the lamp on the phone
goes off momentarilly and then I get a single, continuous tone.

Gordon, would you mind doing this test on your line to see what happens?

If not, I'll try to get hold of someone with a Telewest phone and get
them to try it.

I'm trying to work out what to expect from the line and see if that is
consistent with what I am seeing.  Once I know what the phone line is
meant to do, then I can work out if it is doing and what I can do with
Asterisk to accomodate it.

Mike.

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Re: [asterisk-users] Asterisk 1.6.0 CDR billsec and duration not working from h extension

[Senad Jordanovic]

Eric Chamberlain wrote:
> Can someone tell me what I am doing wrong?  Why doesn't CDR(duration)  
> or CDR(billsec) return the correct values?
> 
> cdr.conf
> 
> endbeforehexten=yes
> 
> 
> extensions.conf
> 
> [macro-Dial]
> ; ${ARG1} - Dial String
> 
> exten => s,1,Dial(${ARG1},,M(post-dial))
> 
> exten => h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long,  
> billed for ${CDR(billsec)} seconds)
> 
> 
> The log shows:
> 
>  -- Executing [EMAIL PROTECTED]:1] NoOp("SIP/10.10.10.170-b7d94f78",  
> "Call was hung up - 0 seconds long, billed for 0 seconds") in new stack
> 
> 
> But cdr-csv/Master.csv has logged time values for duration and billsec:
> 
> "","510555","+410001","pop-inbound","""1510555""  
> <510555>","SIP/10.10.10.170-b7d94f78","SIP/ 
> voipprovider.com-089ae8a0","Dial","SIP/1510555:password::[EMAIL 
> PROTECTED] 
> ,,M(post-dial)","2008-10-09 20:59:00","2008-10-09  
> 20:59:03","2008-10-09 20:59:08", 
> 8,5,"ANSWERED","DOCUMENTATION","1223585940.35"
> 
> 
> --
> Eric Chamberlain

> 

Eric,


Asterisk CDR @logging@ is just no less then short of shite. An absolute 
DEAD end.


WHO ever is in charge of Digium is not doing its JOB.



Senad
www.bicomsystems.com

(Morao sam da im kazem  :) )




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Re: [asterisk-users] SIP problems?

[Alex Balashov]

A very good idea.  I heartily endorse.

Kristian Kielhofner wrote:

> Hello everyone,
> 
>   Since I've been working with SIP more and more I've discovered there
> are still plenty of interop and configuration issues between various
> pieces of equipment in the real world.
> 
>   I enjoy helping with SIP issues in this forum and others but I
> thought it would make more sense to aggregate this information in a
> central location.  For instance, earlier today a user had a problem
> between his Cisco AS5300 and Asterisk 1.2.  The solution was fairly
> technical and not very obvious.  I was more than willing to help here
> but then I thought, wait - what if someone on a Cisco list somewhere
> has a similar problem?  What if I'm not there to read his post and
> reply?  What if he can't find it in the Asterisk archives for some
> reason?  What if he/she never gets the issue worked out?
> 
>   Today I plunked down the $9 for submityoursip.com.  My goal is to
> (eventually) have a site where interop details and implementation
> quirks between various SIP platforms can be easily searched,
> discussed, etc.
> 
>   Trying to work with OCS and Asterisk?  Need a pointer to a TCP/UDP
> SIP proxy?  Can't figure out how to get your
> Polycom/Asterisk/Cisco/Snom/Sonus equipment to agree on a codec,
> method of caller id, or DTMF mode?  This wiki should help.
> 
>   I'll be adding some more details, fixing up syntax, etc in the next
> couple of days but for now I thought I'd get the announcement out to
> see if anyone would like to help:
> 
> - Wiki formatting.  I don't know anything about MediaWiki.  Headings,
> tables, organization, etc.  Help!
> - SIP devices.  Manufacturers, service provider offerings, devices,
> etc.  I've started to make lists to (mostly) empty pages, but this
> part will never be done!
> - Debugs/SIP traces.  Have a strange interop issue?  Post the SIP and
> we'll (at least I will) take a look at it.  Maybe we can figure it out
> for you and add it to the wiki for everyone.
> 
>   One thing I don't want to do is duplicate effort elsewhere,
> copy/paste from other sites, etc.  If you can link to an external
> resource, please do!
> 
>   In case you missed it before the address is
> http://www.submityoursip.com and it's free (of course) and you can
> sign up for an account if you feel like helping me out... :)
> 
>   Thoughts?  Tips?
> 


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

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[asterisk-users] Asterisk 1.6.0 CDR billsec and duration not working from h extension

[Eric Chamberlain]

Can someone tell me what I am doing wrong?  Why doesn't CDR(duration)  
or CDR(billsec) return the correct values?

cdr.conf

endbeforehexten=yes


extensions.conf

[macro-Dial]
; ${ARG1} - Dial String

exten => s,1,Dial(${ARG1},,M(post-dial))

exten => h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long,  
billed for ${CDR(billsec)} seconds)


The log shows:

 -- Executing [EMAIL PROTECTED]:1] NoOp("SIP/10.10.10.170-b7d94f78",  
"Call was hung up - 0 seconds long, billed for 0 seconds") in new stack


But cdr-csv/Master.csv has logged time values for duration and billsec:

"","510555","+410001","pop-inbound","""1510555""  
<510555>","SIP/10.10.10.170-b7d94f78","SIP/ 
voipprovider.com-089ae8a0","Dial","SIP/1510555:password::[EMAIL PROTECTED] 
,,M(post-dial)","2008-10-09 20:59:00","2008-10-09  
20:59:03","2008-10-09 20:59:08", 
8,5,"ANSWERED","DOCUMENTATION","1223585940.35"


--
Eric Chamberlain








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[asterisk-users] SIP problems?

[Kristian Kielhofner]

Hello everyone,

  Since I've been working with SIP more and more I've discovered there
are still plenty of interop and configuration issues between various
pieces of equipment in the real world.

  I enjoy helping with SIP issues in this forum and others but I
thought it would make more sense to aggregate this information in a
central location.  For instance, earlier today a user had a problem
between his Cisco AS5300 and Asterisk 1.2.  The solution was fairly
technical and not very obvious.  I was more than willing to help here
but then I thought, wait - what if someone on a Cisco list somewhere
has a similar problem?  What if I'm not there to read his post and
reply?  What if he can't find it in the Asterisk archives for some
reason?  What if he/she never gets the issue worked out?

  Today I plunked down the $9 for submityoursip.com.  My goal is to
(eventually) have a site where interop details and implementation
quirks between various SIP platforms can be easily searched,
discussed, etc.

  Trying to work with OCS and Asterisk?  Need a pointer to a TCP/UDP
SIP proxy?  Can't figure out how to get your
Polycom/Asterisk/Cisco/Snom/Sonus equipment to agree on a codec,
method of caller id, or DTMF mode?  This wiki should help.

  I'll be adding some more details, fixing up syntax, etc in the next
couple of days but for now I thought I'd get the announcement out to
see if anyone would like to help:

- Wiki formatting.  I don't know anything about MediaWiki.  Headings,
tables, organization, etc.  Help!
- SIP devices.  Manufacturers, service provider offerings, devices,
etc.  I've started to make lists to (mostly) empty pages, but this
part will never be done!
- Debugs/SIP traces.  Have a strange interop issue?  Post the SIP and
we'll (at least I will) take a look at it.  Maybe we can figure it out
for you and add it to the wiki for everyone.

  One thing I don't want to do is duplicate effort elsewhere,
copy/paste from other sites, etc.  If you can link to an external
resource, please do!

  In case you missed it before the address is
http://www.submityoursip.com and it's free (of course) and you can
sign up for an account if you feel like helping me out... :)

  Thoughts?  Tips?

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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Re: [asterisk-users] Sample code fragement for subscribing to hints wanted (was: Re: asterisk-users Digest, Vol 51, Issue 26)

[Philipp Kempgen]

Russell Brown schrieb:
> Quoth Jared Smith...
>>On Wed, 2008-10-08 at 18:51 +0100, Russell Brown wrote:
>>> Can anyone point me at a code fragment (C would be nice) that I could
>>> use to subscribe to hints on a * box?
>>> 
>>> I'd like to write a small (hopefuly efficient) widget to show custom
>>> device states and believe that a subscription to the hint would be the
>>> most efficient but I'm very open to suggestions.
>>
>>Maybe I'm missing something here, but wouldn't it be a whole lot easier
>>to get this information via the Asterisk Manager Interface, rather than
>>writing a C program that knows how to parse SIP messages?
> 
> Yes I could do it through the Manager and a polling loop but wouldn't
> that be somewhat inefficient?  By using the SIP Hints I was hoping that
> my code would only be told when a hint changed.
> 
> Any pointers anyone?

Same thing with the manager interface. No need to poll (apart from
an initial poll after connecting). The manager ifc sends events.
ExtensionStatus, NewChannel, Hangup, Link, Unlink.

Well,
Brian Degenhardt schrieb (on asterisk-dev):
> Now that we agree that you have to write a little daemon that connects
> to Asterisk, let's explore how that's done.  For Switchvox, we've
> written just that for our Switchboard:
> 
> http://www.switchvox.com/sv?cmd=screenshots&pic=23
> 
> Currently this daemon uses the manager interface, which it turns out is
> bordering on unusable to do this sort of thing.  Don't take my word for
> it, ask anyone who's ever tried to track detailed call status over
> manager.  One of our engineers likened it to digging through your trash
> to figure out what you had for dinner, because there isn't a proper way
> to just ask.

:-)

   Philipp Kempgen

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

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

[Steve Anness]

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 


On 10/9/08 11:50 AM, "Michiel van Baak" <[EMAIL PROTECTED]> wrote:

> On 10:21, Thu 09 Oct 08, Steve Anness wrote:
>> First off thank you for your help, using your help in conjunction with a
>> couple of my own changes it partially worked.  I got rid of the iax-incoming
>> context, it seemed useless.  I may be wrong in that assumption.
>> 
>> Looking back at what I have now:
>> 
>> Extensions.conf on server A
>> 
>> [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},20)
>> 
>> Extensions.conf on Server B
>> 
>> [remote-extensions]
>> 
>> 
>> exten => _17XXX,1,Dial(SIP/17${EXTEN}-1,20)
>> exten => _17XXX,n,Voicemail(${EXTEN:0:3}|su)
>> exten => _17XXX,n,Dial(SIP/${EXTEN}-1)
>> 
>> exten => _11XXX,1,Dial(iax2/vvfarm/${EXTEN:2},20)
>> exten => _11XXX,n,Voicemail(${EXTEN:2:3}|su)
>> 
>> I can call from server B to Server A  I added the voicemail line, however;
>> it isn't working like it should.  We have things set-up, as you can see, if
>> someone dials 327 they get the voicemail box for 127.  When I dial 11127
>> from Server B it rings but when it gets time for voicemail to pick up it
>> tries calling 127 instead of 327 looking for a voicemail box.  I was under
>> the impression ${EXTEN:2:3} should cut off the 11 (that part works) and
>> change the first variable to a 3 (that part doesn't work)
>> 
>> I still can't make calls from Server A to Server B  I still get the same
>> error 
>> 
>> [Oct  9 10:26:05] NOTICE[3118]: chan_iax2.c:7773 socket_process: Rejected
>> connect attempt from 64.194.211.170, request '[EMAIL PROTECTED]'
>> does not exist
>> 
>> 
>> 17110-1 does exist, I can pick up the phone connected on server B and dial
>> 17110 and get 17110-1.
> 
> No, it doesn't.
> You only have 17XXX
> Get rid of the -1 or add the -1 to the exten on serverB
> 
>> 
>> Thanks again for your help.
>> 
>> Steve 
>> 
>> On 10/8/08 7:04 PM, "Alejandro Kauffmann" <[EMAIL PROTECTED]> wrote:
>> 
>>> 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
>>> direc

Re: [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

[robb]

I have a TDM400 working quite well, Digium dialled in and recompiled   
chan_zap with some changes , to get BT Callerid working and  I have set 
hangup on polarity in the zaptel.conf which seems to work well


this is a BT home line, not business, if you have a business line you 
should get the DCT set to 800ms and the disconnect clear should work


Robb

Gordon Henderson wrote:

On Fri, 10 Oct 2008, Luis Morales wrote:


Ok!!

Do this finale test.


Who, me or the OP (Mike). My setup works OK and I've no intention of 
doing tests, final or otherwise, thanks.


Gordon



call fron analog lines to any number, wait until the called hang up.
Now tell us the signal tone. If the signal is busy ? or any thing
else

Your set up look ok. Now try with this options into zapata.conf header:


zapata.conf:
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancelwhenbridged=no
immediate=no
faxdetect=no
busydetect=yes
busycount=4

On Fri, Oct 10, 2008 at 11:42 AM, Gordon Henderson
<[EMAIL PROTECTED]> wrote:

On Thu, 9 Oct 2008, Mike wrote:


Folks,

I've seen a few reports that people have had problems with hang up
detection on UK cable phone lines.  I have a TDM400P with two FXO 
ports,

one connected to my BT line and the other connected to my
Telewest/Virgin Media cable line.  If I ring the BT line and then 
clear

down, Asterisk detects this and acts accordingly.  If I ring the
Telewest line, the clear down is not detected, hence Asterisk 
continues

to ring extentions, record voicemail, etc.

I've seen a few posts reporting this issue with the UK cable system 
but

these are generally not resolved.

Has anyone sucessfully configured a UK Telewest line with Astersik?
Does anyone know how Telewest signals that the remote caller has 
cleared

down?


I did an instalaltion yesterday with 2 Teleworst lines.

When I was calling in on my mobile, then hanging up, it seemed to 
work ok
- it did keep the line open for a few seconds, but I see that with 
BT too.



On the same topic, how does BT signal remote clear down?


Dunno...


I have plugged a phone into the Telewest line and it doesn't appear to
receive a tone on clear down, nor does it appear to drop power on the
line.  Just in case, I changed the DCT in zaptel.h from 500ms to 
100ms and

recompiled, however, this did not resolve the issue.


I'm using asterisk 1.2.30, Zaptel 1.2.24. Oslec and an OpenVox card.


I'm working blind at the moment as I cannot find anything which
documents what to expect the Telewest line to do to signal a remote
clear down occured.  I've played around with the kewlstart and
loop-start setting but without knowing what the line is going to do,
it's difficult to know how to configure Asterisk.

Does anyone have any experience of Telewest?


Only 2 sites and they both seem to work OK - One in Bristol the 
other in

Plymouth.

Can't get caller ID to work though )-:

Zaptel.conf:

fxsks=3
fxsks=4
loadzone=uk
defaultzone=uk


zapata.conf:

usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=256
;echocancel=yes
;echotraining=yes
echocancelwhenbridged=no
immediate=no
faxdetect=no


; Channel 3: PSTN line
context=incoming
group=1
usecallerid=no
faxdetect=none
signalling=fxs_ks
rxgain=4
txgain=4
callerid=asreceived
channel => 3

; Channel 4: PSTN line
context=incoming
group=1
usecallerid=no
faxdetect=none
signalling=fxs_ks
rxgain=4
txgain=4
callerid=asreceived
channel => 4


Gordon

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--
- 


Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
- 


"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible"

Leonardo Da'Vinci
- 



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Re: [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

[Gordon Henderson]

On Fri, 10 Oct 2008, Luis Morales wrote:


Ok!!

Do this finale test.


Who, me or the OP (Mike). My setup works OK and I've no intention of doing 
tests, final or otherwise, thanks.


Gordon



call fron analog lines to any number, wait until the called hang up.
Now tell us the signal tone. If the signal is busy ? or any thing
else

Your set up look ok. Now try with this options into zapata.conf header:


zapata.conf:
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancelwhenbridged=no
immediate=no
faxdetect=no
busydetect=yes
busycount=4

On Fri, Oct 10, 2008 at 11:42 AM, Gordon Henderson
<[EMAIL PROTECTED]> wrote:

On Thu, 9 Oct 2008, Mike wrote:


Folks,

I've seen a few reports that people have had problems with hang up
detection on UK cable phone lines.  I have a TDM400P with two FXO ports,
one connected to my BT line and the other connected to my
Telewest/Virgin Media cable line.  If I ring the BT line and then clear
down, Asterisk detects this and acts accordingly.  If I ring the
Telewest line, the clear down is not detected, hence Asterisk continues
to ring extentions, record voicemail, etc.

I've seen a few posts reporting this issue with the UK cable system but
these are generally not resolved.

Has anyone sucessfully configured a UK Telewest line with Astersik?
Does anyone know how Telewest signals that the remote caller has cleared
down?


I did an instalaltion yesterday with 2 Teleworst lines.

When I was calling in on my mobile, then hanging up, it seemed to work ok
- it did keep the line open for a few seconds, but I see that with BT too.


On the same topic, how does BT signal remote clear down?


Dunno...


I have plugged a phone into the Telewest line and it doesn't appear to
receive a tone on clear down, nor does it appear to drop power on the
line.  Just in case, I changed the DCT in zaptel.h from 500ms to 100ms and
recompiled, however, this did not resolve the issue.


I'm using asterisk 1.2.30, Zaptel 1.2.24. Oslec and an OpenVox card.


I'm working blind at the moment as I cannot find anything which
documents what to expect the Telewest line to do to signal a remote
clear down occured.  I've played around with the kewlstart and
loop-start setting but without knowing what the line is going to do,
it's difficult to know how to configure Asterisk.

Does anyone have any experience of Telewest?


Only 2 sites and they both seem to work OK - One in Bristol the other in
Plymouth.

Can't get caller ID to work though )-:

Zaptel.conf:

fxsks=3
fxsks=4
loadzone=uk
defaultzone=uk


zapata.conf:

usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=256
;echocancel=yes
;echotraining=yes
echocancelwhenbridged=no
immediate=no
faxdetect=no


; Channel 3: PSTN line
context=incoming
group=1
usecallerid=no
faxdetect=none
signalling=fxs_ks
rxgain=4
txgain=4
callerid=asreceived
channel => 3

; Channel 4: PSTN line
context=incoming
group=1
usecallerid=no
faxdetect=none
signalling=fxs_ks
rxgain=4
txgain=4
callerid=asreceived
channel => 4


Gordon

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--
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible"

Leonardo Da'Vinci
-

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Re: [asterisk-users] asterisk-users Digest, Vol 51, Issue 26

[Russell Brown]

Quoth Jared Smith...
>On Wed, 2008-10-08 at 18:51 +0100, Russell Brown wrote:
>> Can anyone point me at a code fragment (C would be nice) that I could
>> use to subscribe to hints on a * box?
>> 
>> I'd like to write a small (hopefuly efficient) widget to show custom
>> device states and believe that a subscription to the hint would be the
>> most efficient but I'm very open to suggestions.
>
>Maybe I'm missing something here, but wouldn't it be a whole lot easier
>to get this information via the Asterisk Manager Interface, rather than
>writing a C program that knows how to parse SIP messages?

Yes I could do it through the Manager and a polling loop but wouldn't
that be somewhat inefficient?  By using the SIP Hints I was hoping that
my code would only be told when a hint changed.

Any pointers anyone?

-- 
 Regards,
 Russell
 
| Russell Brown  | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com  |
| Peterborough, England  | WWW Play: http://www.ruffle.me.uk |
 

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[asterisk-users] ATA hangs up with fax detection on...

[Carlos Chavez]

I have a weird problem with a client.  I recently upgraded to Asterisk
1.4.22 and Zaptel 1.4.12.1 on their server and now there is a problem
when a fax call is received.

Basically when faxdetect=incoming is set in zapata.conf the call comes
in and the fax extension dials a Linksys PAP2T where a fax machine is
connected.  The fax answers and almos immediately it hangs up and I get
this:

-- Executing [EMAIL PROTECTED]:1] Dial("Zap/2-1", "SIP/5001") in new
stack 
-- Called 5001 
-- Got SIP response 486 "Busy Here" back from 192.0.0.14 
-- SIP/5001-09495388 is busy 
  == Everyone is busy/congested at this time (1:1/0/0) 

If I set faxdetect=no then I can dial extension 5001 and the fax is
received without problems.  Any idea why fax detection would interfere
with the PAP2T?

-- 
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] Hang up detection with TDM400P and Telewest/Virgin Media line

[Luis Morales]

Ok!!

Do this finale test.

call fron analog lines to any number, wait until the called hang up.
Now tell us the signal tone. If the signal is busy ? or any thing
else

Your set up look ok. Now try with this options into zapata.conf header:


zapata.conf:
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancelwhenbridged=no
immediate=no
faxdetect=no
busydetect=yes
busycount=4

On Fri, Oct 10, 2008 at 11:42 AM, Gordon Henderson
<[EMAIL PROTECTED]> wrote:
> On Thu, 9 Oct 2008, Mike wrote:
>
>> Folks,
>>
>> I've seen a few reports that people have had problems with hang up
>> detection on UK cable phone lines.  I have a TDM400P with two FXO ports,
>> one connected to my BT line and the other connected to my
>> Telewest/Virgin Media cable line.  If I ring the BT line and then clear
>> down, Asterisk detects this and acts accordingly.  If I ring the
>> Telewest line, the clear down is not detected, hence Asterisk continues
>> to ring extentions, record voicemail, etc.
>>
>> I've seen a few posts reporting this issue with the UK cable system but
>> these are generally not resolved.
>>
>> Has anyone sucessfully configured a UK Telewest line with Astersik?
>> Does anyone know how Telewest signals that the remote caller has cleared
>> down?
>
> I did an instalaltion yesterday with 2 Teleworst lines.
>
> When I was calling in on my mobile, then hanging up, it seemed to work ok
> - it did keep the line open for a few seconds, but I see that with BT too.
>
>> On the same topic, how does BT signal remote clear down?
>
> Dunno...
>
>> I have plugged a phone into the Telewest line and it doesn't appear to
>> receive a tone on clear down, nor does it appear to drop power on the
>> line.  Just in case, I changed the DCT in zaptel.h from 500ms to 100ms and
>> recompiled, however, this did not resolve the issue.
>
> I'm using asterisk 1.2.30, Zaptel 1.2.24. Oslec and an OpenVox card.
>
>> I'm working blind at the moment as I cannot find anything which
>> documents what to expect the Telewest line to do to signal a remote
>> clear down occured.  I've played around with the kewlstart and
>> loop-start setting but without knowing what the line is going to do,
>> it's difficult to know how to configure Asterisk.
>>
>> Does anyone have any experience of Telewest?
>
> Only 2 sites and they both seem to work OK - One in Bristol the other in
> Plymouth.
>
> Can't get caller ID to work though )-:
>
> Zaptel.conf:
>
> fxsks=3
> fxsks=4
> loadzone=uk
> defaultzone=uk
>
>
> zapata.conf:
>
> usecallerid=yes
> cidsignalling=v23
> cidstart=polarity
> hidecallerid=no
> callwaiting=no
> threewaycalling=yes
> transfer=yes
> echocancel=256
> ;echocancel=yes
> ;echotraining=yes
> echocancelwhenbridged=no
> immediate=no
> faxdetect=no
>
>
> ; Channel 3: PSTN line
> context=incoming
> group=1
> usecallerid=no
> faxdetect=none
> signalling=fxs_ks
> rxgain=4
> txgain=4
> callerid=asreceived
> channel => 3
>
> ; Channel 4: PSTN line
> context=incoming
> group=1
> usecallerid=no
> faxdetect=none
> signalling=fxs_ks
> rxgain=4
> txgain=4
> callerid=asreceived
> channel => 4
>
>
> Gordon
>
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> To UNSUBSCRIBE or update options visit:
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>



-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible"

Leonardo Da'Vinci
-

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

[Michiel van Baak]

On 10:21, Thu 09 Oct 08, Steve Anness wrote:
> First off thank you for your help, using your help in conjunction with a
> couple of my own changes it partially worked.  I got rid of the iax-incoming
> context, it seemed useless.  I may be wrong in that assumption.
> 
> Looking back at what I have now:
> 
> Extensions.conf on server A
> 
> [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},20)
> 
> Extensions.conf on Server B
> 
> [remote-extensions]
> 
> 
> exten => _17XXX,1,Dial(SIP/17${EXTEN}-1,20)
> exten => _17XXX,n,Voicemail(${EXTEN:0:3}|su)
> exten => _17XXX,n,Dial(SIP/${EXTEN}-1)
> 
> exten => _11XXX,1,Dial(iax2/vvfarm/${EXTEN:2},20)
> exten => _11XXX,n,Voicemail(${EXTEN:2:3}|su)
> 
> I can call from server B to Server A  I added the voicemail line, however;
> it isn't working like it should.  We have things set-up, as you can see, if
> someone dials 327 they get the voicemail box for 127.  When I dial 11127
> from Server B it rings but when it gets time for voicemail to pick up it
> tries calling 127 instead of 327 looking for a voicemail box.  I was under
> the impression ${EXTEN:2:3} should cut off the 11 (that part works) and
> change the first variable to a 3 (that part doesn't work)
> 
> I still can't make calls from Server A to Server B  I still get the same
> error 
> 
> [Oct  9 10:26:05] NOTICE[3118]: chan_iax2.c:7773 socket_process: Rejected
> connect attempt from 64.194.211.170, request '[EMAIL PROTECTED]'
> does not exist
> 
> 
> 17110-1 does exist, I can pick up the phone connected on server B and dial
> 17110 and get 17110-1.

No, it doesn't.
You only have 17XXX
Get rid of the -1 or add the -1 to the exten on serverB

> 
> Thanks again for your help.
> 
> Steve 
> 
> On 10/8/08 7:04 PM, "Alejandro Kauffmann" <[EMAIL PROTECTED]> wrote:
> 
> > 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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> > 
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> > To UNSUBSCRIBE or update options vis

Re: [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

[Gordon Henderson]

On Thu, 9 Oct 2008, Mike wrote:

> Folks,
>
> I've seen a few reports that people have had problems with hang up
> detection on UK cable phone lines.  I have a TDM400P with two FXO ports,
> one connected to my BT line and the other connected to my
> Telewest/Virgin Media cable line.  If I ring the BT line and then clear
> down, Asterisk detects this and acts accordingly.  If I ring the
> Telewest line, the clear down is not detected, hence Asterisk continues
> to ring extentions, record voicemail, etc.
>
> I've seen a few posts reporting this issue with the UK cable system but
> these are generally not resolved.
>
> Has anyone sucessfully configured a UK Telewest line with Astersik?
> Does anyone know how Telewest signals that the remote caller has cleared
> down?

I did an instalaltion yesterday with 2 Teleworst lines.

When I was calling in on my mobile, then hanging up, it seemed to work ok 
- it did keep the line open for a few seconds, but I see that with BT too.

> On the same topic, how does BT signal remote clear down?

Dunno...

> I have plugged a phone into the Telewest line and it doesn't appear to
> receive a tone on clear down, nor does it appear to drop power on the
> line.  Just in case, I changed the DCT in zaptel.h from 500ms to 100ms and
> recompiled, however, this did not resolve the issue.

I'm using asterisk 1.2.30, Zaptel 1.2.24. Oslec and an OpenVox card.

> I'm working blind at the moment as I cannot find anything which
> documents what to expect the Telewest line to do to signal a remote
> clear down occured.  I've played around with the kewlstart and
> loop-start setting but without knowing what the line is going to do,
> it's difficult to know how to configure Asterisk.
>
> Does anyone have any experience of Telewest?

Only 2 sites and they both seem to work OK - One in Bristol the other in 
Plymouth.

Can't get caller ID to work though )-:

Zaptel.conf:

fxsks=3
fxsks=4
loadzone=uk
defaultzone=uk


zapata.conf:

usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=256
;echocancel=yes
;echotraining=yes
echocancelwhenbridged=no
immediate=no
faxdetect=no


; Channel 3: PSTN line
context=incoming
group=1
usecallerid=no
faxdetect=none
signalling=fxs_ks
rxgain=4
txgain=4
callerid=asreceived
channel => 3

; Channel 4: PSTN line
context=incoming
group=1
usecallerid=no
faxdetect=none
signalling=fxs_ks
rxgain=4
txgain=4
callerid=asreceived
channel => 4


Gordon

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Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code

[Norman Franke]

On Oct 9, 2008, at 10:40 AM, [EMAIL PROTECTED]  
wrote:

> Hi I have searched the mailing lists and come across similar threads,
> but no actual solution.  I am trying to use a Cisco AS5300 as a
> gateway for PSTNr.  I have been able to configure it to take outbound
> calls and send them to the PSTN just fine.  Inbound calls however are
> rejected by asterisk with "488 Not acceptable here" code.
>
> here are the details:
>
> AS5300:
> IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE
> SOFTWARE (fc5)


Make sure the context defined the for the cisco exists and matches the  
extensions. Try a catch all pattern: exten => _.*,1,Nop(${EXTEN})

I had some random issues and I ended up defining another entry in  
SIP.conf for the cisco by by IP address, e.g.

[172.31.2.7]
type=friend
host=172.31.2.7
insecure=very
context=cisco
qualify=2000
dtmfmode=inband

That works for me.

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



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Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code

[Kristian Kielhofner]

On 10/9/08, Ketema Harris <[EMAIL PROTECTED]> wrote:
>
> Hi I have searched the mailing lists and come across similar threads, but no
> actual solution.  I am trying to use a Cisco AS5300 as a gateway for PSTNr.
> I have been able to configure it to take outbound calls and send them to the
> PSTN just fine.  Inbound calls however are rejected by asterisk with "488
> Not acceptable here" code.
>
> here are the details:
>

...lots of details stripped...

Thanks for the all of the info.  Wading through the various debugs I
would guess that Asterisk 1.2's SDP parser does not like the multipart
INVITE from the AS5300 with MIME type multipart/mixed, followed by a
regular SDP (application/sdp) and GTD (application/gtd) (ew).

You have two choices:

1)  Disable GTD on the AS5300.  I personally don't like GTD and I
doubt you need the extra ISUP info anyway.  I don't see GTD
specifically enabled on the AS5300 but I only have experience with
AS5X50s on more recent versions of IOS so there might be a default I
don't know about...

2)  Update/upgrade to Asterisk 1.4:

http://bugs.digium.com/view.php?id=10947

  Asterisk 1.4 has much better support for multipart/mixed.  While
this bug references SIP-T, the underlying problem (multipart messages)
is the same.

  Let us know what happens.

-- 
Kristian Kielhofner
http://blog.krisk.org

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Re: [asterisk-users] cisco phones getting SIP 401 unauthorized - solved

[Jerry Geis]

Jerry Geis wrote:
>>
>> Did you check sip.conf to make sure that the port is correctly set to 
>> 5060?
>>
>> Please show the output of Cli> sip show peer  and the 
>> contents of your SEP.cnf file.
>>
>> Dave
>>  
>
>
>
>
>

This all ended up being CRAZY network stuff.

my server has 2 network cards in it. I thought I was the GW but in 
reality the customer change the setup
to point to GW device. so calls wer going OUT of the local network, then 
back in to the server.

I then made MAC address exceptions for the phones to use my server with 
two network cards as the GW for
the phones everything starting working again.

I had been setting the nat yes or enabled or whatever... I was getting 
limited success. But based on what the customer
has with the network I was surprised. Limited success I mean I had half 
channel audios...

Thanks for hte help...

Jerry

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

[Steve Anness]

First off thank you for your help, using your help in conjunction with a
couple of my own changes it partially worked.  I got rid of the iax-incoming
context, it seemed useless.  I may be wrong in that assumption.

Looking back at what I have now:

Extensions.conf on server A

[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},20)

Extensions.conf on Server B

[remote-extensions]


exten => _17XXX,1,Dial(SIP/17${EXTEN}-1,20)
exten => _17XXX,n,Voicemail(${EXTEN:0:3}|su)
exten => _17XXX,n,Dial(SIP/${EXTEN}-1)

exten => _11XXX,1,Dial(iax2/vvfarm/${EXTEN:2},20)
exten => _11XXX,n,Voicemail(${EXTEN:2:3}|su)

I can call from server B to Server A  I added the voicemail line, however;
it isn't working like it should.  We have things set-up, as you can see, if
someone dials 327 they get the voicemail box for 127.  When I dial 11127
from Server B it rings but when it gets time for voicemail to pick up it
tries calling 127 instead of 327 looking for a voicemail box.  I was under
the impression ${EXTEN:2:3} should cut off the 11 (that part works) and
change the first variable to a 3 (that part doesn't work)

I still can't make calls from Server A to Server B  I still get the same
error 

[Oct  9 10:26:05] NOTICE[3118]: chan_iax2.c:7773 socket_process: Rejected
connect attempt from 64.194.211.170, request '[EMAIL PROTECTED]'
does not exist


17110-1 does exist, I can pick up the phone connected on server B and dial
17110 and get 17110-1.

Thanks again for your help.

Steve 

On 10/8/08 7:04 PM, "Alejandro Kauffmann" <[EMAIL PROTECTED]> wrote:

> 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 won¹t 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] Tribox

[Tarek Sawah]

Using the Community Edition


--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
--

> Date: Thu, 9 Oct 2008 00:37:34 +0200
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Tribox
> 
> 
> 2008/10/6 Tarek Sawah 
> i haven't facedthse tpe of problems you mentioned with mysql.. but there is 
> one thing that you need to edit the sip.conf iax.conf or you can use the 
> sample ones in the samples folder..
> other than that.. i've been with trixbox for over three years now.. it has 
> problems with it comes to Queues and call center services.. i've been 
> struggling with it for months now .. plus "as per an earlier post i had here" 
> i'm having problems convincing trixbox to accept my dia plans on two of my 
> three servers.. while i tred installing elastix more than 10 times on 
> different machins.. i don't have those problems..
> besides!!! on trixbox you need to add th ip of the freepbx mirrors to upgrade 
> your modules.. and you have to manipulate your php files to be able to 
> upgrade your box from the website..
> 
> Is it trixbox pro or free version ?

_
See how Windows connects the people, information, and fun that are part of your 
life.
http://clk.atdmt.com/MRT/go/msnnkwxp1020093175mrt/direct/01/
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Re: [asterisk-users] Cisco 7906g & SIP

[David Gibbons]

Please send the TFTP log after using the regular factory reset method I 
described.

Thanks
Dave

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa
Sent: Thursday, October 09, 2008 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7906g & SIP

Hi Dave,
I have tried restore to factory default value (as you have recommended to
me) but without success, however also with only files:

SEP.conf file
contents of the cop file

..but the result isn't changed !
Thanks in advance.

--

   Salvatore.





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

Sent: Thursday, October 09, 2008 2:59 PM
Subject: Re: [asterisk-users] Cisco 7906g & SIP


> Sasa,
>
> Sometimes I have to do a hard reset of the phone in order to get it to
> load the SIP firmware, even when the load file is specified in the
> SEP.conf file.
>
> Make sure that only the contents of the cop file and the SEP.cnf file
> are present in your tftp root. Then unplug the phone and press and hole
> the # key. Plug the phone back in, still holding the # key. When the line
> buttons begin turn on and off in sequence, press 123456789*0#.
>
> This will factory reset the phone and should cause it to check the
> termxx.default.loads file for the proper image. It will then read the SIP
> image name from that file and flash itself with the SIP image.
>
> Dave
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Sasa
> Sent: Thursday, October 09, 2008 8:51 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Cisco 7906g & SIP
>
> Hi Dave,
> the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the
> inside
> has:
>
> apps11.1-1-3-15.sbn
> cnu11.3-1-3-15.sbn
> copstart.sh
> cvm11sip.8-0-3-16.sbn
> dsp11.1-1-3-15.sbn
> jar11sip.8-0-3-16.sbn
> load307
> load369
> SIP11.8-0-4SR1S.loads
> term06.default.loads
> term11.default.loads
>
> I use Cisco7941 without callmanager software but only with SIP support.
> Thanks.
>
> --
>
>   Salvatore.
>
>
>
> - Original Message -
> From: "David Gibbons" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Thursday, October 09, 2008 2:30 PM
> Subject: Re: [asterisk-users] Cisco 7906g & SIP
>
>
>> Sasa,
>>
>> You can actually just rename the .cop file to a .tar.gz file. Cisco
>> doesn't have (to my knowledge) any non-callmanager SIP software. The SIP
>> load is just a SIP load, not a SIP load unique to generic SIP or
>> callmanager.
>>
>> Dave
>>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of Stefan
>> Gofferje
>> Sent: Thursday, October 09, 2008 7:28 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Cisco 7906g & SIP
>>
>> Sasa schrieb:
 I need other files other than those obtained with
 cmterm-7911_7906-sip.8-0-4sr1.cop ??
>>
>> cmterm is the callmanager software. You need to get the non-callmanager
>> SIP-software. Contact your local Cisco representative to buy a license
>> for that.
>>
>> Terve,
>> Stefan
>>
>> --
>> Last words of a stormchaser:
>> "Where is that rotation on the radar?!"
>>
>>
>> ___
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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>>
>
>
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Re: [asterisk-users] Cisco 7906g & SIP

[Sasa]

Hi Dave,
I have tried restore to factory default value (as you have recommended to 
me) but without success, however also with only files:

SEP.conf file
contents of the cop file

..but the result isn't changed !
Thanks in advance.

--

   Salvatore.





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

Sent: Thursday, October 09, 2008 2:59 PM
Subject: Re: [asterisk-users] Cisco 7906g & SIP


> Sasa,
>
> Sometimes I have to do a hard reset of the phone in order to get it to 
> load the SIP firmware, even when the load file is specified in the 
> SEP.conf file.
>
> Make sure that only the contents of the cop file and the SEP.cnf file 
> are present in your tftp root. Then unplug the phone and press and hole 
> the # key. Plug the phone back in, still holding the # key. When the line 
> buttons begin turn on and off in sequence, press 123456789*0#.
>
> This will factory reset the phone and should cause it to check the 
> termxx.default.loads file for the proper image. It will then read the SIP 
> image name from that file and flash itself with the SIP image.
>
> Dave
>
> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Sasa
> Sent: Thursday, October 09, 2008 8:51 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Cisco 7906g & SIP
>
> Hi Dave,
> the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the 
> inside
> has:
>
> apps11.1-1-3-15.sbn
> cnu11.3-1-3-15.sbn
> copstart.sh
> cvm11sip.8-0-3-16.sbn
> dsp11.1-1-3-15.sbn
> jar11sip.8-0-3-16.sbn
> load307
> load369
> SIP11.8-0-4SR1S.loads
> term06.default.loads
> term11.default.loads
>
> I use Cisco7941 without callmanager software but only with SIP support.
> Thanks.
>
> --
>
>   Salvatore.
>
>
>
> - Original Message -
> From: "David Gibbons" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Thursday, October 09, 2008 2:30 PM
> Subject: Re: [asterisk-users] Cisco 7906g & SIP
>
>
>> Sasa,
>>
>> You can actually just rename the .cop file to a .tar.gz file. Cisco
>> doesn't have (to my knowledge) any non-callmanager SIP software. The SIP
>> load is just a SIP load, not a SIP load unique to generic SIP or
>> callmanager.
>>
>> Dave
>>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of Stefan
>> Gofferje
>> Sent: Thursday, October 09, 2008 7:28 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Cisco 7906g & SIP
>>
>> Sasa schrieb:
 I need other files other than those obtained with
 cmterm-7911_7906-sip.8-0-4sr1.cop ??
>>
>> cmterm is the callmanager software. You need to get the non-callmanager
>> SIP-software. Contact your local Cisco representative to buy a license
>> for that.
>>
>> Terve,
>> Stefan
>>
>> --
>> Last words of a stormchaser:
>> "Where is that rotation on the radar?!"
>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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[asterisk-users] Asterisk-Panasonic TDA 600 error

[Javier Prieto Gomez]

Hi I have a Panasonic TDA600 conected with one E1 to the pstn and one PRI with 
my Asterisk using a  Digium TE220B card.
The Panasonic is master clock and Asterisk is slave, additionally  the 
Panasonic takes the clock from the PSTN E1. This scheme works and I´m able to 
make and receive calls the problem is that some times the  Asterisk lose the 
synchronism with the Panasonic and the log shows this error: 
 
 
[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 32: 
Yellow Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo 
cancellation on channel 32[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected 
alarm on channel 33: Yellow Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: 
Unable to disable echo cancellation on channel 33[Oct  1 16:22:14] 
WARNING[6130] chan_zap.c: Detected alarm on channel 34: Yellow Alarm[Oct  1 
16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo cancellation on 
channel 34[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 
35: Yellow Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable 
echo cancellation on channel 35[Oct  1 16:22:14] WARNING[6130] chan_zap.c: 
Detected alarm on channel 36: Yellow Alarm[Oct  1 16:22:14] WARNING[6130] 
chan_zap.c: Unable to disable echo cancellation on channel 36[Oct  1 16:22:14] 
WARNING[6130] chan_zap.c: Detected alarm on channel 37: Yellow Alarm[Oct  1 
16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo cancellation on 
channel 37[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 
38: Yellow Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable 
echo cancellation on channel 38[Oct  1 16:22:14] WARNING[6130] chan_zap.c: 
Detected alarm on channel 39: Yellow Alarm[Oct  1 16:22:14] WARNING[6130] 
chan_zap.c: Unable to disable echo cancellation on channel 39[Oct  1 16:22:14] 
WARNING[6130] chan_zap.c: Detected alarm on channel 40: No Alarm[Oct  1 
16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo cancellation on 
channel 40[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 
41: No Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo 
cancellation on channel 41[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected 
alarm on channel 42: No Alarm[Oct  1 16:22:14] NOTICE[6129] chan_zap.c: PRI got 
event: Alarm (4) on Primary D-channel of span 2[Oct  1 16:22:14] WARNING[6130] 
chan_zap.c: Unable to disable echo cancellation on channel 42[Oct  1 16:22:14] 
WARNING[6130] chan_zap.c: Detected alarm on channel 43: No Alarm[Oct  1 
16:22:14] WARNING[6129] chan_zap.c: No D-channels available!  Using Primary 
channel 47 as D-channel anyway![Oct  1 16:22:14] WARNING[6130] chan_zap.c: 
Unable to disable echo cancellation on channel 43[Oct  1 16:22:14] NOTICE[6129] 
chan_zap.c: PRI got event: No more alarm (5) on Primary D-channel of span 2[Oct 
 1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 44: No 
Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo 
cancellation on channel 44[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected 
alarm on channel 45: No Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable 
to disable echo cancellation on channel 45[Oct  1 16:22:14] WARNING[6130] 
chan_zap.c: Detected alarm on channel 46: No Alarm[Oct  1 16:22:14] 
WARNING[6130] chan_zap.c: Unable to disable echo cancellation on channel 46[Oct 
 1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 48: No 
Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo 
cancellation on channel 48[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected 
alarm on channel 49: No Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable 
to disable echo cancellation on channel 49[Oct  1 16:22:14] WARNING[6130] 
chan_zap.c: Detected alarm on channel 50: No Alarm[Oct  1 16:22:14] 
WARNING[6130] chan_zap.c: Unable to disable echo cancellation on channel 50[Oct 
 1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 51: No 
Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo 
cancellation on channel 51[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected 
alarm on channel 52: No Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable 
to disable echo cancellation on channel 52[Oct  1 16:22:14] WARNING[6130] 
chan_zap.c: Detected alarm on channel 53: No Alarm[Oct  1 16:22:14] 
WARNING[6130] chan_zap.c: Unable to disable echo cancellation on channel 53[Oct 
 1 16:22:14] WARNING[6130] chan_zap.c: Detected alarm on channel 54: No 
Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable to disable echo 
cancellation on channel 54[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Detected 
alarm on channel 55: No Alarm[Oct  1 16:22:14] WARNING[6130] chan_zap.c: Unable 
to disable echo cancellation on channel 55[Oct  1 16:22:14] WARNING[6130] 
chan_zap.c: Detected alarm on channel 56: No Alarm[Oct  1 16:22:14] 
WARNING[6130] chan_zap.c: Unable to disable ech

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

[Fred Posner]

On Oct 8, 2008, at 11:41 PM, 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




Yes. Very easy way... in meetme, add the option "I" or "i".

I - records name and announces it on join and leave.
i - records name, announces it on join and leave, and allows person to  
review their recording


There's a lot of great options with meetme such as usercount, admin  
mode, etc.


Good write up here:

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


Fred Posner
[EMAIL PROTECTED]

Tel: +1 (212) 937-7844 x501
Fax: +1 (954) 252-4187

www.teamforrest.com

Main:   +1 (212) 937-7844
South:  +1 (352) 379-7334
Midwest:+1 (734) 274-8969
West:   +1 (503) 914-0999

Using VoIP?
SIP:[EMAIL PROTECTED]



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Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code

[Alex Balashov]

Not offhand / without seeing the Asterisk side.

On Thu, October 9, 2008 10:26 am, Ketema Harris wrote:
> dtmf mode was set in the sip.conf
>
> dtmfmode=rfc2833
>
> I will remove the other codecs from sip.conf and see what effect it
> has.  Do you see any other potential issues in the configs?
>
> thanks
>
>
> On Oct 9, 2008, at 9:36 AM, Alex Balashov wrote:
>
>>
>> This is due to an SDP mismatch of some sort, codec or otherwise.
>>
>> Perhaps you have not set your Asterisk SIP peers to support RFC2833
>> DTMF?  Try dtmfmode=rfc2833.  Either that, or your Asterisk SIP peers
>> are not accepting the gateway's offer of G.711u.
>>
>> Of course, I have seen interop bugs in Asterisk in the past where
>> inbound
>> calls from Cisco ISDN gateways whose SDP payload advertises a
>> different
>> preferred codec--but one that is still acceptable further down the
>> preference chain--is denied.  You may want to try to set both sides to
>> G.711u explicitly, i.e.
>>
>>  disallow=all
>>  allow=ulaw
>>
>> On the Asterisk side.  Also make sure dtmfmode is set.
>>
>> On Thu, October 9, 2008 9:25 am, Ketema Harris wrote:
>>
>>> Hi I have searched the mailing lists and come across similar threads,
>>> but no actual solution.  I am trying to use a Cisco AS5300 as a
>>> gateway for PSTNr.  I have been able to configure it to take outbound
>>> calls and send them to the PSTN just fine.  Inbound calls however are
>>> rejected by asterisk with "488 Not acceptable here" code.
>>>
>>> here are the details:
>>>
>>> AS5300:
>>> IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE
>>> SOFTWARE (fc5)
>>>
>>> Current configuration : 3939 bytes
>>>
>>> version 12.3
>>> service timestamps debug datetime msec
>>> service timestamps log datetime msec
>>> no service password-encryption
>>> !
>>> hostname K_AS5300_3
>>> !
>>> boot-start-marker
>>> boot-end-marker
>>> !
>>> enable password **
>>> !
>>> resource-pool disable
>>> clock timezone EST -5
>>> clock summer-time EDT recurring
>>> !
>>> no aaa new-model
>>> ip subnet-zero
>>> !
>>> !
>>> isdn switch-type primary-dms100
>>> !
>>> !
>>> voice service voip
>>>  sip
>>>   bind all source-interface FastEthernet0
>>>
>>> controller T1 0
>>>  framing esf
>>>  clock source line primary
>>>  linecode b8zs
>>>  pri-group timeslots 1-24
>>> !
>>> controller T1 1
>>>  framing esf
>>>  clock source line secondary 1
>>>  linecode b8zs
>>>  pri-group timeslots 1-24
>>> !
>>> controller T1 2
>>>  framing esf
>>>  linecode b8zs
>>>  pri-group timeslots 1-24
>>> !
>>> controller T1 3
>>>  framing esf
>>>  linecode b8zs
>>>  pri-group timeslots 1-24
>>> !
>>> !
>>> !
>>> interface Ethernet0
>>>  no ip address
>>>  shutdown
>>> !
>>> interface Serial0:23
>>>  no ip address
>>>  encapsulation hdlc
>>>  isdn switch-type primary-dms100
>>>  isdn incoming-voice modem 64
>>>  no cdp enable
>>> !
>>> interface Serial1:23
>>>  no ip address
>>>  encapsulation hdlc
>>>  isdn switch-type primary-dms100
>>>  isdn incoming-voice modem 64
>>>  no cdp enable
>>> !
>>> interface Serial2:23
>>>  no ip address
>>>  encapsulation hdlc
>>>  isdn switch-type primary-dms100
>>>  isdn incoming-voice modem 64
>>>  no cdp enable
>>> !
>>> interface Serial3:23
>>>  no ip address
>>>  encapsulation hdlc
>>>  isdn switch-type primary-dms100
>>>  isdn incoming-voice modem 64
>>>  no cdp enable
>>> !
>>> interface FastEthernet0
>>>  ip address 172.31.2.7 255.255.255.0
>>>  duplex auto
>>>  speed auto
>>> !
>>> ip classless
>>> ip route 0.0.0.0 0.0.0.0 172.31.2.1
>>> no ip http server
>>> !
>>> !
>>> !
>>> !
>>> !
>>> !
>>> voice-port 0:D
>>> !
>>> voice-port 1:D
>>> !
>>> voice-port 2:D
>>> !
>>> voice-port 3:D
>>> !
>>> !
>>> !
>>> dial-peer voice 100 voip
>>>  application session
>>>  destination-pattern 678...
>>>  signaling forward unconditional
>>>  session protocol sipv2
>>>  session target sip-server
>>>  session transport udp
>>>  dtmf-relay rtp-nte
>>>  codec g711ulaw
>>>  no vad
>>> !
>>> dial-peer voice 101 voip
>>>  destination-pattern 770...
>>>  progress_ind setup enable 3
>>>  session protocol sipv2
>>>  session target sip-server
>>>  session transport udp
>>>  dtmf-relay rtp-nte
>>>  codec g711ulaw
>>>  no vad
>>> !
>>> dial-peer voice 102 voip
>>>  destination-pattern 404...
>>>  progress_ind setup enable 3
>>>  session protocol sipv2
>>>  session target sip-server
>>> session transport udp
>>>  dtmf-relay rtp-nte
>>>  codec g711ulaw
>>>  no vad
>>> !
>>> dial-peer voice 103 voip
>>>  destination-pattern 470...
>>>  progress_ind setup enable 3
>>>  session protocol sipv2
>>>  session target sip-server
>>>  session transport udp
>>>  dtmf-relay rtp-nte
>>>  codec g711ulaw
>>>  no vad
>>> !
>>> dial-peer voice 200 pots
>>>  application session
>>>  incoming called-number .
>>>  destination-pattern 91..
>>>  direct-inward-dial
>>>  port 0:D
>>>  prefix 1
>>> !
>>> dial-peer voice 201 pots
>>>  application session
>>>  incoming called-number .
>>>  destination-pattern 9..
>

Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code

[Ketema Harris]

dtmf mode was set in the sip.conf

dtmfmode=rfc2833

I will remove the other codecs from sip.conf and see what effect it  
has.  Do you see any other potential issues in the configs?

thanks


On Oct 9, 2008, at 9:36 AM, Alex Balashov wrote:

>
> This is due to an SDP mismatch of some sort, codec or otherwise.
>
> Perhaps you have not set your Asterisk SIP peers to support RFC2833
> DTMF?  Try dtmfmode=rfc2833.  Either that, or your Asterisk SIP peers
> are not accepting the gateway's offer of G.711u.
>
> Of course, I have seen interop bugs in Asterisk in the past where  
> inbound
> calls from Cisco ISDN gateways whose SDP payload advertises a  
> different
> preferred codec--but one that is still acceptable further down the
> preference chain--is denied.  You may want to try to set both sides to
> G.711u explicitly, i.e.
>
>  disallow=all
>  allow=ulaw
>
> On the Asterisk side.  Also make sure dtmfmode is set.
>
> On Thu, October 9, 2008 9:25 am, Ketema Harris wrote:
>
>> Hi I have searched the mailing lists and come across similar threads,
>> but no actual solution.  I am trying to use a Cisco AS5300 as a
>> gateway for PSTNr.  I have been able to configure it to take outbound
>> calls and send them to the PSTN just fine.  Inbound calls however are
>> rejected by asterisk with "488 Not acceptable here" code.
>>
>> here are the details:
>>
>> AS5300:
>> IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE
>> SOFTWARE (fc5)
>>
>> Current configuration : 3939 bytes
>>
>> version 12.3
>> service timestamps debug datetime msec
>> service timestamps log datetime msec
>> no service password-encryption
>> !
>> hostname K_AS5300_3
>> !
>> boot-start-marker
>> boot-end-marker
>> !
>> enable password **
>> !
>> resource-pool disable
>> clock timezone EST -5
>> clock summer-time EDT recurring
>> !
>> no aaa new-model
>> ip subnet-zero
>> !
>> !
>> isdn switch-type primary-dms100
>> !
>> !
>> voice service voip
>>  sip
>>   bind all source-interface FastEthernet0
>>
>> controller T1 0
>>  framing esf
>>  clock source line primary
>>  linecode b8zs
>>  pri-group timeslots 1-24
>> !
>> controller T1 1
>>  framing esf
>>  clock source line secondary 1
>>  linecode b8zs
>>  pri-group timeslots 1-24
>> !
>> controller T1 2
>>  framing esf
>>  linecode b8zs
>>  pri-group timeslots 1-24
>> !
>> controller T1 3
>>  framing esf
>>  linecode b8zs
>>  pri-group timeslots 1-24
>> !
>> !
>> !
>> interface Ethernet0
>>  no ip address
>>  shutdown
>> !
>> interface Serial0:23
>>  no ip address
>>  encapsulation hdlc
>>  isdn switch-type primary-dms100
>>  isdn incoming-voice modem 64
>>  no cdp enable
>> !
>> interface Serial1:23
>>  no ip address
>>  encapsulation hdlc
>>  isdn switch-type primary-dms100
>>  isdn incoming-voice modem 64
>>  no cdp enable
>> !
>> interface Serial2:23
>>  no ip address
>>  encapsulation hdlc
>>  isdn switch-type primary-dms100
>>  isdn incoming-voice modem 64
>>  no cdp enable
>> !
>> interface Serial3:23
>>  no ip address
>>  encapsulation hdlc
>>  isdn switch-type primary-dms100
>>  isdn incoming-voice modem 64
>>  no cdp enable
>> !
>> interface FastEthernet0
>>  ip address 172.31.2.7 255.255.255.0
>>  duplex auto
>>  speed auto
>> !
>> ip classless
>> ip route 0.0.0.0 0.0.0.0 172.31.2.1
>> no ip http server
>> !
>> !
>> !
>> !
>> !
>> !
>> voice-port 0:D
>> !
>> voice-port 1:D
>> !
>> voice-port 2:D
>> !
>> voice-port 3:D
>> !
>> !
>> !
>> dial-peer voice 100 voip
>>  application session
>>  destination-pattern 678...
>>  signaling forward unconditional
>>  session protocol sipv2
>>  session target sip-server
>>  session transport udp
>>  dtmf-relay rtp-nte
>>  codec g711ulaw
>>  no vad
>> !
>> dial-peer voice 101 voip
>>  destination-pattern 770...
>>  progress_ind setup enable 3
>>  session protocol sipv2
>>  session target sip-server
>>  session transport udp
>>  dtmf-relay rtp-nte
>>  codec g711ulaw
>>  no vad
>> !
>> dial-peer voice 102 voip
>>  destination-pattern 404...
>>  progress_ind setup enable 3
>>  session protocol sipv2
>>  session target sip-server
>> session transport udp
>>  dtmf-relay rtp-nte
>>  codec g711ulaw
>>  no vad
>> !
>> dial-peer voice 103 voip
>>  destination-pattern 470...
>>  progress_ind setup enable 3
>>  session protocol sipv2
>>  session target sip-server
>>  session transport udp
>>  dtmf-relay rtp-nte
>>  codec g711ulaw
>>  no vad
>> !
>> dial-peer voice 200 pots
>>  application session
>>  incoming called-number .
>>  destination-pattern 91..
>>  direct-inward-dial
>>  port 0:D
>>  prefix 1
>> !
>> dial-peer voice 201 pots
>>  application session
>>  incoming called-number .
>>  destination-pattern 9..
>>  direct-inward-dial
>>  port 0:D
>> !
>> dial-peer voice 202 pots
>>  application session
>>  incoming called-number .
>>  destination-pattern 91..
>>  direct-inward-dial
>>  port 1:D
>>  prefix 1
>> !
>> dial-peer voice 203 pots
>>  application session
>>  incoming called-number .
>>  destination-patt

Re: [asterisk-users] make func_realtime work like app_realtime (1.6)

[Tilghman Lesher]

On Wednesday 08 October 2008 22:05:14 Rob Hillis wrote:
> Tilghman Lesher wrote:
> > On Wednesday 08 October 2008 13:22:25 Rob Hillis wrote:
> >> Wesley Haut wrote:
> >>> Yell at me if you will, but I hate func_realtime - it's not very
> >>> usable nor is it change-friendly (update your database and your
> >>> dialplan completely breaks).
> >>
> >> I agree completely.  As it stands, the REALTIME() function is nearly
> >> completely useless.  If Asterisk had better string manipulation
> >> functionality, it would be /marginally/ better, though still not much
> >> good.
> >>
> >> A far better approach would be to allow you to specify the specific
> >> field you want to retrieve - the same way that you do for a write.
> >> /That/ would make the function many times more useful than it currently
> >> is.
> >
> > What if I made it work with the HASH() dialplan function, similar to how
> > func_odbc works?  Keys are column names, values are the associated field
> > value
>
> I can't say I'm familiar with this method, but a quick look at "core
> show function HASH" would seem to indicate that it would be much better
> than what we have at the moment.  However, for those occasions where you
> only want to pull one variable out of realtime, I'd still like to have
> the option of specifying one field name to retrieve.
>
> Possibly the most useful method would be to return the single field if a
> field name was specified, otherwise return the array with all values in
> it.  That would allow people to pick the method most suitable for them.

Test away!  The method for retrieving a single field is called REALTIME_FIELD,
and the method for retrieving all values into a hash is called REALTIME_HASH.

http://bugs.digium.com/view.php?id=13651

-- 
Tilghman

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Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code

[Alex Balashov]

This is due to an SDP mismatch of some sort, codec or otherwise.

Perhaps you have not set your Asterisk SIP peers to support RFC2833
DTMF?  Try dtmfmode=rfc2833.  Either that, or your Asterisk SIP peers
are not accepting the gateway's offer of G.711u.

Of course, I have seen interop bugs in Asterisk in the past where inbound
calls from Cisco ISDN gateways whose SDP payload advertises a different
preferred codec--but one that is still acceptable further down the
preference chain--is denied.  You may want to try to set both sides to
G.711u explicitly, i.e.

  disallow=all
  allow=ulaw

On the Asterisk side.  Also make sure dtmfmode is set.

On Thu, October 9, 2008 9:25 am, Ketema Harris wrote:

> Hi I have searched the mailing lists and come across similar threads,
> but no actual solution.  I am trying to use a Cisco AS5300 as a
> gateway for PSTNr.  I have been able to configure it to take outbound
> calls and send them to the PSTN just fine.  Inbound calls however are
> rejected by asterisk with "488 Not acceptable here" code.
>
> here are the details:
>
> AS5300:
> IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE
> SOFTWARE (fc5)
>
> Current configuration : 3939 bytes
>
> version 12.3
> service timestamps debug datetime msec
> service timestamps log datetime msec
> no service password-encryption
> !
> hostname K_AS5300_3
> !
> boot-start-marker
> boot-end-marker
> !
> enable password **
> !
> resource-pool disable
> clock timezone EST -5
> clock summer-time EDT recurring
> !
> no aaa new-model
> ip subnet-zero
> !
> !
> isdn switch-type primary-dms100
> !
> !
> voice service voip
>   sip
>bind all source-interface FastEthernet0
>
> controller T1 0
>   framing esf
>   clock source line primary
>   linecode b8zs
>   pri-group timeslots 1-24
> !
> controller T1 1
>   framing esf
>   clock source line secondary 1
>   linecode b8zs
>   pri-group timeslots 1-24
> !
> controller T1 2
>   framing esf
>   linecode b8zs
>   pri-group timeslots 1-24
> !
> controller T1 3
>   framing esf
>   linecode b8zs
>   pri-group timeslots 1-24
> !
> !
> !
> interface Ethernet0
>   no ip address
>   shutdown
> !
> interface Serial0:23
>   no ip address
>   encapsulation hdlc
>   isdn switch-type primary-dms100
>   isdn incoming-voice modem 64
>   no cdp enable
> !
> interface Serial1:23
>   no ip address
>   encapsulation hdlc
>   isdn switch-type primary-dms100
>   isdn incoming-voice modem 64
>   no cdp enable
> !
> interface Serial2:23
>   no ip address
>   encapsulation hdlc
>   isdn switch-type primary-dms100
>   isdn incoming-voice modem 64
>   no cdp enable
> !
> interface Serial3:23
>   no ip address
>   encapsulation hdlc
>   isdn switch-type primary-dms100
>   isdn incoming-voice modem 64
>   no cdp enable
> !
> interface FastEthernet0
>   ip address 172.31.2.7 255.255.255.0
>   duplex auto
>   speed auto
> !
> ip classless
> ip route 0.0.0.0 0.0.0.0 172.31.2.1
> no ip http server
> !
> !
> !
> !
> !
> !
> voice-port 0:D
> !
> voice-port 1:D
> !
> voice-port 2:D
> !
> voice-port 3:D
> !
> !
> !
> dial-peer voice 100 voip
>   application session
>   destination-pattern 678...
>   signaling forward unconditional
>   session protocol sipv2
>   session target sip-server
>   session transport udp
>   dtmf-relay rtp-nte
>   codec g711ulaw
>   no vad
> !
> dial-peer voice 101 voip
>   destination-pattern 770...
>   progress_ind setup enable 3
>   session protocol sipv2
>   session target sip-server
>   session transport udp
>   dtmf-relay rtp-nte
>   codec g711ulaw
>   no vad
> !
> dial-peer voice 102 voip
>   destination-pattern 404...
>   progress_ind setup enable 3
>   session protocol sipv2
>   session target sip-server
> session transport udp
>   dtmf-relay rtp-nte
>   codec g711ulaw
>   no vad
> !
> dial-peer voice 103 voip
>   destination-pattern 470...
>   progress_ind setup enable 3
>   session protocol sipv2
>   session target sip-server
>   session transport udp
>   dtmf-relay rtp-nte
>   codec g711ulaw
>   no vad
> !
> dial-peer voice 200 pots
>   application session
>   incoming called-number .
>   destination-pattern 91..
>   direct-inward-dial
>   port 0:D
>   prefix 1
> !
> dial-peer voice 201 pots
>   application session
>   incoming called-number .
>   destination-pattern 9..
>   direct-inward-dial
>   port 0:D
> !
> dial-peer voice 202 pots
>   application session
>   incoming called-number .
>   destination-pattern 91..
>   direct-inward-dial
>   port 1:D
>   prefix 1
> !
> dial-peer voice 203 pots
>   application session
>   incoming called-number .
>   destination-pattern 9..
>   direct-inward-dial
>   port 1:D
> !
> dial-peer voice 204 pots
>   application session
>   incoming called-number .
>   destination-pattern 91..
>   direct-inward-dial
> dial-peer voice 204 pots
>   application session
>   incoming called-number .
>   destination-pattern 91..
>   direct-inward-dial
>   port 2:D
>   prefix 1
> !
> dial-peer 

Re: [asterisk-users] cisco phones getting SIP 401 unauthorized

[[EMAIL PROTECTED]]

As a followup to my previous email, change nat_enable to "1" and reboot 
the phones.

Jerry Geis wrote:
>> Did you check sip.conf to make sure that the port is correctly set to 5060?
>>
>> Please show the output of Cli> sip show peer  and the contents 
>> of your SEP.cnf file.
>>
>> Dave
>>   
> 
> sip.conf has :
> 
> bindport=5060   ; UDP Port to bind to (SIP standard port 
> is 5060)
> bindaddr=X.X.X.X; ress to bind to (0.0.0.0 binds to all)
> srvlookup=yes   ; Enable DNS SRV lookups on outbound calls
>  
> 
> One extension cisco in sip.conf is:
> [402]
> type=friend
> dtmfmode=rfc2833
> username=402
> secret=XXX
> disallow=all
> allow=ulaw
> allow=alaw
> host=dynamic
> context=local-sip
> nat=yes
> canreinvite=no
> callerid="John Smith" <402>
> 
> sip show peer 402
>   * Name   : 402
>   Secret   : 
>   MD5Secret: 
>   Context  : smvoice-sip
>   Subscr.Cont. : 
>   Language : >
>   AMA flags: Unknown
>   CallingPres  : Presentation Allowed, Not Screened
>   Callgroup:
>   Pickupgroup  :
>   Mailbox  :
>   VM Extension : asterisk
>   LastMsgsSent : 32767/65535
>   Call limit   : 0
>   Dynamic  : Yes
>   Callerid : "John Smith" <402>
>   Expire   : -1
>   Insecure : no
>   Nat  : Always
>   ACL  : No
>   CanReinvite  : No
>   PromiscRedir : No
>   User=Phone   : No
>   Trust RPID   : No
>   Send RPID: No
>   DTMFmode : rfc2833
>   LastMsg  : 0
>   ToHost   :
>   Addr->IP : (Unspecified) Port 0
>   Defaddr->IP  : 0.0.0.0 Port 5060
>   Def. Username: 402
>   SIP Options  : (none)
>   Codecs   : 0xc (ulaw|alaw)
>   Codec Order  : (ulaw,alaw)
>   Status   : Unmonitored
>   Useragent:
>   Reg. Contact :
> 
> 
> SIP Config file:
> 
> # SIP Configuration Generic File (start)
> 
> 
> # Proxy Server
> proxy1_address: "X.X.X.X"  
> proxy2_address: "X.X.X.X"  
> proxy3_address: "X.X.X.X"  
> proxy4_address: "X.X.X.X"  
> proxy5_address: "X.X.X.X"  
> proxy6_address: "X.X.X.X"  
> 
> # Line 1 Settings
> line1_name: "402" ; Line 1 Extension\User ID
> line1_displayname: "402"   ; Line 1 Display Name
> line1_authname: "402" ; Line 1 Registration Authentication
> line1_password: "402" ; Line 1 Registration Password
> 
> # Line 2 Settings
> line2_name: "403"  ; Line 2 Extension\User ID
> line2_displayname: "403"   ; Line 2 Display Name
> line2_authname: "403" ; Line 2 Registration Authentication
> line2_password: "403" ; Line 2 Registration Password
> 
> # Emergency Proxy info
> proxy_emergency: ""
> proxy_emergency_port: "5060"
> 
> # Backup Proxy info
> proxy_backup: ""
> proxy_backup_port: "5060"
> 
> # Outbound Proxy info
> outbound_proxy: ""
> outbound_proxy_port: "5060"
> 
> # NAT/Firewall Traversal
> nat_enable: "0"
> nat_address: ""
> voip_control_port: "5060"
> start_media_port: "16384"
> end_media_port:  "32766"
> nat_received_processing: "1"
> 
> # Phone Label (Text desired to be displayed in upper right corner)
> phone_label: "JDA  402  "; Has no effect on SIP messaging
> 
> # Time Zone phone will reside in
> time_zone: EST 
> 
> # Telnet Level (enable or disable the ability to telnet into this phone
> telnet_level: "0"  ; 0-Disabled (default), 1-Enabled, 2-Privileged
> 
> # Phone prompt/password for telnet/console session
> phone_prompt: "Go Away"  ; Telnet/Console Prompt
> phone_password: "cisco"  ; Telnet/Console Password
> 
> proxy_register: 1 
> 
> # Enable_VAD (1-enabled, 0-disabled)
> enable_vad: "0"
> 
> # Network Media Type (auto, full100, full10, half100, half10)
> network_media_type: "auto"
> user_info: phone
> 
> # URL for external Directory location
> #logo_url: "http://10.0.1.3/10-20logo.bmp";; URL for 
> branding logo to be used on phone display
> 
> # SIP Configuration Generic File (stop)
> 
> 
> 
> 
> 
> 
> 
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Re: [asterisk-users] cisco phones getting SIP 401 unauthorized

[[EMAIL PROTECTED]]

What did the firewall change from and to?  Did you have NAT enabled in * 
AND on the Cisco phones?  FYI, if you have NAT enabled in both places, 
it will work if you have NAT in your setup or not.  If you don't have it 
enabled in both places, then it may or may not work depending on your setup.

Matt Gibson wrote:
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
> Sent: Wednesday, October 08, 2008 10:13 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] cisco phones getting SIP 401 unauthorized
> 
>> Hi Jerry, 
>>
>> Hm, okay. We had to use md5secret (instead of secret) in the sip.conf for
>> our 7970's to get them to successfully register with asterisk. However, if
>> you had them working before then I doubt this is the issue. You can try
>> anyway though,
>>
>> http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret
>>
>> We use both secret= and md5secret= with the same password in each, one
>> encrypted and one not encrypted - this seemed to let our 7970 register.
>>
>>   
> Matt,
> 
> I looked at this and did what it says for 1 of my phones.
> Still no go.
> 
> I have a mix of polycom and cisco phones at this location and
> the polycom continue to work. The cisco are now having issues.
> The only change that had been made is the customer changed their firewall.
> 
> All addresses remained the same just a new firewall. Polycom works cisco 
> is not registering.
> 
> Any thoughts?
> 
> Jerry
> 
> 
> 
> 
> Hi Jerry, 
> 
> Hmm. We had to replace our router with one that supported SIP ALG (we went
> cisco). However, since you mention all the phones are in the LAN this
> shouldn't make a difference. 
> 
> Does the problem go away if you go back to the old firewall?
> 
> Thanks,
> Matt
> 
> 
> 
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[asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code

[Ketema Harris]

Hi I have searched the mailing lists and come across similar threads,  
but no actual solution.  I am trying to use a Cisco AS5300 as a  
gateway for PSTNr.  I have been able to configure it to take outbound  
calls and send them to the PSTN just fine.  Inbound calls however are  
rejected by asterisk with "488 Not acceptable here" code.


here are the details:

AS5300:
IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE  
SOFTWARE (fc5)


Current configuration : 3939 bytes

version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname K_AS5300_3
!
boot-start-marker
boot-end-marker
!
enable password **
!
resource-pool disable
clock timezone EST -5
clock summer-time EDT recurring
!
no aaa new-model
ip subnet-zero
!
!
isdn switch-type primary-dms100
!
!
voice service voip
 sip
  bind all source-interface FastEthernet0

controller T1 0
 framing esf
 clock source line primary
 linecode b8zs
 pri-group timeslots 1-24
!
controller T1 1
 framing esf
 clock source line secondary 1
 linecode b8zs
 pri-group timeslots 1-24
!
controller T1 2
 framing esf
 linecode b8zs
 pri-group timeslots 1-24
!
controller T1 3
 framing esf
 linecode b8zs
 pri-group timeslots 1-24
!
!
!
interface Ethernet0
 no ip address
 shutdown
!
interface Serial0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-dms100
 isdn incoming-voice modem 64
 no cdp enable
!
interface Serial1:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-dms100
 isdn incoming-voice modem 64
 no cdp enable
!
interface Serial2:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-dms100
 isdn incoming-voice modem 64
 no cdp enable
!
interface Serial3:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-dms100
 isdn incoming-voice modem 64
 no cdp enable
!
interface FastEthernet0
 ip address 172.31.2.7 255.255.255.0
 duplex auto
 speed auto
!
ip classless
ip route 0.0.0.0 0.0.0.0 172.31.2.1
no ip http server
!
!
!
!
!
!
voice-port 0:D
!
voice-port 1:D
!
voice-port 2:D
!
voice-port 3:D
!
!
!
dial-peer voice 100 voip
 application session
 destination-pattern 678...
 signaling forward unconditional
 session protocol sipv2
 session target sip-server
 session transport udp
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 101 voip
 destination-pattern 770...
 progress_ind setup enable 3
 session protocol sipv2
 session target sip-server
 session transport udp
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 102 voip
 destination-pattern 404...
 progress_ind setup enable 3
 session protocol sipv2
 session target sip-server
session transport udp
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 103 voip
 destination-pattern 470...
 progress_ind setup enable 3
 session protocol sipv2
 session target sip-server
 session transport udp
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 200 pots
 application session
 incoming called-number .
 destination-pattern 91..
 direct-inward-dial
 port 0:D
 prefix 1
!
dial-peer voice 201 pots
 application session
 incoming called-number .
 destination-pattern 9..
 direct-inward-dial
 port 0:D
!
dial-peer voice 202 pots
 application session
 incoming called-number .
 destination-pattern 91..
 direct-inward-dial
 port 1:D
 prefix 1
!
dial-peer voice 203 pots
 application session
 incoming called-number .
 destination-pattern 9..
 direct-inward-dial
 port 1:D
!
dial-peer voice 204 pots
 application session
 incoming called-number .
 destination-pattern 91..
 direct-inward-dial
dial-peer voice 204 pots
 application session
 incoming called-number .
 destination-pattern 91..
 direct-inward-dial
 port 2:D
 prefix 1
!
dial-peer voice 205 pots
 application session
 incoming called-number .
 destination-pattern 9..
 direct-inward-dial
 port 2:D
!
dial-peer voice 206 pots
 application session
 incoming called-number .
 destination-pattern 91..
 direct-inward-dial
 port 3:D
 prefix 1
!
dial-peer voice 207 pots
 application session
 incoming called-number .
 destination-pattern 9..
 direct-inward-dial
 port 3:D
!
sip-ua
 retry invite 4
 retry response 3
 retry bye 2
 retry cancel 2
 sip-server ipv4:172.31.2.29
!
!
line con 0
line aux 0
line vty 0 4
 password 
 login
!
ntp clock-period 17179848
ntp peer 192.43.244.18
end

Asterisk:
Asterisk 1.2.12.1 on a x86_64 running Linux

sip.conf:

[general]
context=default ; Default context for incoming calls
bindport=5060   ; UDP Port to bind to (SIP standard  
port is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds  
to all)
srvlookup=yes   ; Enable DNS SRV lookups on outbound  
calls


[as5300_1]
type=peer
host=172.31.2.7
permit=172.31.2.7/255.255.255.255
defaultip=172.31.2.7
disallow=all
allow=ulaw
allow=gsm
allow=alaw
nat=no
canreinvite=yes
dtmfmode=rfc2833

I have al

Re: [asterisk-users] Cisco 7906g & SIP

[David Gibbons]

Sasa,

Sometimes I have to do a hard reset of the phone in order to get it to load the 
SIP firmware, even when the load file is specified in the SEP.conf file.

Make sure that only the contents of the cop file and the SEP.cnf file are 
present in your tftp root. Then unplug the phone and press and hole the # key. 
Plug the phone back in, still holding the # key. When the line buttons begin 
turn on and off in sequence, press 123456789*0#.

This will factory reset the phone and should cause it to check the 
termxx.default.loads file for the proper image. It will then read the SIP image 
name from that file and flash itself with the SIP image.

Dave

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa
Sent: Thursday, October 09, 2008 8:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7906g & SIP

Hi Dave,
the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside
has:

apps11.1-1-3-15.sbn
cnu11.3-1-3-15.sbn
copstart.sh
cvm11sip.8-0-3-16.sbn
dsp11.1-1-3-15.sbn
jar11sip.8-0-3-16.sbn
load307
load369
SIP11.8-0-4SR1S.loads
term06.default.loads
term11.default.loads

I use Cisco7941 without callmanager software but only with SIP support.
Thanks.

--

   Salvatore.



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

Sent: Thursday, October 09, 2008 2:30 PM
Subject: Re: [asterisk-users] Cisco 7906g & SIP


> Sasa,
>
> You can actually just rename the .cop file to a .tar.gz file. Cisco
> doesn't have (to my knowledge) any non-callmanager SIP software. The SIP
> load is just a SIP load, not a SIP load unique to generic SIP or
> callmanager.
>
> Dave
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Stefan
> Gofferje
> Sent: Thursday, October 09, 2008 7:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Cisco 7906g & SIP
>
> Sasa schrieb:
>>> I need other files other than those obtained with
>>> cmterm-7911_7906-sip.8-0-4sr1.cop ??
>
> cmterm is the callmanager software. You need to get the non-callmanager
> SIP-software. Contact your local Cisco representative to buy a license
> for that.
>
> Terve,
> Stefan
>
> --
> Last words of a stormchaser:
> "Where is that rotation on the radar?!"
>
>
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Re: [asterisk-users] Question on using DMZ

[Atis Lezdins]

On Thu, Oct 9, 2008 at 6:38 AM, C. Savinovich
<[EMAIL PROTECTED]> wrote:
>
>  I am tinkering with a new router, a Linksys wrt610n dual-band, etc.  But
> the when I connect it, the softphones(x-lite) on the computers don't even
> register.  After a couple of hours of hassle, I found out that if I dmz the
> router to the computer I am using, the softphone starts to work.  Problem
> is, there are about 6 computers in this office, all using x-lite.
>
>   Can anybody suggest what to do here to so that I can enable all 6
> computers connected to this router?

You should forward different ports for each softphone, and change
ports in each of them. As i remember, x-lite uses 5060 and 8000-8005,
so forward those to first computer, then change settings for x-lite on
second computer (5061 and 8006-8010) and forward them to second ip,
etc.

DMZ is just alias for "forward all ports to one ip", so not much use.

As alternative you can set up VPN on router and asterisk box, so
asterisk will treat all internal addresses as local.

Regards,
Atis

>
> Thanks
> CS
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie
> Sent: Wednesday, October 08, 2008 11:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] retransmitting NAT
>
> Hi,
>
> What does retransmitting NAT means? I have a client that uses SPA 942,
> and his phone sometimes cannot be called. i did a sip sebug and i keep
> on seeing retransmitting NAT.
>
> on the realtime it shows that it is registered, so when i try to call it
> , asterisk thinks it is still online so it tries to reach it instead of
> saying it's unavailable,
>
> [Oct  9 11:10:33] -- Called 103100
>
> it stops there until it reached the timeout i set then it will say
> unavailable.
>
> is there a way that realtime will know that the phone is not registered
> anymore? or what could be causing the retransmitting of NAT? has anyone
> encountered the same prob? thank you
>
> regards,
> nhadie
>
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-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] Cisco 7906g & SIP

[Sasa]

Hi Dave,
the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside 
has:

apps11.1-1-3-15.sbn
cnu11.3-1-3-15.sbn
copstart.sh
cvm11sip.8-0-3-16.sbn
dsp11.1-1-3-15.sbn
jar11sip.8-0-3-16.sbn
load307
load369
SIP11.8-0-4SR1S.loads
term06.default.loads
term11.default.loads

I use Cisco7941 without callmanager software but only with SIP support.
Thanks.

--

   Salvatore.



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

Sent: Thursday, October 09, 2008 2:30 PM
Subject: Re: [asterisk-users] Cisco 7906g & SIP


> Sasa,
>
> You can actually just rename the .cop file to a .tar.gz file. Cisco 
> doesn't have (to my knowledge) any non-callmanager SIP software. The SIP 
> load is just a SIP load, not a SIP load unique to generic SIP or 
> callmanager.
>
> Dave
>
> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Stefan 
> Gofferje
> Sent: Thursday, October 09, 2008 7:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Cisco 7906g & SIP
>
> Sasa schrieb:
>>> I need other files other than those obtained with
>>> cmterm-7911_7906-sip.8-0-4sr1.cop ??
>
> cmterm is the callmanager software. You need to get the non-callmanager
> SIP-software. Contact your local Cisco representative to buy a license
> for that.
>
> Terve,
> Stefan
>
> --
> Last words of a stormchaser:
> "Where is that rotation on the radar?!"
>
>
> ___
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Howto analyze concurrent ISDN channel usage

[Luis Morales]

Try with fop,

http://www.asternic.org/

Regards,

Luis Morales

On Fri, Oct 10, 2008 at 2:40 AM, Patrick
<[EMAIL PROTECTED]> wrote:
> Hi,
>
> Does anyone have a suggestion how I can analyze the concurrent usage of
> ISDN channels? A client complains about their clients sometimes getting
> a busy tone when trying to call them. I suspect they just need to add an
> additional ISDN2 line but I need some conclusive information that they
> are indeed maxing out their ISDN channels.
>
> Thanks,
> Patrick
>
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-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible"

Leonardo Da'Vinci
-

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Re: [asterisk-users] Ringtones for the console

[Robert Augustyn]

Thank you very much. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Julien Claassen
> Sent: Thursday, October 09, 2008 4:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Ringtones for the console
> 
> Hi!
>One further notice about ringtone6: This existed long 
> before today. It's called "schon wie-der drei-zehn To-te", 
> the dashes are there to mark syllables. The translation is: 
> "Again 13 dead people". the the melody of a German 
> radiostation's traffic news. A German commedian came up with 
> the lyrics. :-) It just had to go in here. :-)
>Kindest regards
> Julien
> 
> 
> Music was my first love and it will be my last (John Miles)
> 
>  FIND MY WEB-PROJECT AT:  
> http://ltsb.sourceforge.net the Linux TextBased Studio guide 
> === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de
> 
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Re: [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

[Luis Morales]

Mike,

Can you tell us :

- asterisk version
- zaptel version

When you call over this line, when you hangup did you hear an busy
tone ? or any class tone ? To do this test connect your lines to
analog phone and make a call. Let's us know the results.

Regards,

Luis Morales

On Fri, Oct 10, 2008 at 4:48 AM, Mike <[EMAIL PROTECTED]> wrote:
> Folks,
>
> I've seen a few reports that people have had problems with hang up
> detection on UK cable phone lines.  I have a TDM400P with two FXO ports,
> one connected to my BT line and the other connected to my
> Telewest/Virgin Media cable line.  If I ring the BT line and then clear
> down, Asterisk detects this and acts accordingly.  If I ring the
> Telewest line, the clear down is not detected, hence Asterisk continues
> to ring extentions, record voicemail, etc.
>
> I've seen a few posts reporting this issue with the UK cable system but
> these are generally not resolved.
>
> Has anyone sucessfully configured a UK Telewest line with Astersik?
> Does anyone know how Telewest signals that the remote caller has cleared
> down?
>
> On the same topic, how does BT signal remote clear down?
>
> I have plugged a phone into the Telewest line and it doesn't appear to
> receive a tone on clear down, nor does it appear to drop power on the
> line.  Just in case, I changed the DCT in zaptel.h from 500ms to 100ms and
> recompiled, however, this did not resolve the issue.
>
> I'm working blind at the moment as I cannot find anything which
> documents what to expect the Telewest line to do to signal a remote
> clear down occured.  I've played around with the kewlstart and
> loop-start setting but without knowing what the line is going to do,
> it's difficult to know how to configure Asterisk.
>
> Does anyone have any experience of Telewest?
>
> Thanks,
> Mike.
>
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-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible"

Leonardo Da'Vinci
-

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Re: [asterisk-users] Cisco 7906g & SIP

[David Gibbons]

Sasa,

You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't 
have (to my knowledge) any non-callmanager SIP software. The SIP load is just a 
SIP load, not a SIP load unique to generic SIP or callmanager.

Dave

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje
Sent: Thursday, October 09, 2008 7:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7906g & SIP

Sasa schrieb:
>> I need other files other than those obtained with
>> cmterm-7911_7906-sip.8-0-4sr1.cop ??

cmterm is the callmanager software. You need to get the non-callmanager
SIP-software. Contact your local Cisco representative to buy a license
for that.

Terve,
Stefan

--
Last words of a stormchaser:
"Where is that rotation on the radar?!"


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Re: [asterisk-users] Cisco 7906g & SIP

[Sasa]

Hi,
if possible use 7906G without callmanager software but only with SIP 
protocol support ?
Thanks.

--

   Salvatore.



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

Sent: Thursday, October 09, 2008 1:27 PM
Subject: Re: [asterisk-users] Cisco 7906g & SIP


> Sasa schrieb:
>>> I need other files other than those obtained with
>>> cmterm-7911_7906-sip.8-0-4sr1.cop ??
>
> cmterm is the callmanager software. You need to get the non-callmanager
> SIP-software. Contact your local Cisco representative to buy a license
> for that.
>
> Terve,
> Stefan
>
> -- 
> Last words of a stormchaser:
> "Where is that rotation on the radar?!"
>
>
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> 


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Re: [asterisk-users] Cisco 7906g & SIP

[Stefan Gofferje]

Sasa schrieb:
>> I need other files other than those obtained with
>> cmterm-7911_7906-sip.8-0-4sr1.cop ??

cmterm is the callmanager software. You need to get the non-callmanager
SIP-software. Contact your local Cisco representative to buy a license
for that.

Terve,
Stefan

-- 
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[asterisk-users] Asterisk restarts on call parking

[David Harty]

Hello,

I have an intermittent problem whereby the Asterisk (1.2.30.1 on fc9)
process unexpectedly stops.

It seems to occur when a call is being parked.

 

Below is an output from the full log, but as you can see it's not throwing
much light on the problem.

 

Any help with this is greatly appreciated.

 

Cheers,

Dave

 

 

Oct  8 14:37:45 DEBUG[2602] channel.c: Generator got voice, switching to
phase locked mode

Oct  8 14:37:45 DEBUG[2602] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:45 DEBUG[2615] chan_sip.c: Acked pending invite 102

Oct  8 14:37:45 DEBUG[2615] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match Found

Oct  8 14:37:45 DEBUG[2615] chan_sip.c: build_route: Contact hop:


Oct  8 14:37:45 DEBUG[5642] chan_zap.c: Requested indication -1 on channel
Zap/23-1

Oct  8 14:37:45 DEBUG[5551] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:45 DEBUG[5551] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:45 DEBUG[5551] channel.c: Scheduling timer at 160 sample
intervals

Oct  8 14:37:45 DEBUG[1167] manager.c: Manager received command
'QueueStatus'

Oct  8 14:37:46 DEBUG[19495] manager.c: Manager received command
'QueueStatus'

Oct  8 14:37:46 DEBUG[5676] channel.c: Scheduling timer at 138 sample
intervals

Oct  8 14:37:46 DEBUG[5676] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:46 DEBUG[5676] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:46 DEBUG[5676] channel.c: Scheduling timer at 160 sample
intervals

Oct  8 14:37:46 DEBUG[5676] chan_sip.c: Setting NAT on RTP to 0

Oct  8 14:37:46 DEBUG[5676] chan_sip.c: Outgoing Call for qmember.two

Oct  8 14:37:46 DEBUG[5676] channel.c: Generator got voice, switching to
phase locked mode

Oct  8 14:37:46 DEBUG[5676] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:46 DEBUG[2615] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 102: Found

Oct  8 14:37:46 DEBUG[5551] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:46 DEBUG[5551] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:46 DEBUG[5551] channel.c: Scheduling timer at 160 sample
intervals

Oct  8 14:37:47 DEBUG[5551] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:47 DEBUG[5551] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:47 DEBUG[5551] chan_sip.c: update_call_counter(chloe.darcy) -
decrement call limit counter

Oct  8 14:37:47 DEBUG[2615] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 103: Match Found

Oct  8 14:37:47 DEBUG[5642] channel.c: Got DTMF on channel
(SIP/qmember.one-0877e828)

Oct  8 14:37:47 DEBUG[5642] channel.c: Bridge stops bridging channels
Zap/23-1 and SIP/qmember.one-0877e828

Oct  8 14:37:47 DEBUG[5642] res_features.c: Feature interpret:
chan=Zap/23-1, peer=SIP/qmember.one-0877e828, sense=2, features=2

Oct  8 14:37:47 DEBUG[5642] chan_zap.c: Requested indication 16 on channel
Zap/23-1

Oct  8 14:37:47 DEBUG[5642] channel.c: Scheduling timer at 160 sample
intervals

Oct  8 14:37:47 DEBUG[5642] channel.c: Scheduling timer at 160 sample
intervals

Oct  8 14:37:48 DEBUG[5642] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:48 DEBUG[5642] channel.c: Scheduling timer at 160 sample
intervals

Oct  8 14:37:48 DEBUG[5642] channel.c: Generator got voice, switching to
phase locked mode

Oct  8 14:37:48 DEBUG[5642] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:48 DEBUG[5642] channel.c: Scheduling timer at 0 sample
intervals

Oct  8 14:37:48 DEBUG[5642] chan_zap.c: Requested indication 17 on channel
Zap/23-1

Oct  8 14:37:48 DEBUG[5642] chan_zap.c: Requested indication 16 on channel
Zap/23-1

Oct  8 14:37:48 DEBUG[5642] channel.c: Scheduling timer at 160 sample
intervals

Oct  8 14:37:48 DEBUG[5642] channel.c: Scheduling timer at 160 sample
intervals

*** SERVER STOPS

 

*** SERVER STARTS

Oct  8 14:37:52 NOTICE[5746] cdr.c: CDR simple logging enabled.

Oct  8 14:37:52 DEBUG[5746] res_config_mysql.c: MySQL RealTime Host:
127.0.0.1

Oct  8 14:37:52 DEBUG[5746] res_config_mysql.c: MySQL RealTime Port: 3306

Oct  8 14:37:52 DEBUG[5746] res_config_mysql.c: MySQL RealTime User:
asterisk

Oct  8 14:37:52 DEBUG[5746] res_config_mysql.c: MySQL RealTime Password:
asterisk

Oct  8 14:37:52 DEBUG[5746] res_config_mysql.c: MySQL RealTime: Successfully
connected to database.

Oct  8 14:37:52 NOTICE[5746] config.c: Registered Config Engine mysql

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Re: [asterisk-users] Cisco 7906g & SIP

[Sasa]

Hi, sorry for my insistence but for me is a big problem ! :-( ...someone 
have the same problem ?
Thanks in advance.

--

   Salvatore.



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

Sent: Tuesday, October 07, 2008 2:53 PM
Subject: Re: [asterisk-users] Cisco 7906g & SIP


> Hi, in tftp server I have the followings files:
>
> apps11.1-1-3-15.sbn
> cnu11.3-1-3-15.sbn
> copstart.sh
> cvm11sip.8-0-3-16.sbn
> dsp11.1-1-3-15.sbn
> jar11sip.8-0-3-16.sbn
> load307
> load369
> SIP11.8-0-4SR1S.loads
> term06.default.loads
> term11.default.loads
>
> ..and on 7906g in status menu I have:
>
> load file: sccp11.8-3-2s
> app load id: jar11sccp.8-3-1-22.sbn
> jvm load id: cvm11sccp.8-3-1-22.sbn
> os load id: cnu11.8-3-1-22.sbn
> boot load id: tnp06.3-0-1-31.bin
> dsp load id: dsp11.8-3-1-22.sbn
>
> I need other files other than those obtained with
> cmterm-7911_7906-sip.8-0-4sr1.cop ??
> Thanks in advance.
>
> --
>
>   Salvatore.
>
>
>
> - Original Message - 
> From: "Duncan Turnbull" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Tuesday, October 07, 2008 1:04 PM
> Subject: Re: [asterisk-users] Cisco 7906g & SIP
>
>
>> Are you sure you have set the 7960 to SIP?
>>
>> By default they use SCCP, so you need to go through the process of
>> changing them over, which ideally would just be done with the edits you
>> have already in the load files but generally means going back to an
>> early version of the SIP code then working upwards from there.
>>
>> You can check the current hardware in the status, if its SIP it will be
>> something like POS-0806... (I haven't got a phone handy to check) but
>> there is a reasonable amount of info on voipinfo about the process
>>
>> Cheers Duncan
>>
>> Sasa wrote:
>>> Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk
>>> 1.2.26.
>>> I have uploaded in my tftp server the firmware
>>> 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in
>>> SEPmacaddress.cnf.xml I have:
>>>
>>> SIP11.8-0-4SR1S
>>>
>>> ..but in tftp log server I have:
>>>
>>> Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving
>>> CTLSEPmacaddress.tlv to 192.168.0.155:49152
>>> Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving
>>> SEPmacaddress.cnf.xml to 192.168.0.155:49153
>>>
>>> ..and in asterisk CLI I have:
>>>
>>> -- Starting Skinny session from 192.168.0.155
>>> Device SEPmacaddress is attempting to register
>>>
>>> Now when 7906G started is loaded:
>>>
>>> load file: sccp11.8-3-2s
>>> boot load id: tnp06.3-0-1-31.bin
>>>
>>> ..why isn't loaded sip firmware ??
>>> Thanks in advance.
>>>
>>> --
>>>
>>>Salvatore.
>>>
>>>
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>
>
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[asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

[Mike]

Folks,

I've seen a few reports that people have had problems with hang up
detection on UK cable phone lines.  I have a TDM400P with two FXO ports,
one connected to my BT line and the other connected to my
Telewest/Virgin Media cable line.  If I ring the BT line and then clear
down, Asterisk detects this and acts accordingly.  If I ring the
Telewest line, the clear down is not detected, hence Asterisk continues
to ring extentions, record voicemail, etc.

I've seen a few posts reporting this issue with the UK cable system but
these are generally not resolved.

Has anyone sucessfully configured a UK Telewest line with Astersik?
Does anyone know how Telewest signals that the remote caller has cleared
down?

On the same topic, how does BT signal remote clear down?

I have plugged a phone into the Telewest line and it doesn't appear to
receive a tone on clear down, nor does it appear to drop power on the
line.  Just in case, I changed the DCT in zaptel.h from 500ms to 100ms and
recompiled, however, this did not resolve the issue.

I'm working blind at the moment as I cannot find anything which
documents what to expect the Telewest line to do to signal a remote
clear down occured.  I've played around with the kewlstart and
loop-start setting but without knowing what the line is going to do,
it's difficult to know how to configure Asterisk.

Does anyone have any experience of Telewest?

Thanks,
Mike.

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Re: [asterisk-users] Ringtones for the console

[Julien Claassen]

Hi!
   One further notice about ringtone6: This existed long before today. It's 
called "schon wie-der drei-zehn To-te", the dashes are there to mark 
syllables. The translation is: "Again 13 dead people". the the melody of a 
German radiostation's traffic news. A German commedian came up with the 
lyrics. :-) It just had to go in here. :-)
   Kindest regards
Julien


Music was my first love and it will be my last (John Miles)

 FIND MY WEB-PROJECT AT: 
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT: ===
http://www.juliencoder.de

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Re: [asterisk-users] Howto analyze concurrent ISDN channel usage

[Stefan Schmidt]

Patrick schrieb:
> Hi,
>
> Does anyone have a suggestion how I can analyze the concurrent usage of 
> ISDN channels? A client complains about their clients sometimes getting 
> a busy tone when trying to call them. I suspect they just need to add an 
> additional ISDN2 line but I need some conclusive information that they 
> are indeed maxing out their ISDN channels.
>
> Thanks,
> Patrick
>
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>   
hello,

you could use mrtg to get stats of the overall usage of the server. or
you run a script by your own.

something like this:

sudo asterisk -rx "show channels" | grep 'Dial(Zap' | wc

best regards

steve smith

-- 
Für weitere Fragen stehen wir gerne unter [EMAIL PROTECTED] oder
059944 - 2440 zur Verfügung.

Mit freundlichen Grüssen
-- 
Stefan Schmidt
Sysadmin/VOIP // [EMAIL PROTECTED] // Tel 059944-2440//
-
SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 //
A-1160 Wien // Fax 059944-9000 // www.sil.at  //
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[asterisk-users] Ringtones for the console

[Julien Claassen]

Hello Robert and all others of course! :-)
   Here are the ringtones (ring files):
http://juliencoder.de/ringtones.tar.bz2
   All are in .wav-format, once in CD-quality and once in phone-line-quality. 
All means all 6 of them. If the console/dsp only plays gsm or some other 
special format, which I don't believe, asterisk has a file conversion utility, 
that should, if necessary, do the trick.
   Any feedback is welcome. You will be downloading 2.3m, I hope this sounds 
reasonable. :-)
   Kindest regards
 Julien


Music was my first love and it will be my last (John Miles)

 FIND MY WEB-PROJECT AT: 
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT: ===
http://www.juliencoder.de

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[asterisk-users] Interrupt Asterisk's SayDigits()

[broadband Voice]

Has anyone done a modification where you can Interrupt Asterisk's
SayDigits(). This will be helpful in order to be able to interrupt an
announce and dial digits without waiting to hear all the announcements.
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Re: [asterisk-users] H323

[broadband Voice]

Yes, this has already been answered. Search previous post for
implementation.

On Thu, Oct 9, 2008 at 3:34 AM, michel freiha <[EMAIL PROTECTED]> wrote:

>  Dear all,
> Does asterisk supports H323?If yes how to enable it?
>
> Regards
>
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[asterisk-users] H323

[michel freiha]

Dear all,
Does asterisk supports H323?If yes how to enable it?

Regards
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[asterisk-users] Howto analyze concurrent ISDN channel usage

[Patrick]

Hi,

Does anyone have a suggestion how I can analyze the concurrent usage of 
ISDN channels? A client complains about their clients sometimes getting 
a busy tone when trying to call them. I suspect they just need to add an 
additional ISDN2 line but I need some conclusive information that they 
are indeed maxing out their ISDN channels.

Thanks,
Patrick

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[asterisk-users] Friday Oct 10 @12 Noon EDT VoIP Users Conference

[randulo]

Hi,

Tomorrow we'll be talking to some of the guys at
http://www.safisystems.com/ about their visual call flow and IVR
software. Also more about asterisk 1.6 changes and (we hope)
improvements, wide-band audio, and possibly the betas for X-Pro for
Asterisk and Skype channel modules.

You get there by reading this: http://bit.ly/voip or simply:

PSTN (724) 444-7444  Enter  22622# 1#

SIP [EMAIL PROTECTED] DTMF 22622# 1#

IRC.freenode.net #voip-users-conference

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