date:20060603

On Sat, 2006-06-03 at 04:01 -0400, Chris Mason (Lists) wrote:
 I have no problem with paying Digium the $10 for G729 licenses, everyone 
 has to make money. It's the administration of the licenses that sucks. I 
 experiment with different hardware a lot, and make up demo machines to 
 install for customers with available hardware. I have to put G729 
 licenses on them, usually $100 each time, and when I install the real 
 hardware for the client, I can't transfer the licenses. If I scrap that 
 machine or change the interfaces, that's a $100 loss. I believe when you 
 buy a number of licenses, that should determine how many instances you 
 can use, regardless of how you want to deploy them.
 In short, the method of enforcement is poor and leads to resentment from 
 customers. Surely Digium can construct a better system?

well you could patch your system to allow mac addr changes, which is not
always a good thing since you may have the real mac address on a
different system, you could write a library that is loaded with
LD_PRELOAD or whatever that maps certain calls used by the codec to get
the mac address so it always returns what you want (systrace is an
example of remapping calls like this although that is likely overkill
for this example).  You could change the codec binary to bypass the
check, its only a couple bytes.  All of this lets you alter the
licensing model, I am not saying to do this to avoid getting licenses in
the first place (and in fact the first two methods require a valid
license at some point) but it lets you decide which system that license
runs on.  

I didnt see a EULA that forbids this, but then I didnt look either, so
you may want to be careful with that if you choose any of these methods.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com we pay you to terminate calls with us!


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[Asterisk-Users] MWI lost after migration

2006-06-03 Thread Mimmus

Hi,
I just migrated my Asterisk installation from 1.2.1 to another server with
1.2.8. Among a lot of things, I copied the whole content of
/var/spool/asterisk/voicemail/default directory.
All is OK but now I'm not able to see MWI indication for new messages on all
my Grandstream GXP2000 phones (before migration, it worked).
Peraphs do I missed something?

Thanks
-- 
Domenico Viggiani

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Re: [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered

make an 'lsmod' and look for any old ISDN architecture modules such as hisax 
or isdn etc. There shall be no other modules loaded then hfcmulti and the 
misdn stuff. You don't need CAPI, maybe this is even the clue to your not 
working S0 card. 
Beronet provides all you need in order to get a BN8S0 card to run. Just 
download the install-misdn-mqueue script from the beronet website and the 
cardinstallation guide and everything will be fine. This is my experience. 
You do not need to patch the kernel an recompile it. Just install the kernel 
sources. Then execute the install-misdn-mqueue script. This is doing the 
rest.
Another point of failure may be the hotplug system and drivers. If you got 
BN8S0 card to run you maybe have some asterisk crashes which depend on the 
machine you are running. I have an IBM xSeries and my Debian loads some 
hotplug drivers as default. This is also something you should get rid of.
Just blacklist it in /etc/hotplug together with the old style isdn and capi. 
As far as I know chan_misdn does not support CAPI yet.

 hi
 i am experiencing some problems with the configuration of an BN8S0 Beronet
 card. I've downloaded last CVS of mISDN ans mISDNuser, i patch the kernel
 2.6.16.18 and the enabled the following:

 * ISDN support
   x x   Old ISDN4Linux  ---
   x x ---   CAPI subsystem
   x x M   CAPI2.0 support
   x x [*] Verbose reason code reporting (kernel size +=7K)
   x x [*] CAPI2.0 Middleware support (EXPERIMENTAL)
   x x M CAPI2.0 /dev/capi support
   x x [*]   CAPI2.0 filesystem support
   x x --- CAPI hardware drivers
   x x Active AVM cards  ---
   x x Active Eicon DIVA Server cards  ---
   x x Modular ISDN driver  ---

  M Support modular ISDN driver
   x x [*]   Enable memory leak debug for mISDN
   x x [*]   Support for AVM Fritz!Cards
   x x [ ]   Support for HFC PCI cards
   x x [*]   Support for HFC multiport cards (HFC-4S/8S/E1)
   x x [*] HFC multiport driver with memory mapped IO


 after the kernel recompilation and the reboot i build and install
 mISDNuser with make and make install.

 The card is recognized by the system, this is the output of lspci:

 00:0d.0 ISDN controller: Cologne Chip Designs GmbH ISDN network
 Controller [HFC-8S] (rev 01)

 and i can load hfcmulti and mISDN_dsp without problems...
 all channels are configured in TE mode, i use this modprobe:

 /sbin/modprobe hfcmulti layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf
 protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2 type=0x08
 /sbin/modprobe mISDN_dsp

 and this is the output of dmesg:

 Modular ISDN Stack core $Revision: 1.34 $
 mISDNd: kernel daemon started
 mISDNd: test event done
 mISDN: HFC-multi driver Rev. 1.41
 0 devices registered
 mISDN_dsp: Audio DSP  Rev. 1.17 (debug=0x0) EchoCancellor MG2
 mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies.


 why i get 0 devices registrered? where am i wrong?
 thanks in advance

 nik
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Re: [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered

Oh sorry, chan_misdn supports CAPI. The other question is wether mISDN itself 
provides CAPI support..

 hi
 i am experiencing some problems with the configuration of an BN8S0 Beronet
 card. I've downloaded last CVS of mISDN ans mISDNuser, i patch the kernel
 2.6.16.18 and the enabled the following:

 * ISDN support
   x x   Old ISDN4Linux  ---
   x x ---   CAPI subsystem
   x x M   CAPI2.0 support
   x x [*] Verbose reason code reporting (kernel size +=7K)
   x x [*] CAPI2.0 Middleware support (EXPERIMENTAL)
   x x M CAPI2.0 /dev/capi support
   x x [*]   CAPI2.0 filesystem support
   x x --- CAPI hardware drivers
   x x Active AVM cards  ---
   x x Active Eicon DIVA Server cards  ---
   x x Modular ISDN driver  ---

  M Support modular ISDN driver
   x x [*]   Enable memory leak debug for mISDN
   x x [*]   Support for AVM Fritz!Cards
   x x [ ]   Support for HFC PCI cards
   x x [*]   Support for HFC multiport cards (HFC-4S/8S/E1)
   x x [*] HFC multiport driver with memory mapped IO


 after the kernel recompilation and the reboot i build and install
 mISDNuser with make and make install.

 The card is recognized by the system, this is the output of lspci:

 00:0d.0 ISDN controller: Cologne Chip Designs GmbH ISDN network
 Controller [HFC-8S] (rev 01)

 and i can load hfcmulti and mISDN_dsp without problems...
 all channels are configured in TE mode, i use this modprobe:

 /sbin/modprobe hfcmulti layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf
 protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2 type=0x08
 /sbin/modprobe mISDN_dsp

 and this is the output of dmesg:

 Modular ISDN Stack core $Revision: 1.34 $
 mISDNd: kernel daemon started
 mISDNd: test event done
 mISDN: HFC-multi driver Rev. 1.41
 0 devices registered
 mISDN_dsp: Audio DSP  Rev. 1.17 (debug=0x0) EchoCancellor MG2
 mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies.


 why i get 0 devices registrered? where am i wrong?
 thanks in advance

 nik
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Re: [Asterisk-Users] MWI lost after migration

2006-06-03 Thread Thomas Kenyon

Mimmus wrote:
 Hi,
 I just migrated my Asterisk installation from 1.2.1 to another server with
 1.2.8. Among a lot of things, I copied the whole content of
 /var/spool/asterisk/voicemail/default directory.
 All is OK but now I'm not able to see MWI indication for new messages on all
 my Grandstream GXP2000 phones (before migration, it worked).
 Peraphs do I missed something?

 Thanks
   
I thought that so long as there was a mailbox= statement in the
channel definition, this always worked.


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Re: [Asterisk-Users] X100P fails to initialize

 So I took a chance with an X100P knock-off on eBay. I'm running Asterisk +
 FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and kernel
 2.6.16.16. Everything has been fine up until now.
 I compile the 1.2.5 Zaptel drivers without a problem, get the udev
 configuration in, modprobe zaptel, and finally modprobe wcfxo. At this
 point, I get the message:
 
 ZT_CHANCONFIG failed on channel 1: No such device or address (6)
 FATAL: Error running install command for wcfxo
 
 dmesg gives me:
 
 Zapata Telephony Interface Registered on major 196
 Zaptel Version: 1.2.5 Echo Canceller: MG2
 Failed to initailize DAA, giving up...
 wcfxo: probe of :00:0e.0 failed with error -5

this is a problem if the card shares interrupt with something on MOtherBOard.
configure the slot to other irq, or move the card to other irq.
lspci -v will be your friend. Check if someone uses same IRQ as motorola card.


-- 
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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Re: [Asterisk-Users] Config Revision Control

2006-06-03 Thread Tim Panton



On 2 Jun 2006, at 21:42, Douglas Garstang wrote:

Has anyone got any neat solutions for Asterisk .conf file revision  
control?


We have multiple Asterisk boxes here, that we'd like to maintain a  
_mostly_ common set of conf files on. They aren't all the same  
though. There's subtle differences. For example, in sip.conf,  
iax.conf etc, the bindaddr setting is different. Dundi.conf is very  
different between each system.


At the moment I have a file tree on a separate server, and I use  
the m4 processor to replace certain unique sections of the files. I  
have a bunch of scripts to build sip.conf etc and then rsync the  
files out to the servers. It works, mostly, but it isn't elegant.


I'd like to revision control all this. I don't know how it could be  
done with revision control though. As I said, not all the files are  
the same. I don't know if we'd run a version control client on each  
Asterisk box, or if we'd run it centrally, and then use rsync  
again, to copy the files out.


Here are some general rules we apply to this sort of config/ 
distribution/version-control set up

across multiple non-identical servers
(Mostly learnt in the web-server world, but it applies generally I  
think)


	1) Never check in derived files. Only check in the _inputs_ to M4,  
never the outputs.
	2) do the 'localization' (ie server specific config) on the target  
server itself, ideally
automatically - never depend on user input - get the server to tell  
the install script what

it needs to know.
3) don't user rsync unless all the files are identical

so you might have
checkitallout.sh
(use appropriate svn stuff - I'm a cvs user, so can't help there)
then a script that customizes and installs the files as needed,
with likes like:

cpp -Dhost=`uname -n` -Darch=`uname -m` iax.conf.generic /etc/ 
asterisk/iax.conf


or even
cpp -I`uname -n` iax.conf.generic /etc/asterisk/iax.conf
with
#include iax.conf.local

in iax.conf.generic and subdirectories for each host with their own  
iax.conf.local


Good luck.

Tim Panton
[EMAIL PROTECTED]



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Re: [Asterisk-Users] Page cmd FOP

2006-06-03 Thread Nicolás Gudiño


Hi,

On 6/1/06, Mike Clark [EMAIL PROTECTED] wrote:

We have a location with around 50 Polycom phones. Asterisk version is
1.2.1 We have implemented paging through the Polycoms, which works
great. We are now trying to get FOP  .26 going for the receptionist. It
seems to work fine, except that when someone does and overhead page,
about 3/4 of the phones will continue to show that they are on the phone
after the page is complete and hung up. It clears up for any extension
when they use that phone. Any ideas?


I will need to look at op_server.pl level 1 debug output while doing
the page until the problem shows up to see if it is a bug in FOP or
not. You can send the capture off list to me together with a
description of your problem and a copy of your op_buttons.cfg file.

You can continue asking FOP related questions in its mailing list, you
can subscribe from the webpage: http://www.asternic.org

Regards,

--
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered

2006-06-03 Thread nik600


thanks to your reply

i've also tried to use install-misdn-mqueue but:

[EMAIL PROTECTED]:/usr/src/install-misdn-mqueue# /etc/init.d/misdn-init scan
[OK] found the following devices:
card=1,0x8
[ii] run /etc/init.d/misdn-init config to store this information to
/etc/misdn-init.conf

as you can see, the scan finish succesfully, but if try to start misd i get:
[EMAIL PROTECTED]:/usr/src/install-misdn-mqueue#
/etc/init.d/misdn-init start
FATAL: Module capi not found.
FATAL: Error inserting mISDN_capi
(/lib/modules/2.6.16.18/extra/mISDN_capi.ko): Unknown symbol in
module, or unknown parameter (see dmesg)
-
Loading module(s) for your misdn-cards:
-
modprobe --ignore-install hfcmulti type=0x8
protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2
layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf poll=128 debug=0


and dmesg:
Modular ISDN Stack core $Revision: 1.34 $
mISDNd: kernel daemon started
ISDN L1 driver version 1.16
ISDN L2 driver version 1.27
mISDN: DSS1 Rev. 1.38
mISDN_dsp: Audio DSP  Rev. 1.17 (debug=0x0) EchoCancellor MG2
mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies.
mISDN_capi: Unknown symbol capi_cmd2str
mISDN_capi: Unknown symbol capi_cmsg_header
mISDN_capi: Unknown symbol detach_capi_ctr
mISDN_capi: Unknown symbol capi_cmsg2message
mISDN_capi: Unknown symbol capi_ctr_reseted
mISDN_capi: Unknown symbol capi_ctr_ready
mISDN_capi: Unknown symbol capi_message2cmsg
mISDN_capi: Unknown symbol capi_ctr_handle_message
mISDN_capi: Unknown symbol attach_capi_ctr
mISDNd: test event done
mISDN: HFC-multi driver Rev. 1.41
0 devices registered

finally, if i try to start asterisk:

[chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri))
mISDN_close: fid(18) isize(131072) inbuf(0xb7580008) irp(0xb7580008)
iend(0xb7580008)
Jun  3 08:42:31 ERROR[8296]: chan_misdn.c:3775 load_module: Unable to
initialize mISDN
Jun  3 08:42:31 WARNING[8296]: loader.c:414 __load_resource:
chan_misdn.so: load_module failed, returning -1
Jun  3 08:42:31 WARNING[8296]: loader.c:554 load_modules: Loading
module chan_misdn.so failed!
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Re: [Asterisk-Users] lspci doesn't show digium card te405p


- Bruno de Assumpção Loureiro [EMAIL PROTECTED] wrote:

 This motherboard doesn't have compatibility problem in digium list
 website. So, is it a compatibility problem or Digium card TE405p ??

That is a brand-new motherboard, so I doubt we've had any reports related to 
it. Certainly if the board does not show up at all in the lspci output then you 
either have a completely dead TE405P or there is a major compatibility issue; 
either way your best bet is to contact Digium Technical Support and let them 
help you figure it out.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Problems and questions with setting up a Feature Group D trunk to a Nortel DMS-10 switch


- John Fulton [EMAIL PROTECTED] wrote:

 I do use featdmf in the zapata.conf.
 
 In the zaptel.conf, it says to use the em setting for Feature Group 
 D, there is no featd setting in there.

Right, sorry for the confusion. It sounds like you have both files configured 
properly.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Prices of g729 codec


- Chris Mason (Lists) [EMAIL PROTECTED] wrote:

 licenses on them, usually $100 each time, and when I install the real
 
 hardware for the client, I can't transfer the licenses. If I scrap
 that 

Our support department is very accomodating when it comes to handling licensing 
issues like this; I'm surprised to hear that you can't transfer the licenses 
as we do that exact thing all the time.

If you have a specific support ticket number where you requested this and it 
was declined, please email it to me off-list.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Prices of g729 codec


- Sahil Gupta [EMAIL PROTECTED] wrote:
 We recently had around 60-80 licenses become useless because Digium 
 refused to renew the keys on that.  That was a bit of money kissed 
 goodbye.

Unless you had been clearly abusing the key licensing system, our support 
department will never refuse to enable a new registration on your license 
key(s). There is no 'renew the keys', though, since they don't expire.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Config Revision Control


- Michiel van Baak [EMAIL PROTECTED] wrote:

 Then the svn automerge thingie Kevin wrote for the asterisk
 svn tree is automerging changes to the 'common' tree to all
 the server trees.

Glad to see someone else is making use of it too :-)

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered

Do you still have the precompiled kernel installed? Try to boot to it. Install 
kernel sources and try to rerun the install-misdn-mqueue script. This is 
doing the needed stuff for you. Maybe the recompilation of your kernel caused 
the problem. I have installed three systems with misdn the last two months 
(two of them with the BN8S0) and I did not have problems in that with the 
vendor delivered precompiled kernel. To not need to recompile the kernel is 
one of the reasons they build the misdn in a kernel and a user space part. 
You can also contact beronet in order to get some support. They support their 
cards until they are correctly recognized in asterisk.

Cheers
 thanks to your reply

 i've also tried to use install-misdn-mqueue but:

 [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue# /etc/init.d/misdn-init
 scan [OK] found the following devices:
 card=1,0x8
 [ii] run /etc/init.d/misdn-init config to store this information to
 /etc/misdn-init.conf

 as you can see, the scan finish succesfully, but if try to start misd i
 get: [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue#
 /etc/init.d/misdn-init start
 FATAL: Module capi not found.
 FATAL: Error inserting mISDN_capi
 (/lib/modules/2.6.16.18/extra/mISDN_capi.ko): Unknown symbol in
 module, or unknown parameter (see dmesg)
 -
  Loading module(s) for your misdn-cards:
 -
 modprobe --ignore-install hfcmulti type=0x8
 protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2
 layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf poll=128 debug=0


 and dmesg:
 Modular ISDN Stack core $Revision: 1.34 $
 mISDNd: kernel daemon started
 ISDN L1 driver version 1.16
 ISDN L2 driver version 1.27
 mISDN: DSS1 Rev. 1.38
 mISDN_dsp: Audio DSP  Rev. 1.17 (debug=0x0) EchoCancellor MG2
 mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies.
 mISDN_capi: Unknown symbol capi_cmd2str
 mISDN_capi: Unknown symbol capi_cmsg_header
 mISDN_capi: Unknown symbol detach_capi_ctr
 mISDN_capi: Unknown symbol capi_cmsg2message
 mISDN_capi: Unknown symbol capi_ctr_reseted
 mISDN_capi: Unknown symbol capi_ctr_ready
 mISDN_capi: Unknown symbol capi_message2cmsg
 mISDN_capi: Unknown symbol capi_ctr_handle_message
 mISDN_capi: Unknown symbol attach_capi_ctr
 mISDNd: test event done
 mISDN: HFC-multi driver Rev. 1.41
 0 devices registered

 finally, if i try to start asterisk:

  [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri))
 mISDN_close: fid(18) isize(131072) inbuf(0xb7580008) irp(0xb7580008)
 iend(0xb7580008)
 Jun  3 08:42:31 ERROR[8296]: chan_misdn.c:3775 load_module: Unable to
 initialize mISDN
 Jun  3 08:42:31 WARNING[8296]: loader.c:414 __load_resource:
 chan_misdn.so: load_module failed, returning -1
 Jun  3 08:42:31 WARNING[8296]: loader.c:554 load_modules: Loading
 module chan_misdn.so failed!
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Re: [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered

2006-06-03 Thread nik600


thanks to your reply
using slackware the precompiled kernel is of the 2.4 series.

I've also tried to remove all modules of my 2.6 kernel, download it ,
configure it and boot it.
Then, using a new 2.6.16.18 kernel (and working) i've run the make
install of the beronet utility, but i'm still getting the same error.
:-(

Beronet has a support forum or support mailing list?


From their website it seems that the support is a payed service...


hi and thanks

On 6/3/06, Christophorus Laube [EMAIL PROTECTED] wrote:

Do you still have the precompiled kernel installed? Try to boot to it. Install
kernel sources and try to rerun the install-misdn-mqueue script. This is
doing the needed stuff for you. Maybe the recompilation of your kernel caused
the problem. I have installed three systems with misdn the last two months
(two of them with the BN8S0) and I did not have problems in that with the
vendor delivered precompiled kernel. To not need to recompile the kernel is
one of the reasons they build the misdn in a kernel and a user space part.
You can also contact beronet in order to get some support. They support their
cards until they are correctly recognized in asterisk.

Cheers
 thanks to your reply

 i've also tried to use install-misdn-mqueue but:

 [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue# /etc/init.d/misdn-init
 scan [OK] found the following devices:
 card=1,0x8
 [ii] run /etc/init.d/misdn-init config to store this information to
 /etc/misdn-init.conf

 as you can see, the scan finish succesfully, but if try to start misd i
 get: [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue#
 /etc/init.d/misdn-init start
 FATAL: Module capi not found.
 FATAL: Error inserting mISDN_capi
 (/lib/modules/2.6.16.18/extra/mISDN_capi.ko): Unknown symbol in
 module, or unknown parameter (see dmesg)
 -
  Loading module(s) for your misdn-cards:
 -
 modprobe --ignore-install hfcmulti type=0x8
 protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2
 layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf poll=128 debug=0


 and dmesg:
 Modular ISDN Stack core $Revision: 1.34 $
 mISDNd: kernel daemon started
 ISDN L1 driver version 1.16
 ISDN L2 driver version 1.27
 mISDN: DSS1 Rev. 1.38
 mISDN_dsp: Audio DSP  Rev. 1.17 (debug=0x0) EchoCancellor MG2
 mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies.
 mISDN_capi: Unknown symbol capi_cmd2str
 mISDN_capi: Unknown symbol capi_cmsg_header
 mISDN_capi: Unknown symbol detach_capi_ctr
 mISDN_capi: Unknown symbol capi_cmsg2message
 mISDN_capi: Unknown symbol capi_ctr_reseted
 mISDN_capi: Unknown symbol capi_ctr_ready
 mISDN_capi: Unknown symbol capi_message2cmsg
 mISDN_capi: Unknown symbol capi_ctr_handle_message
 mISDN_capi: Unknown symbol attach_capi_ctr
 mISDNd: test event done
 mISDN: HFC-multi driver Rev. 1.41
 0 devices registered

 finally, if i try to start asterisk:

  [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri))
 mISDN_close: fid(18) isize(131072) inbuf(0xb7580008) irp(0xb7580008)
 iend(0xb7580008)
 Jun  3 08:42:31 ERROR[8296]: chan_misdn.c:3775 load_module: Unable to
 initialize mISDN
 Jun  3 08:42:31 WARNING[8296]: loader.c:414 __load_resource:
 chan_misdn.so: load_module failed, returning -1
 Jun  3 08:42:31 WARNING[8296]: loader.c:554 load_modules: Loading
 module chan_misdn.so failed!
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RE: [Asterisk-Users] Polycom-Asterisk hints/presence

2006-06-03 Thread Damon Estep

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Mike Fedyk
 Sent: Friday, June 02, 2006 10:16 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Polycom-Asterisk hints/presence

 How do you setup asterisk so that the assistant sees the lights but
 doesn't hear the rings?

It was not intentional, but with 1.2.8 asterisk and 1.6.5 Polycom sip;

1. add a hint priority to the dialplan.
2. set the subscribe context for the subscriber
3. create a directory entry on the phone and enable buddy watch
4. assign the directory entry to a speed dial key. This can only be done
if you have not set all of the Polycom keys to register on the primary
account used for the phone. If you have a 501, and you set the first 2
for registration, the third will display speed dial 1

The buddy watch feature on the Polycom phone is enabled by adding the
appropriate xml tag to the config file for the phone - if you need more
info let me know.

The result I expected was that the watched extensions would ring, but
a sip debug in asterisk reveals that asterisk does not send a different
notify for ring and it does not send a notify when the state changes
from ringing to inuse. There is only one notify sent when the watched
extensions starts ringing, and it is a status of inuse

I have not been able to determine if the Polycom would ring if a status
notification of ringing was sent from asterisk.

Damon

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Re: [Asterisk-Users] All non US 48 area codes?

2006-06-03 Thread William Piper

Here is a link that can give you the US list:
http://www.bennetyee.org/ucsd-pages/area.html
On 6/2/06, voiplist [EMAIL PROTECTED] wrote:
Is there a list somewhere or a way to find the following:1- All non US 48 area codes which can be dialed as 1+10
2- All strange area codes which are used for premium services such as900-XXX-3- Anything else that should be restricted if one was to restrict allcalls to US 48 onlyI have found many list but it's tough looking at the entire list of
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[Asterisk-Users] X-Asterisk-HangupCause: Normal Clearing

2006-06-03 Thread Stephane Ricard









Hi,



I am initiating a SIP call
from Asterisk. After about 10 minutes, I loose audio in both directions but
the call seem to stay up. Can someone please help me understand what is
happening here. Been struggling on this for a while now. This one is
preventing me from fully enjoying my Asterisk installation L



Here are the 2 last debug
items from the console.



 -- SIP read from 62.123.211.31:5060:

 INFO sip:[EMAIL PROTECTED] SIP/2.0

 t: STEPHANE RICARD
sip:[EMAIL PROTECTED];tag=3Das1ea35b0b

 f:
sip:[EMAIL PROTECTED];tag=3D47270277584177094

 i: [EMAIL PROTECTED]

 CSeq: 76518 INFO

 v: SIP/2.0/UDP =


62.123.211.31:5060;branch=3Dz9hG4bK566fcdf06c789739901a9d3fdb0cd2f5

 Max-Forwards: 18

 x-nt-corr-id:
[EMAIL PROTECTED]

 k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec

 l: 0



 Receiving INFO!

 Transmitting (no NAT) to 62.123.211.31:5060:

 SIP/2.0 403 Unauthorized

 Via: SIP/2.0/UDP =


62.123.211.31:5060;branch=3Dz9hG4bK566fcdf06c789739901a9d3fdb0cd2f5;recei=

 ved=3D62.123.211.31

 From:
sip:[EMAIL PROTECTED];tag=3D47270277584177094

 To: STEPHANE RICARD
sip:[EMAIL PROTECTED];tag=3Das1ea35b0b

 Call-ID: [EMAIL PROTECTED]

 CSeq: 76518 INFO

 User-Agent: Asterisk PBX

 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY

 Contact: sip:[EMAIL PROTECTED]

 Content-Length: 0

 X-Asterisk-HangupCause: Normal Clearing



Thanks in advance.

Stephane








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Re: [Asterisk-Users] Problem with IAX2 dialin with portunity

Hi,

Matthias Fechner wrote:
 [portunity-in]
 type=user
 context=incoming-portunity
 permit=82.139.223.1/255.255.255.255

now I have the next problem.
I can connect an iax phone and a sip phone to my asterisk.
The problem is with incoming phone calls.
If I use xlite everything is working perfectly but diax and idefisk are
not working. So I think it is a problem with the IAX2 configuration. I
got the call but i cannot hear anything and the calling person cannot
her me.
If I transfer the call to hold the calling person can hear MoH.

Here is the debug log from asterisk:
___BEGIN___
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
   Timestamp: 5ms  SCall: 00058  DCall: 0 [82.139.223.1:4569]
   VERSION : 2
   CALLED NUMBER   : s
   CALLING NUMBER  : 03062006
   CALLING NAME: diax0.9.15a
   LANGUAGE: en
   FORMAT  : 2
   CAPABILITY  : 64798
   ADSICPE : 0
   DATE TIME   : 2006-06-03  17:00:04

-- Accepting UNAUTHENTICATED call from 82.139.223.1:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw|alaw|gsm),
priority = mine
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
   Timestamp: 1ms  SCall: 1  DCall: 00058 [82.139.223.1:4569]
   FORMAT  : 4

-- Executing Dial(IAX2/portunity-out-1, IAX2/idefixSIP/idefix)
in new stack
-- Called idefix
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
   Timestamp: 4ms  SCall: 6  DCall: 0 [192.168.0.151:4569]
   VERSION : 2
   CALLED NUMBER   : s
   CODEC_PREFS : (ulaw|alaw|gsm)
   CALLING NUMBER  : 03062006
   CALLING PRESNTN : 0
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   CALLING NAME: diax0.9.15a
   LANGUAGE: de
   USERNAME: idefix
   FORMAT  : 14
   CAPABILITY  : 63502
   ADSICPE : 0
   DATE TIME   : 2006-06-03  17:00:06

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
   Timestamp: 4ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
   Timestamp: 1ms  SCall: 00058  DCall: 1 [82.139.223.1:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
   Timestamp: 00016ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
   FORMAT  : 14

-- Call accepted by 192.168.0.151 (format unknown)
-- Format for call is (gsm|ulaw|alaw)
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
   Timestamp: 00016ms  SCall: 6  DCall: 00310 [192.168.0.151:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass:
RINGING
   Timestamp: 3ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 3ms  SCall: 6  DCall: 00310 [192.168.0.151:4569]
-- IAX2/idefix-6 is ringing
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass:
RINGING
   Timestamp: 4ms  SCall: 1  DCall: 00058 [82.139.223.1:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 4ms  SCall: 00058  DCall: 1 [82.139.223.1:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass: PING
   Timestamp: 02000ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 003 Type: IAX Subclass: PONG
   Timestamp: 02000ms  SCall: 6  DCall: 00310 [192.168.0.151:4569]
   RR_JITTER   : 0
   RR_LOSS : 0
   RR_PKTS : 1
   RR_DELAY: 40
   RR_DROPPED  : 0
   RR_OUTOFORDER   : 0

Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 02000ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 002 Type: CONTROL Subclass:
ANSWER
   Timestamp: 03485ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass: ACK
   Timestamp: 03485ms  SCall: 6  DCall: 00310 [192.168.0.151:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 002 Type: CONTROL Subclass:
ANSWER
   Timestamp: 03488ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 005 Type: IAX Subclass: ACK
   Timestamp: 03488ms  SCall: 6  DCall: 00310 [192.168.0.151:4569]
-- IAX2/idefix-6 answered IAX2/portunity-out-1
-- Attempting native bridge of IAX2/portunity-out-1 and IAX2/idefix-6
-- Operating with different codecs 4[(ulaw)] 14[(gsm|ulaw|alaw)] ,
can't native bridge...
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 001 Type: CONTROL Subclass:
(255?)
   Timestamp: 03489ms  SCall: 1  DCall: 00058 [82.139.223.1:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 001 Type: CONTROL Subclass:
ANSWER
   Timestamp: 03492ms  SCall: 1  DCall: 00058

Re: [Asterisk-Users] DID from Latvia?

2006-06-03 Thread David K Parker

Well, I'm glad my mis-step benefited somebody else. Cost me $22 and not even sure the inbound service works yet. Their online setup docs don't match the config menus on their web site.
On 6/2/06, Ira [EMAIL PROTECTED] wrote:
At 03:03 PM 6/2/2006, you wrote:In fact, the very first words on the website are Voxbone providesinternational VOIP virtual numbers and worldwide originationservices via VoIP.
Notice the words origination. I didn't know it was necessarry topublish on the website what you do **NOT** do as well.But for those of us new to this game, some of those really obviouswords just don't mean what they do to you. Thankfully I figured it
out before mis-stepping, but for a very small * user like me the ideathat one provider provides dial tone and another ring was a bit oddat first and the words origination and termination don't really helpuntil you understand what they mean.
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Re: [Asterisk-Users] X100P fails to initialize

2006-06-03 Thread Josué Conti

Hello.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
I think that this error, is saying that its X100P is not connected in slot PCI correctly. He makes a test, he changes the X100P of slot and he sends for the list the commands lspci, dmesg, cat /proc/interrupts. I wait to have helped.

Best RegardsJosué
2006/6/3, Woodoo People .pGa! [EMAIL PROTECTED]:
So I took a chance with an X100P knock-off on eBay. I'm running Asterisk + FreePBX on a PowerMac G3 (beige desktop) using Slackintosh
10.2 and kernel 2.6.16.16. Everything has been fine up until now. I compile the 1.2.5 Zaptel drivers without a problem, get the udev configuration in, modprobe zaptel, and finally modprobe wcfxo. At this
point, I get the message: ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wcfxo dmesg gives me: Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.5 Echo Canceller: MG2 Failed to initailize DAA, giving up... wcfxo: probe of :00:0e.0 failed with error -5this is a problem if the card shares interrupt with something on MOtherBOard.
configure the slot to other irq, or move the card to other irq.lspci -v will be your friend. Check if someone uses same IRQ as motorola card.--WoodOO-[P]an[G]alaktikan[A]gent-People ][
http://shadow.pganet.com[EMAIL PROTECTED][EMAIL PROTECTED]___--Bandwidth and Colocation provided by
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[Asterisk-Users] What's asterisk on FreeBSD like now a days?

2006-06-03 Thread Jason Lixfeld

I need a simple system with MoH, Meetme and timing using a TDM400P  
with an FXO.


Any user reports?
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Re: [Asterisk-Users] Asterisk - Qsig

2006-06-03 Thread Josué Conti

Michael,thank´s for thisattention.
I go to test with equipment Siemens HiPath and features. I sending for you and the list an email with the results of the tests, ok? How was
zapata.conf of its asterisk with qsig? You it only changed switchtype=euroisdn, for switchtype=qsig?
Thank youfor its attention.
Greetings Josué
2006/6/3, Michael Konietzny [EMAIL PROTECTED]:
Hello Josué,we're running Asterisk in combination of the T-Com Octopus E800 withQSig Protocoll. The protocoll itself is supported but some features are
missing, or i didn't found out yethow to use them. I'm also interested in how to use qsig fordeterminating if other phones are available for calling and so on.Greetings,MichaelJosué Conti schrieb:
Hello all, as good? It would like to make a question, asterisk supports the protocol qsig, for interconnections in ISDN with equipment Siemens HiPath 4000 or same Ericsson MD110, so that it could identify to the name and the
number of hosts and also to use some features of asterisk in the Siemens/Ericsson equipment. Greetings Josué
___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users--Mit freundlichen GrüßenMichael Konietzny
e_mail:[EMAIL PROTECTED]handy:0176 / 24 79 8656phone:03529 / 527597address:Feldstrasse 701809 Heidenau___
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Re: [Asterisk-Users] Asterisk - Qsig

2006-06-03 Thread Michael Konietzny

Hello Josué,

yes i currently only switched switchtype in zapata.conf to the value
qsig. The only real PRI feature i've found out is the PRI_CAUSE
variable set on Hangup().

Greetings,
  Michael

Josué Conti schrieb:
 Michael, thank´s for this attention.
 I go to test with equipment Siemens HiPath and features. I sending for
 you and the list an email with the results of the tests, ok? How was
 zapata.conf of its asterisk with qsig? You it only changed
 switchtype=euroisdn, for switchtype=qsig?
 Thank you for its attention.
 Greetings
 Josué

 2006/6/3, Michael Konietzny [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]:

 Hello Josué,

 we're running Asterisk in combination of the T-Com Octopus E800 with
 QSig Protocoll. The protocoll itself is supported but some
 features are
 missing, or i didn't found out yet
 how to use them. I'm also interested in how to use qsig for
 determinating if other phones are available for calling and so on.

 Greetings,
 Michael

 Josué Conti schrieb:
  Hello all, as good?
  It would like to make a question, asterisk supports the protocol
 qsig,
  for interconnections in ISDN with equipment Siemens HiPath 4000 or
  same Ericsson MD110, so that it could identify to the name and the
  number of hosts and also to use some features of asterisk in the
  Siemens/Ericsson equipment.
  Greetings
  Josué
 
 

 
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 --
 Mit freundlichen Grüßen

 Michael Konietzny


 e_mail:
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]

 handy:
 0176 / 24 79 8656

 phone:
 03529 / 527597

 address:
 Feldstrasse 7
 01809 Heidenau

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-- 
Mit freundlichen Grüßen

Michael Konietzny


e_mail:  
[EMAIL PROTECTED]

handy:   
0176 / 24 79 8656

phone: 
03529 / 527597

address:  
Feldstrasse 7
01809 Heidenau

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[Asterisk-Users] is '9' needed for outside numbers

2006-06-03 Thread M.Hockings

I have a small setup with half a dozen phones and a couple of soft 
phones that share a voip line and a pstn line.  Right now I have it 
configured to require a 9 prefix to dial to a number outside the 
building.  Where we are it is 10 digit dialing for even local numbers, 
the local dialable area codes start with 2,4,6 or 9 and our internal 
extensions all start with 3.  So I was thinking that I could make it 
more natural to just eliminate the requirement to dial the 9 prefix for 
an outside number?  Can anyone see problems with doing this?


Thanks for any opinions,

Mike

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[Asterisk-Users] Asterisk + PRI Card - Nortel BCM

2006-06-03 Thread Mr. Jones


Has anyone fed a Nortel BCM from Asterisk?

I'm interested in switching our company over, but don't want to
replace all the handsets in one fell swoop.

I imagine some of the PRI cards can emulate a switch?

I'd still like to pass CallerID into the Nortel, etc but all the
external traffic would be VOIP, not TDM.
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Re: [Asterisk-Users] Asterisk + PRI Card - Nortel BCM


- Mr. Jones [EMAIL PROTECTED] wrote:

 I imagine some of the PRI cards can emulate a switch?

Asterisk (and Zaptel) handles all call signaling not the cards. What that means 
to you is that any Asterisk-supporting T1/E1 card can operate in PRI mode and 
act as the network end as well as acting as CPE. Many people use Asterisk in 
exactly the fashion you are describing.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] is '9' needed for outside numbers

2006-06-03 Thread Ira


At 10:06 AM 6/3/2006, you wrote:
So I was thinking that I could make it more natural to just 
eliminate the requirement to dial the 9 prefix for an outside 
number?  Can anyone see problems with doing this?


Works perfect, jut remember to leave the code that recognizes 9 for 
the people who have a hard time learning the new way. You can ever 
get fancier and add the 1 if they leave it off. or add the local 3 
digit prefix if they just dial the 7 digit number. I do those things 
and it works perfect.


Ira 


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RE: [Asterisk-Users] MWI lost after migration

2006-06-03 Thread Mimmus

 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Thomas Kenyon
 
 Mimmus wrote:
  Hi,
  I just migrated my Asterisk installation from 1.2.1 to 
 another server 
  with 1.2.8. Among a lot of things, I copied the whole content of 
  /var/spool/asterisk/voicemail/default directory.
  All is OK but now I'm not able to see MWI indication for 
 new messages 
  on all my Grandstream GXP2000 phones (before migration, it worked).
  Peraphs do I missed something?
 
  Thanks

 I thought that so long as there was a mailbox= statement 
 in the channel definition, this always worked.

My dialplan was derived from AMP and I had [EMAIL PROTECTED] in sip.conf
instead of [EMAIL PROTECTED]

Thanks
DV

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Re: [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered

That's right, mISDN only supports kernels up from version 2.6.9. So I see you 
did have to compile a kernel yourself. 
Beronet has a telephone number where they offer support. This is german one. 
They also have a support mail address, just have a look at their site 
http://www.beronet.com and choose english language if you do not speak or 
better read german.
I often phoned to Beronet and the support was great. If you can give them ssh 
access to your machine for a while and they will show you what to do. 
Another point of failure may be your jumper settings. The mISDN driver can 
only recognize wether the ports are jumpered TE or NT, not wether ptp or 
ptmp. This what you have to tell it. 
Again, just download the card_installation_guide.pdf from 
http://www.beronet.com/downloads. This helped me a lot.

 thanks to your reply
 using slackware the precompiled kernel is of the 2.4 series.

 I've also tried to remove all modules of my 2.6 kernel, download it ,
 configure it and boot it.
 Then, using a new 2.6.16.18 kernel (and working) i've run the make
 install of the beronet utility, but i'm still getting the same error.

 :-(

 Beronet has a support forum or support mailing list?

 From their website it seems that the support is a payed service...

 hi and thanks

 On 6/3/06, Christophorus Laube [EMAIL PROTECTED] wrote:
  Do you still have the precompiled kernel installed? Try to boot to it.
  Install kernel sources and try to rerun the install-misdn-mqueue script.
  This is doing the needed stuff for you. Maybe the recompilation of your
  kernel caused the problem. I have installed three systems with misdn the
  last two months (two of them with the BN8S0) and I did not have problems
  in that with the vendor delivered precompiled kernel. To not need to
  recompile the kernel is one of the reasons they build the misdn in a
  kernel and a user space part. You can also contact beronet in order to
  get some support. They support their cards until they are correctly
  recognized in asterisk.
 
  Cheers
 
   thanks to your reply
  
   i've also tried to use install-misdn-mqueue but:
  
   [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue#
   /etc/init.d/misdn-init scan [OK] found the following devices:
   card=1,0x8
   [ii] run /etc/init.d/misdn-init config to store this information to
   /etc/misdn-init.conf
  
   as you can see, the scan finish succesfully, but if try to start misd i
   get: [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue#
   /etc/init.d/misdn-init start
   FATAL: Module capi not found.
   FATAL: Error inserting mISDN_capi
   (/lib/modules/2.6.16.18/extra/mISDN_capi.ko): Unknown symbol in
   module, or unknown parameter (see dmesg)
   -
Loading module(s) for your misdn-cards:
   -
   modprobe --ignore-install hfcmulti type=0x8
   protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2
   layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf poll=128 debug=0
  
  
   and dmesg:
   Modular ISDN Stack core $Revision: 1.34 $
   mISDNd: kernel daemon started
   ISDN L1 driver version 1.16
   ISDN L2 driver version 1.27
   mISDN: DSS1 Rev. 1.38
   mISDN_dsp: Audio DSP  Rev. 1.17 (debug=0x0) EchoCancellor MG2
   mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies.
   mISDN_capi: Unknown symbol capi_cmd2str
   mISDN_capi: Unknown symbol capi_cmsg_header
   mISDN_capi: Unknown symbol detach_capi_ctr
   mISDN_capi: Unknown symbol capi_cmsg2message
   mISDN_capi: Unknown symbol capi_ctr_reseted
   mISDN_capi: Unknown symbol capi_ctr_ready
   mISDN_capi: Unknown symbol capi_message2cmsg
   mISDN_capi: Unknown symbol capi_ctr_handle_message
   mISDN_capi: Unknown symbol attach_capi_ctr
   mISDNd: test event done
   mISDN: HFC-multi driver Rev. 1.41
   0 devices registered
  
   finally, if i try to start asterisk:
  
[chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri))
   mISDN_close: fid(18) isize(131072) inbuf(0xb7580008) irp(0xb7580008)
   iend(0xb7580008)
   Jun  3 08:42:31 ERROR[8296]: chan_misdn.c:3775 load_module: Unable to
   initialize mISDN
   Jun  3 08:42:31 WARNING[8296]: loader.c:414 __load_resource:
   chan_misdn.so: load_module failed, returning -1
   Jun  3 08:42:31 WARNING[8296]: loader.c:554 load_modules: Loading
   module chan_misdn.so failed!
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[Asterisk-Users] Busy Signals after hangup

2006-06-03 Thread Rick Smith



I've not seen an answer to this in any forum.

I make a call through Asterisk, with a VOIP phone, doesn't matter which.

The call gets made, I leave a voicemail, or complete the call in some 
manner, and the other side hangs up.  I hear a busy signal on the phone 
on my end.


If I have an extension that looks like this, after the hangup() is 
executed, my phone gives busy signals until I hangup and pick up to get 
another dial tone.


exten = 199,1,Answer()
exten = 199,2,Dial(SIP/100,20)
exten = 199,3,Hangup


why?  And how to fix ?   This is annoying...

R

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Re: [Asterisk-Users] Asterisk + PRI Card - Nortel BCM

2006-06-03 Thread Mr. Jones


Excellent.  -

So I can basically make a crossover cable to my Nortel, and pass calls
to the old phones from the PTSN (via my VOIP originator ) in to it?

I guess I'm off to look for sample configs.

Thx

Brian

On 6/3/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:


- Mr. Jones [EMAIL PROTECTED] wrote:

 I imagine some of the PRI cards can emulate a switch?

Asterisk (and Zaptel) handles all call signaling not the cards. What that means 
to you is that any Asterisk-supporting T1/E1 card can operate in PRI mode and 
act as the network end as well as acting as CPE. Many people use Asterisk in 
exactly the fashion you are describing.

--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Problem with IAX2 dialin with portunity

2006-06-03 Thread Tim Panton



On 3 Jun 2006, at 16:11, Matthias Fechner wrote:


Hi,

Matthias Fechner wrote:

[portunity-in]
type=user
context=incoming-portunity
permit=82.139.223.1/255.255.255.255


now I have the next problem.
I can connect an iax phone and a sip phone to my asterisk.
The problem is with incoming phone calls.
If I use xlite everything is working perfectly but diax and idefisk  
are

not working. So I think it is a problem with the IAX2 configuration. I
got the call but i cannot hear anything and the calling person cannot
her me.
If I transfer the call to hold the calling person can hear MoH.

Here is the debug log from asterisk:
___BEGIN___
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX  
Subclass: NEW

   Timestamp: 5ms  SCall: 00058  DCall: 0 [82.139.223.1:4569]
   VERSION : 2
   CALLED NUMBER   : s
   CALLING NUMBER  : 03062006
   CALLING NAME: diax0.9.15a
   LANGUAGE: en
   FORMAT  : 2
   CAPABILITY  : 64798
   ADSICPE : 0
   DATE TIME   : 2006-06-03  17:00:04

-- Accepting UNAUTHENTICATED call from 82.139.223.1:

requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw|alaw|gsm),
priority = mine

Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
   Timestamp: 1ms  SCall: 1  DCall: 00058 [82.139.223.1:4569]
   FORMAT  : 4

-- Executing Dial(IAX2/portunity-out-1, IAX2/idefixSIP/ 
idefix)

in new stack
-- Called idefix
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX  
Subclass: NEW

   Timestamp: 4ms  SCall: 6  DCall: 0 [192.168.0.151:4569]
   VERSION : 2
   CALLED NUMBER   : s
   CODEC_PREFS : (ulaw|alaw|gsm)
   CALLING NUMBER  : 03062006
   CALLING PRESNTN : 0
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   CALLING NAME: diax0.9.15a
   LANGUAGE: de
   USERNAME: idefix
   FORMAT  : 14
   CAPABILITY  : 63502
   ADSICPE : 0
   DATE TIME   : 2006-06-03  17:00:06

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX  
Subclass: ACK

   Timestamp: 4ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX  
Subclass: ACK

   Timestamp: 1ms  SCall: 00058  DCall: 1 [82.139.223.1:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
   Timestamp: 00016ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
   FORMAT  : 14

-- Call accepted by 192.168.0.151 (format unknown)
-- Format for call is (gsm|ulaw|alaw)
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX  
Subclass: ACK

   Timestamp: 00016ms  SCall: 6  DCall: 00310 [192.168.0.151:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass:
RINGING
   Timestamp: 3ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX  
Subclass: ACK

   Timestamp: 3ms  SCall: 6  DCall: 00310 [192.168.0.151:4569]
-- IAX2/idefix-6 is ringing
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass:
RINGING
   Timestamp: 4ms  SCall: 1  DCall: 00058 [82.139.223.1:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX  
Subclass: ACK

   Timestamp: 4ms  SCall: 00058  DCall: 1 [82.139.223.1:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 001 Type: IAX  
Subclass: PING

   Timestamp: 02000ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 003 Type: IAX  
Subclass: PONG

   Timestamp: 02000ms  SCall: 6  DCall: 00310 [192.168.0.151:4569]
   RR_JITTER   : 0
   RR_LOSS : 0
   RR_PKTS : 1
   RR_DELAY: 40
   RR_DROPPED  : 0
   RR_OUTOFORDER   : 0

Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 002 Type: IAX  
Subclass: ACK

   Timestamp: 02000ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 002 Type: CONTROL Subclass:
ANSWER
   Timestamp: 03485ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 004 Type: IAX  
Subclass: ACK

   Timestamp: 03485ms  SCall: 6  DCall: 00310 [192.168.0.151:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 002 Type: CONTROL Subclass:
ANSWER
   Timestamp: 03488ms  SCall: 00310  DCall: 6 [192.168.0.151:4569]
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 005 Type: IAX  
Subclass: ACK

   Timestamp: 03488ms  SCall: 6  DCall: 00310 [192.168.0.151:4569]
-- IAX2/idefix-6 answered IAX2/portunity-out-1
-- Attempting native bridge of IAX2/portunity-out-1 and IAX2/ 
idefix-6

-- Operating with different codecs 4[(ulaw)] 14[(gsm|ulaw|alaw)] ,
can't native bridge...
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 001 Type: CONTROL Subclass:
(255?)
   Timestamp: 03489ms  SCall: 1  DCall: 00058 [82.139.223.1:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 001 Type: CONTROL Subclass:
ANSWER

Re: [Asterisk-Users] Asterisk + PRI Card - Nortel BCM


- Mr. Jones [EMAIL PROTECTED] wrote:

 So I can basically make a crossover cable to my Nortel, and pass
 calls
 to the old phones from the PTSN (via my VOIP originator ) in to it?

Exactly. Many examples of this on the voip-info wiki.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Busy Signals after hangup


- Rick Smith [EMAIL PROTECTED] wrote:

 exten = 199,1,Answer()
 exten = 199,2,Dial(SIP/100,20)
 exten = 199,3,Hangup
 
 
 why?  And how to fix ?   This is annoying...

This is handled entirely by your phone. Asterisk has already closed the channel 
to the phone, so it cannot do anything to control this behavior.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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[Asterisk-Users] TDM PCI Master Abort

2006-06-03 Thread Stephen Bosch

Hello:

I am configuring a TDM-400 card (the dev kit) with Trixbox
([EMAIL PROTECTED]). When I try to apply settings in FreePBX, the machine
locks up, except at the console, where I see

TDM PCI Master Abort

scrolling repeatedly down the screen.

I think this problem has been seen before, but nobody seems to know what
causes it. Any suggestions?

The machine has an Asus A7N266 board in it.

-Stephen-
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Re: [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered

You may also have a look at 
http://www.voip-info.org/wiki/view/Asterisk+mISDN+channels
 thanks to your reply
 using slackware the precompiled kernel is of the 2.4 series.

 I've also tried to remove all modules of my 2.6 kernel, download it ,
 configure it and boot it.
 Then, using a new 2.6.16.18 kernel (and working) i've run the make
 install of the beronet utility, but i'm still getting the same error.

 :-(

 Beronet has a support forum or support mailing list?

 From their website it seems that the support is a payed service...

 hi and thanks

 On 6/3/06, Christophorus Laube [EMAIL PROTECTED] wrote:
  Do you still have the precompiled kernel installed? Try to boot to it.
  Install kernel sources and try to rerun the install-misdn-mqueue script.
  This is doing the needed stuff for you. Maybe the recompilation of your
  kernel caused the problem. I have installed three systems with misdn the
  last two months (two of them with the BN8S0) and I did not have problems
  in that with the vendor delivered precompiled kernel. To not need to
  recompile the kernel is one of the reasons they build the misdn in a
  kernel and a user space part. You can also contact beronet in order to
  get some support. They support their cards until they are correctly
  recognized in asterisk.
 
  Cheers
 
   thanks to your reply
  
   i've also tried to use install-misdn-mqueue but:
  
   [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue#
   /etc/init.d/misdn-init scan [OK] found the following devices:
   card=1,0x8
   [ii] run /etc/init.d/misdn-init config to store this information to
   /etc/misdn-init.conf
  
   as you can see, the scan finish succesfully, but if try to start misd i
   get: [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue#
   /etc/init.d/misdn-init start
   FATAL: Module capi not found.
   FATAL: Error inserting mISDN_capi
   (/lib/modules/2.6.16.18/extra/mISDN_capi.ko): Unknown symbol in
   module, or unknown parameter (see dmesg)
   -
Loading module(s) for your misdn-cards:
   -
   modprobe --ignore-install hfcmulti type=0x8
   protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2
   layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf poll=128 debug=0
  
  
   and dmesg:
   Modular ISDN Stack core $Revision: 1.34 $
   mISDNd: kernel daemon started
   ISDN L1 driver version 1.16
   ISDN L2 driver version 1.27
   mISDN: DSS1 Rev. 1.38
   mISDN_dsp: Audio DSP  Rev. 1.17 (debug=0x0) EchoCancellor MG2
   mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies.
   mISDN_capi: Unknown symbol capi_cmd2str
   mISDN_capi: Unknown symbol capi_cmsg_header
   mISDN_capi: Unknown symbol detach_capi_ctr
   mISDN_capi: Unknown symbol capi_cmsg2message
   mISDN_capi: Unknown symbol capi_ctr_reseted
   mISDN_capi: Unknown symbol capi_ctr_ready
   mISDN_capi: Unknown symbol capi_message2cmsg
   mISDN_capi: Unknown symbol capi_ctr_handle_message
   mISDN_capi: Unknown symbol attach_capi_ctr
   mISDNd: test event done
   mISDN: HFC-multi driver Rev. 1.41
   0 devices registered
  
   finally, if i try to start asterisk:
  
[chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri))
   mISDN_close: fid(18) isize(131072) inbuf(0xb7580008) irp(0xb7580008)
   iend(0xb7580008)
   Jun  3 08:42:31 ERROR[8296]: chan_misdn.c:3775 load_module: Unable to
   initialize mISDN
   Jun  3 08:42:31 WARNING[8296]: loader.c:414 __load_resource:
   chan_misdn.so: load_module failed, returning -1
   Jun  3 08:42:31 WARNING[8296]: loader.c:554 load_modules: Loading
   module chan_misdn.so failed!
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Re: [Asterisk-Users] TDM PCI Master Abort

2006-06-03 Thread Jeremy McNamara


Stephen Bosch wrote:

TDM PCI Master Abort



Does that motherboard support PCI v2.2?
Have you tried a different slot?
Is the MOLEX power connector plugged in?
Are the modules fully seated?




Jeremy McNamara
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[Asterisk-Users] Integrating Asterisk

Have either of you any experience integrating Asterisk-related devices into existing phone equipment (trunks/pots lines, etc. (I'm somewhat new to legacy  voip based telephony)) to ensure
specific or random outbound calls route through Asterisk vs bell company (ATT)? Thanks in advance,Dakota
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[Asterisk-Users] Bullet-proof System

I want to provide VoIP hosting service to 2-10+ non-profit organizations
we grant services too, and possibly some small businesses. The server environment we're looking at starting out on (systems previously used for Web development, so I have these at a very low cost over the next 18 months), is at a hosting facility providing standard dedicated server features (generator backup, 24/7 call support if server is down, etc.) is: 
* Two load balanced servers with following specs using hardware load balancing with Coyote Point switch.Dual Xeon 3Ghz processors2GB RAMRAID5 configuration (80GB available)3Mbps bi-directional bandwidth with $25/month per 1Mb increase
Does the above sound like a fairly sound system to provide VoIP hosting? Anything seem like overkill, or anything you think should be added? Thanks in advance,Dakota
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[Asterisk-Users] Recommended Web Interface

I'm currently reviewing the latest release of FreePBX (formerly known as [EMAIL PROTECTED]). Do either of you know whether FreePBX is robust enough to handle
multiple clients, or have any recommendations on front-end Web interface to
manage client config  provide clients access to manage their level of
access (similar to how Vonage, Teliax, and others provide client access to their web management
console)? The latest FreePBX is module driven - pretty cool. I've plans to step through the Asterisk: Future of Telephony with old laptop in order to get a different view of Asterisk. Am used to Linux  CLI -- do either of you have any preferences? My guess is that some purists may look at FreePBX as a lesser product but ... I think it's simply a product base built right on top of Asterisk to help new Asterisk people hit the ground running (and provides some extras such as SugarCRM, Credit Card app, etc.). 
Thanks,Dakota
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[Asterisk-Users] Asterisk 1.2.8

Hi,

is a new port for Asterisk 1.2.8 for FreeBSD out?
Regarding to the changelog there some bugs fixed with iax and the
codecs. I hope with asterisk 1.2.8 my problem with IAX2 is solved.

Best regards,
Matthias
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Re: [Asterisk-Users] TDM PCI Master Abort

2006-06-03 Thread Stephen Bosch

Jeremy McNamara wrote:
 Stephen Bosch wrote:
 
 TDM PCI Master Abort
 
 
 
 Does that motherboard support PCI v2.2?

I don't know. I'll have to check. Is that a requirement?

 Have you tried a different slot?

No, not yet. I have tried forcing IRQ assignments, but perhaps I need to
try a physically different slot.

 Is the MOLEX power connector plugged in?

Yes.

 Are the modules fully seated?

Yes.

-Stephen-
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Re: [Asterisk-Users] TDM PCI Master Abort

2006-06-03 Thread Jeremy McNamara


Stephen Bosch wrote:

I don't know. I'll have to check. Is that a requirement?



Yes - Most absolutely.


http://www.digium.com/en/products/hardware/tdm400p.php




Jeremy McNamara
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Re: [Asterisk-Users] Asterisk 1.2.8

Hi,

* Matthias Fechner [EMAIL PROTECTED] [03-06-06 22:13]:
 is a new port for Asterisk 1.2.8 for FreeBSD out?
 Regarding to the changelog there some bugs fixed with iax and the
 codecs. I hope with asterisk 1.2.8 my problem with IAX2 is solved.

sry, mail should go to [EMAIL PROTECTED]


Best regards,
Matthias
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Re: [Asterisk-Users] Integrating Asterisk

2006-06-03 Thread Martin Joseph



On Jun 3, 2006, at 12:53 PM, Dakota Burns wrote:

Have either of you any experience integrating Asterisk-related devices 
into existing phone equipment (trunks/pots lines, etc. (I'm somewhat 
new to legacy  voip based telephony)) to ensure specific or random 
outbound calls route through Asterisk vs bell company (ATT)? 



Random?

Try rephrasing your question into a more meaningful form.

Marty

PS Yes everyone here (both of us) integrate asterisk with everything.

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Re: [Asterisk-Users] Problem with IAX2 dialin with portunity

Hello Tim,

* Tim Panton [EMAIL PROTECTED] [03-06-06 19:12]:
 You have a weird codec problem.
 Try changing the iax config to limit it  to ulaw and see if that helps:
 
 [portunity-in]
 type=user
 context=incoming-portunity
 permit=82.139.223.1/255.255.255.255
 disallow=all
 allow=ulaw

sry that doesn't help.

 You might also want to upgrade to asterisk 1.2.8 - which has
 some fixes in the IAX code - but I don't know if any are related to
 this - I haven't had a chance to install it yet.

ah great if FreeBSD port is up-to-date I will upgrade and give some
feedback here.

Best regards,
Matthias
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Fw: [Asterisk-Users] Compiling chan_bluetooth

2006-06-03 Thread Danko Miocevic

Just to close the thread. The problem was that I was using an old version of 
the code.

If anyone has the same problem, you can download the code from here:

http://www.thetechguide.com/howto/asterisk/bluetoothfiles.tar.gz

Good luck,
   Danko




- Original Message - 
From: Danko Miocevic [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Tuesday, May 30, 2006 8:48 PM
Subject: Re: [Asterisk-Users] Compiling chan_bluetooth


I´ve found a solution to my problem, I forgot to install the posix 
development libraries.. now the error has dissapeared to

make place to a new error! :D I still can´t compile.
The new error says:

cc: chan_gsm_bt.o: no se usó el fichero de entrada del enlazador porque no 
se hizo enlace
cc: -lbluetooth: no se usó el fichero de entrada del enlazador porque no 
se hizo enlace


It says that the file wasn´t used because the linker didn´t make link... 
don´t know what to do..

Any ideas?

Danko


- Original Message - 
From: Danko Miocevic [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Saturday, May 27, 2006 12:51 PM
Subject: [Asterisk-Users] Compiling chan_bluetooth


Hello, I´m trying to use my phone with asterisk to get GSM connectivity 
but I can´t compile the code.
I got the asterisk, chan_bluetooth, zaptel and libpri sources, the last 
two ones compiled perfectly.

I have added this to the /usr/src/asterisk/channels/Makefile:
   include /usr/src/chan_bluetooth/Makefile
and I´ve also added chan_bluetooth.so to the CHANNEL_LIBS var.
When I do make install in the asterisk directory I get lots of this 
error:


/usr/src/chan_bluetooth/chan_bluetooth.c:2469: error: dereferencing 
pointer to incomplete type


and some others like:

/usr/src/chan_bluetooth/chan_bluetooth.c: En la función 
`remove_sdp_records':
/usr/src/chan_bluetooth/chan_bluetooth.c:2937: error: `sdp_session_t' 
undeclared (first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2937: error: `sdp' undeclared 
(first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2938: error: `sdp_list_t' 
undeclared (first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2938: error: `attr' undeclared 
(first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2939: error: `sdp_record_t' 
undeclared (first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2939: error: `rec' undeclared 
(first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2948: aviso: implicit 
declaration of function `sdp_connect'
/usr/src/chan_bluetooth/chan_bluetooth.c:2948: error: `BDADDR_ANY' 
undeclared (first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2948: error: `BDADDR_LOCAL' 
undeclared (first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2948: error: `SDP_RETRY_IF_BUSY' 
undeclared (first use in this function)


I´m working on a stable debian, kernel 2.6 and my asterisk is 1.2. I 
really don´t know what is happening, if someone

has an idea I´d be glad to hear it. Thanks for reading,

Danko
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Re: [Asterisk-Users] Prices of g729 codec

On Fri, 2006-06-02 at 12:12 -0400, Andrew Kohlsmith wrote:
 The Intel g729 code is licensed for educational use ONLY.  Commercial use is 
 forbidden without paying the patent holder.  $10 a port won't break the bank 
 of any business with a shred of a hope of a chance of surviving, and you stay 
 legitimate.
 
but $10 only gets you one license, what if you are vonage sized and need
to support a million customers?  What if you accept that you can settle
for a 5:1 ratio, then its only 200,000 or $2M.  Just for codec licenses,
not to mention all the other costs of being a business.  What if you are
smaller than vonage, say 10k channels in use, then that smaller entity,
probably without the hundreds of millions of VC that vonage got you
would have to come up with $100k.  Still more than $10.  

If you are going to bring businesses into it, at least accept that a
business would most likely pay more than $10 for their licensing needs.

And the inten IPP g729 stuff isnt licensed at all, educational or
otherwise.  Read the information from intel on that.  While it is
generally accepted that educational uses can use patents without a
license that doesnt always guarantee that fact.


 Try buying a legit g729 license from the patent holder if you're a home user 
 or small business wanting to transcode g729.  They only want to license 
 hundreds of instances at a time, if not thousands.  Digium negotiated a 
 pretty damn good license fee so that they could offer the codec and sell it 
 in onesie-twosie quantities to little guys like us at an affordable price. 
 
no its not that they want quantity becuase they will sell just one
license, they only want to deal with people that implement the systems
not the end users of the system.  They claim the reasoning for this is
to make it easier for end users to know that they have licenses -
basically if you have it you are licensed.  Even if that isnt the case.
Check www.sipro.com for more info on g729 licensing.

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com we pay you to terminate calls with us!


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[Asterisk-Users] Size limitations of extensions.conf

2006-06-03 Thread voiplist


Can someone tell me the size (or any other) limitations for the extensions.conf?

We have managed to keep our file pretty small thanks to AGI but we are
about to setup a bunch of call restrictions based on area and country
code.

One line per area code in the US alone adds a LOT of text to this file.

Is it a bad thing to have 5 or 6000 lines of text in your
extensions.conf on a production system?

Will it affect the performance?
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Re: [Asterisk-Users] Integrating Asterisk

What I was attempting to visualize is the following case: 10 people in an organization pick-up their phones to make an outbound call. Before integrating Asterisk, all calls route through their current non-VoIP based phone provider. After integrating 1 trunk from a VoIP service provider into their system that provides 4 simultaneous calls (Teliax's Corporate plan), and dropping 4 legacy lines, if 10 people make calls simultaneously, some will be VoIP and some will be legacy based. Based on the above example, I'm questioning whether it would be best to configure a Sipura 3000 for every analog phone (I'm guessing the non-profits will want to keep their existing analog phones), or utilize another device (or devices) to connect the company's internet service into their existing Trunks or POTS. I think the former would be easier  something I know how to do, but the latter may be smarter  more cost effective. So the latter is what I'm questioning whether either of you have experience implementing. 
Thanks,DakotaOn 6/3/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Jun 3, 2006, at 12:53 PM, Dakota Burns wrote: Have either of you any experience integrating Asterisk-related devices into existing phone equipment (trunks/pots lines, etc. (I'm somewhat new to legacy  voip based telephony)) to ensure specific or random
 outbound calls route through Asterisk vs bell company (ATT)?Random?Try rephrasing your question into a more meaningful form.MartyPS Yes everyone here (both of us) integrate asterisk with everything.
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Re: [Asterisk-Users] X100P fails to initialize

 Zapata Telephony Interface Registered on major 196
 Zaptel Version: 1.2.5 Echo Canceller: MG2
 Failed to initailize DAA, giving up...
 wcfxo: probe of :00:0e.0 failed with error -5

these lines means, your x100p is not initialized - therefore cannot 
be used by zaptel. the problem below, reported by ztconfig.
as the usage of zaptel device is following:
modprobe zaptel
modprobe module_of_card (like wcfxo)
if it found, ztcfg
if you see the device in /proc/zaptel/1 (or 2 or so)
you can start asterisk and enjoy the device.

you can believe me, this problem is in relation with irq sharing.
(as i meet with that problem every time i have installed more than one
card in a box - what i did more than 20 times)

  Hello.
 ZT_CHANCONFIG failed on channel 1: No such device or address (6)
 I think that this error, is saying that its X100P is not connected in slot
 PCI correctly. He makes a test, he changes the X100P of slot and he sends
 for the list the commands lspci, dmesg, cat /proc/interrupts. I wait to
 have helped.
 Best Regards
 Josué
 
 2006/6/3, Woodoo People .pGa! [EMAIL PROTECTED]:
 
  So I took a chance with an X100P knock-off on eBay. I'm running Asterisk
 +
  FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and
 kernel
  2.6.16.16. Everything has been fine up until now.
  I compile the 1.2.5 Zaptel drivers without a problem, get the udev
  configuration in, modprobe zaptel, and finally modprobe wcfxo. At this
  point, I get the message:
 
  ZT_CHANCONFIG failed on channel 1: No such device or address (6)
  FATAL: Error running install command for wcfxo
 
  dmesg gives me:
 
  Zapata Telephony Interface Registered on major 196
  Zaptel Version: 1.2.5 Echo Canceller: MG2
  Failed to initailize DAA, giving up...
  wcfxo: probe of :00:0e.0 failed with error -5
 
 this is a problem if the card shares interrupt with something on
 MOtherBOard.
 configure the slot to other irq, or move the card to other irq.
 lspci -v will be your friend. Check if someone uses same IRQ as motorola
 card.
 
 
 --
 WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
 [EMAIL PROTECTED]@
 RedHat.users
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Re: Fw: [Asterisk-Users] Compiling chan_bluetooth

does chan_bluetooth working well now? (integrating sound and signal channels
in BT?) If yes, it's better than using a cheap GSM adapter (like 150Euro)?

ps: i have tested it in last year with nokia6310, but with no luck.

 Just to close the thread. The problem was that I was using an old version 
 of the code.
 If anyone has the same problem, you can download the code from here:
 
 http://www.thetechguide.com/howto/asterisk/bluetoothfiles.tar.gz
 
 Good luck,
Danko
 
 
 
 
 - Original Message - 
 From: Danko Miocevic [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, May 30, 2006 8:48 PM
 Subject: Re: [Asterisk-Users] Compiling chan_bluetooth
 
 
 I´ve found a solution to my problem, I forgot to install the posix 
 development libraries.. now the error has dissapeared to
 make place to a new error! :D I still can´t compile.
 The new error says:
 
 cc: chan_gsm_bt.o: no se usó el fichero de entrada del enlazador porque 
 no se hizo enlace
 cc: -lbluetooth: no se usó el fichero de entrada del enlazador porque no 
 se hizo enlace
 
 It says that the file wasn´t used because the linker didn´t make link... 
 don´t know what to do..
 Any ideas?
 
 Danko
 
 
 - Original Message - 
 From: Danko Miocevic [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Saturday, May 27, 2006 12:51 PM
 Subject: [Asterisk-Users] Compiling chan_bluetooth
 
 
 Hello, I´m trying to use my phone with asterisk to get GSM connectivity 
 but I can´t compile the code.
 I got the asterisk, chan_bluetooth, zaptel and libpri sources, the last 
 two ones compiled perfectly.
 I have added this to the /usr/src/asterisk/channels/Makefile:
include /usr/src/chan_bluetooth/Makefile
 and I´ve also added chan_bluetooth.so to the CHANNEL_LIBS var.
 When I do make install in the asterisk directory I get lots of this 
 error:
 
 /usr/src/chan_bluetooth/chan_bluetooth.c:2469: error: dereferencing 
 pointer to incomplete type
 
 and some others like:
 
 /usr/src/chan_bluetooth/chan_bluetooth.c: En la función 
 `remove_sdp_records':
 /usr/src/chan_bluetooth/chan_bluetooth.c:2937: error: `sdp_session_t' 
 undeclared (first use in this function)
 /usr/src/chan_bluetooth/chan_bluetooth.c:2937: error: `sdp' undeclared 
 (first use in this function)
 /usr/src/chan_bluetooth/chan_bluetooth.c:2938: error: `sdp_list_t' 
 undeclared (first use in this function)
 /usr/src/chan_bluetooth/chan_bluetooth.c:2938: error: `attr' undeclared 
 (first use in this function)
 /usr/src/chan_bluetooth/chan_bluetooth.c:2939: error: `sdp_record_t' 
 undeclared (first use in this function)
 /usr/src/chan_bluetooth/chan_bluetooth.c:2939: error: `rec' undeclared 
 (first use in this function)
 /usr/src/chan_bluetooth/chan_bluetooth.c:2948: aviso: implicit 
 declaration of function `sdp_connect'
 /usr/src/chan_bluetooth/chan_bluetooth.c:2948: error: `BDADDR_ANY' 
 undeclared (first use in this function)
 /usr/src/chan_bluetooth/chan_bluetooth.c:2948: error: `BDADDR_LOCAL' 
 undeclared (first use in this function)
 /usr/src/chan_bluetooth/chan_bluetooth.c:2948: error: `SDP_RETRY_IF_BUSY' 
 undeclared (first use in this function)
 
 I´m working on a stable debian, kernel 2.6 and my asterisk is 1.2. I 
 really don´t know what is happening, if someone
 has an idea I´d be glad to hear it. Thanks for reading,
 
 Danko
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-- 
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Re: [Asterisk-Users] SIP voice recorder

 I believe that Cisco does the monitoring/recording that way. We've been
 working with a company that has implemented Cisco's approach and they
 are having problems with the recording due to network design (eg, high-
 availability dual-everything. Port mirroring is only picking up half the
 conversation).
 
 Their recording method apparently works when it can see both sides of
 the conversation. Don't know anything about their software for that
 function however.

this mirror port problem can be solved if you connect the asterisk
box thru the monitor-box what's operating in bridge mode. that also
make the possibility (if you have a 2port nic) to connect asterisk directly
the switch via another cable, using Spanning Tree with higher cost.

-- 
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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Re: [Asterisk-Users] Recommended Web Interface

2006-06-03 Thread Doug Lytle


Dakota Burns wrote:
I'm currently reviewing the latest release of FreePBX (formerly known 
as [EMAIL PROTECTED]).  Do either of you know whether FreePBX is robust 
enough to handle multiple clients, or have any recommendations on 
front-end Web interface to manage client config
Actually, FreePBX is a front-end that was used by [EMAIL PROTECTED]  And, 
from what I've read today, they are now calling it TrixBox.


http://www.trixbox.org/modules/smartsection/item.php?itemid=5

Doug

-- Ben Franklin quote: Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety.


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Re: [Asterisk-Users] X100P fails to initialize

2006-06-03 Thread Tzafrir Cohen

On Sat, Jun 03, 2006 at 11:15:57PM +0200, Woodoo People .pGa! wrote:
  Zapata Telephony Interface Registered on major 196
  Zaptel Version: 1.2.5 Echo Canceller: MG2
  Failed to initailize DAA, giving up...
  wcfxo: probe of :00:0e.0 failed with error -5
 
 these lines means, your x100p is not initialized - therefore cannot 
 be used by zaptel. the problem below, reported by ztconfig.
 as the usage of zaptel device is following:
 modprobe zaptel
 modprobe module_of_card (like wcfxo)
 if it found, ztcfg
 if you see the device in /proc/zaptel/1 (or 2 or so)
 you can start asterisk and enjoy the device.
 
 you can believe me, this problem is in relation with irq sharing.
 (as i meet with that problem every time i have installed more than one
 card in a box - what i did more than 20 times)

In one at least one board the problem was resolved by adding the boot
parameter 'pci=noacpi' , IIRC.

Which is, indeed, something in the neighbourhood of IRQ assignments.

-- 
Tzafrir Cohen  sip:[EMAIL PROTECTED]
icq#16849755   iax:[EMAIL PROTECTED]
+972-50-7952406   
[EMAIL PROTECTED]  http://www.xorcom.com
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Re: [Asterisk-Users] Size limitations of extensions.conf

On Sat, 2006-06-03 at 16:03 -0500, voiplist wrote:
 Can someone tell me the size (or any other) limitations for the 
 extensions.conf?
 
 We have managed to keep our file pretty small thanks to AGI but we are
 about to setup a bunch of call restrictions based on area and country
 code.
 
 One line per area code in the US alone adds a LOT of text to this file.
 
 Is it a bad thing to have 5 or 6000 lines of text in your
 extensions.conf on a production system?
 
 Will it affect the performance?

it adds memory and increases load time, it also causes asterisk to walk
a longer tree each time it has to do something in that context at least
rather than not ...


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com we pay you to terminate calls with us!


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Re: [Asterisk-Users] Size limitations of extensions.conf

2006-06-03 Thread voiplist


So what are the smart folks doing when it comes to retricting/allowing
which area/country codes can and can't be called?

AGI? We can go AGI but we are trying to avoid yet more calls to AGI
apps for obvious reasons.

So, is it smarter to use AGI or have it in the text file?

Thanks..

On 6/3/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:

On Sat, 2006-06-03 at 16:03 -0500, voiplist wrote:
 Can someone tell me the size (or any other) limitations for the 
extensions.conf?

 We have managed to keep our file pretty small thanks to AGI but we are
 about to setup a bunch of call restrictions based on area and country
 code.

 One line per area code in the US alone adds a LOT of text to this file.

 Is it a bad thing to have 5 or 6000 lines of text in your
 extensions.conf on a production system?

 Will it affect the performance?

it adds memory and increases load time, it also causes asterisk to walk
a longer tree each time it has to do something in that context at least
rather than not ...


--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com we pay you to terminate calls with us!


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Re: [Asterisk-Users] TDM PCI Master Abort

2006-06-03 Thread Stephen Bosch

Jeremy McNamara wrote:
 Stephen Bosch wrote:
 
 I don't know. I'll have to check. Is that a requirement?
 
 
 
 Yes - Most absolutely.
 
 
 http://www.digium.com/en/products/hardware/tdm400p.php

I've confirmed that the board supports PCI 2.2.

I've also updated the BIOS on the motherboard, but the problem is still
there.

I'm going to try moving the card to a different PCI slot (a desperation
move).

-Stephen-
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Adding Asterisk between existing phone system and PSTN Re: [Asterisk-Users] Integrating Asterisk

2006-06-03 Thread Mike Fedyk


Dakota Burns wrote:

What I was attempting to visualize is the following case:
10 people in an organization pick-up their phones to make an outbound 
call.  Before integrating Asterisk, all calls route through their 
current non-VoIP based phone provider.  After  integrating 1 trunk 
from a VoIP service provider  into their system that provides 4 
simultaneous calls (Teliax's Corporate plan)
First of all, don't use a service that limits the number of calls.  Get 
the per minute plan.  That way you won't have to worry about hitting any 
soft-caps.

, and dropping 4 legacy lines
Don't drop the lines until you have a setup that works reliably without 
hickups for at least 3 months (more if you can convince them to keep the 
lines that long).
, if 10 people make calls simultaneously, some will be VoIP and some 
will be legacy based.  Based on the above example, I'm questioning 
whether it would be best to configure a Sipura 3000 for every analog 
phone (I'm guessing  the non-profits will want to keep their existing 
analog phones)
Only use ATAs when you have to.  They cost about $80 anyway, why not get 
a spa-841 instead?  And why are you guessing?  You should know if they 
want to keep the phones they have.  And what type of phones are they?
, or utilize another device (or devices) to connect the company's 
internet service into their existing Trunks or POTS.  I think the 
former would be easier  something I know how to do, but the latter 
may be smarter  more cost effective.  So the latter is what I'm 
questioning whether either of you have experience implementing. 
Let me be frank.  I'm relatively new to phone systems, but I can tell 
you need to do a lot more research before even thinking about doing an 
implementation.


If you want to keep the analog phones, they probably already go to a 
wiring closet.  You'll want to put either an asterisk box with a 
tdm2400p with 12 FXS and 12 FXO (look up the tdm2400p before asking why 
I say 12 instead of 10).  Or if you have voice T1s at that location you 
may want a channel bank instead.  I haven't used any channel banks so 
others will have to step in to give suggestions on that.


My point is that you need to post what you want your client's results to 
be instead of how to do what you think should be done.  The details I 
mentioned above are only part of one possible direction to go in, and 
there is more to it than that also and it may not even be the best for 
your situation too. 

Have you looked at their network to see if can handle the large number 
of small packets that voip produces?  What about their Internet 
connection? What is it that your client wants in a phone system that 
their current one isn't doing?  How is adding asterisk and an ATA for 
each analog phone going to help?


So, post what you already have and what you want the end result to be 
from an end-user's perspective and we can probably point you in the 
right direction.

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[Asterisk-Users] ADIT 600 = Asterisk Help

2006-06-03 Thread Bart Fisher


I've been reading the Google searches trying to understand how to tie
together Adit 600 to Asterisk to provide 2 way service.  I'm about blind 
from reading.


I assume, the answer is using MGCP between the boxes.  However, the examples
I found don't really explain fully enough to know how to modify examples to 
work for me.


I'll have in the ADIT with T1's. There is a CMG and FXS card installed -
later I'd like to add a FXO card.

The goal would be able to route calls to and from ADIT from the T1's to 
Asterisk and route some

Asterisk extensions to the FXS card.

If you have done this, would you mind posting or sending me your mgcp.conf 
with some remarks explaining how  why and a your CMG config?


Thanks for the time

Bart



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Re: [Asterisk-Users] ADIT 600 = Asterisk Help

2006-06-03 Thread Doug Lytle


Bart Fisher wrote:
I assume, the answer is using MGCP between the boxes.  However, the 
examples
I found don't really explain fully enough to know how to modify 
examples to work for me.



Adit 600 TDM to a Digium T1 card



The goal would be able to route calls to and from ADIT from the T1's 
to Asterisk and route some

Asterisk extensions to the FXS card.

I'll be looking to do this as well in a couple weeks. 

I'll be getting a PRI from a local provider and am guessing that I'll 
need a 2nd T1 to handle that and use the 1st T1 to handle the Adit.  I 
need to supply some analog lines to the facility.  But currently, I have 
all analog lines going to the punch down hooked into the Adit, from 
there a T1 cross over cable from the Adit to the PRI card in the 
Asterisk system.  2 FXO and 2 FXS cards.  Around 5 analog phones hanging 
off the FXS.


There is a section on the Wiki on the Adit:

http://www.voip-info.org/wiki/index.php?page=Asterisk%20hardware%20channel%20bank%20check


-- Ben Franklin quote: Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety.


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Re: [Asterisk-Users] Asterisk + PRI Card - Nortel BCM

2006-06-03 Thread Wojciech Tryc

Asterisk with Digium's single span PRI works just fine with BCM. Contact me 
off the list if you need details.

Thanks,
Wojtek
- Original Message - 
From: Mr. Jones [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Saturday, June 03, 2006 1:08 PM
Subject: [Asterisk-Users] Asterisk + PRI Card - Nortel BCM

Has anyone fed a Nortel BCM from Asterisk?

I'm interested in switching our company over, but don't want to
replace all the handsets in one fell swoop.

I imagine some of the PRI cards can emulate a switch?

I'd still like to pass CallerID into the Nortel, etc but all the
external traffic would be VOIP, not TDM.
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[Asterisk-Users] Sangoma A101 configuration

2006-06-03 Thread mhiguera

Im trying to install and configure sangoma ... 
every thing is OK but when type the command wanrouter
start the following error apears:
wan Driver not found.

Thanks for any help
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Re: [Asterisk-Users] Recommended Web Interface

2006-06-03 Thread Philippe Lindheimer

Dakota,freepbx is a web application and associated core dialplan that allows you to do many things on top of asterisk by generating the dialplan customizations ontop of the base that it provides. Once you spend some time understanding it, you can usually do most things that you want within the gui and almost anything else in custom dialplan applications that you can write and will coexist with freepbx. The dialplan foundation is a bit 'fat' because of the rich features set and potential that it can provide but allows for much flexibility. It is not for everyone as it clearly has its pros and cons, but there should be very little, if anything, that you couldn't do using this as a base that you can do on a 'raw' system (since you can always write custom dialplan code).[EMAIL PROTECTED], now named trixbox, has been using freepbx (or amp, which was the previous name for freepbx)
 as its main interface and dialplan, and has also added several other packages as you have mentioned.Freepbx does not provide an environment to easily run mulitple businesses on a single server. It does provide an ability to give different levels of access to different freepbx 'users.' However, I personally do not believe it is a good interface for end users. It may be reasonable to give a non-telephony IT admin or other knowledgable customer access to do certain basic functions, but beyond that it is not really geared for end users, IMHO. However - I think going forward you will see more end users portals that will provide access to change their settings within such an environment, so that eventually you may be able to have an enduser portal where they can set their features (forward, cw, dnd, follow-me settings, voicemail, etc.) on the web in addition to what they can do from the phone.philippe 
 From: "Dakota Burns" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Sat, 3 Jun 2006 15:07:20 -0500Subject: [Asterisk-Users] Recommended Web InterfaceI'm currently reviewing the latest release of FreePBX (formerly known as [EMAIL PROTECTED]). Do either of you know whether FreePBX is robust enough to handle multiple clients, or have any recommendations on front-end Web interface to manage client config  provide clients access to manage their level of access (similar to how Vonage, Teliax, and others provide client access to their web management console)? The latest FreePBX is module driven - pretty cool. I've plans to step through the "Asterisk: Future of Telephony" with old laptop in order to get a different view of Asterisk. Am used to Linux  CLI -- do either of you have any preferences?
 My guess is that some purists may look at FreePBX as a lesser product but ... I think it's simply a product base built right on top of Asterisk to help new Asterisk people hit the ground running (and provides some extras such as SugarCRM, Credit Card app, etc.). Thanks,Dakota __Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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Re: [Asterisk-Users] Integrating Asterisk

2006-06-03 Thread Martin Joseph



On Jun 3, 2006, at 2:04 PM, Dakota Burns wrote:


What I was attempting to visualize is the following case:
10 people in an organization pick-up their phones to make an outbound 
call.  Before integrating Asterisk, all calls route through their 
current non-VoIP based phone provider.  After  integrating 1 trunk 
from a VoIP service provider  into their system that provides 4 
simultaneous calls (Teliax's Corporate plan), and dropping 4 legacy 
lines, if 10 people make calls simultaneously, some will be VoIP and 
some will be legacy based.  Based on the above example, I'm 
questioning whether it would be best to configure a Sipura 3000 for 
every analog phone (I'm guessing  the non-profits will want to keep 
their existing analog phones), or utilize another device (or devices) 
to connect the company's internet service into their existing Trunks 
or POTS.  I think the former would be easier  something I know how to 
do, but the latter may be smarter  more cost effective.  So the 
latter is what I'm questioning whether either of you have experience 
implementing. 


Personally I think it's better to get rid of the POTS lines and got to 
a real VoiP terminator.


I am really an experimenter only,  but my initial goal was to setup a 
way to share my existing PSTN line via an FXO like the wellgate 3701a.  
This turns out to be quite a bit of a problem due to crappy hardware (I 
started with the HT-488 but found it to be useless)  and problems with 
my local loop (ie echo).


Even with all the fussing I have done, I still have a very bad echo for 
the first few seconds of some calls, until the echo can. trains and 
knocks the echo out.


Conversely,  with Voip providers like Teliax (very good), Nufone.net 
(very good), I found that there are no such issues, and the most 
serious QUALITY issues are due to the routing of my data over the 
public internet to these companies.


SO, in conclusion.  Just because a particular Voip terminator is good, 
doesn't mean they will work well for you.  Check the routes to them!  
Having said that, I found a third Voip call terminator that is very 
close to me (sellvoip.net), and have configured that as my primary 
terminator (asterisk will fail over to nufone and teliax if needed).  
This arrangement works great, allows for inward dialing, and is very 
cost efficient.  If I had realized this to begin with, I would have 
skipped the whole PSTN aspect of my setup.


Asterisk is SUPER flexible.  you can set it up to route calls based on 
many criteria.  For example, my setup routes 7 digit calls through my 
PSTN, because I already pay qwest 18$(us) per month, so these calls are 
free.  If I dial 10 digits (US long distance) the calls are routed 
through sellvoip.net. If I dial an Israeli cell phone, the calls are 
routed through teliax (better rate).


Hope this helps a bit.
Marty



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Re: [Asterisk-Users] Integrating Asterisk

2006-06-03 Thread mitcheloc


Just be sure that if you ditch your POTS line that you have a proper
way to terminate 911 calls!

On 6/3/06, Martin Joseph [EMAIL PROTECTED] wrote:


On Jun 3, 2006, at 2:04 PM, Dakota Burns wrote:

 What I was attempting to visualize is the following case:
 10 people in an organization pick-up their phones to make an outbound
 call. Before integrating Asterisk, all calls route through their
 current non-VoIP based phone provider. After integrating 1 trunk
 from a VoIP service provider into their system that provides 4
 simultaneous calls (Teliax's Corporate plan), and dropping 4 legacy
 lines, if 10 people make calls simultaneously, some will be VoIP and
 some will be legacy based. Based on the above example, I'm
 questioning whether it would be best to configure a Sipura 3000 for
 every analog phone (I'm guessing the non-profits will want to keep
 their existing analog phones), or utilize another device (or devices)
 to connect the company's internet service into their existing Trunks
 or POTS. I think the former would be easier  something I know how to
 do, but the latter may be smarter  more cost effective. So the
 latter is what I'm questioning whether either of you have experience
 implementing.

Personally I think it's better to get rid of the POTS lines and got to
a real VoiP terminator.

I am really an experimenter only,  but my initial goal was to setup a
way to share my existing PSTN line via an FXO like the wellgate 3701a.
This turns out to be quite a bit of a problem due to crappy hardware (I
started with the HT-488 but found it to be useless)  and problems with
my local loop (ie echo).

Even with all the fussing I have done, I still have a very bad echo for
the first few seconds of some calls, until the echo can. trains and
knocks the echo out.

Conversely,  with Voip providers like Teliax (very good), Nufone.net
(very good), I found that there are no such issues, and the most
serious QUALITY issues are due to the routing of my data over the
public internet to these companies.

SO, in conclusion.  Just because a particular Voip terminator is good,
doesn't mean they will work well for you.  Check the routes to them!
Having said that, I found a third Voip call terminator that is very
close to me (sellvoip.net), and have configured that as my primary
terminator (asterisk will fail over to nufone and teliax if needed).
This arrangement works great, allows for inward dialing, and is very
cost efficient.  If I had realized this to begin with, I would have
skipped the whole PSTN aspect of my setup.

Asterisk is SUPER flexible.  you can set it up to route calls based on
many criteria.  For example, my setup routes 7 digit calls through my
PSTN, because I already pay qwest 18$(us) per month, so these calls are
free.  If I dial 10 digits (US long distance) the calls are routed
through sellvoip.net. If I dial an Israeli cell phone, the calls are
routed through teliax (better rate).

Hope this helps a bit.
Marty



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Re: [Asterisk-Users] X100P fails to initialize

2006-06-03 Thread Lachek Butalek


Thanks to everyone for their tips and suggestions. I finally got the
card working by using the YellowDog Linux kernel from ppckernel.org.
There must have been some setting in the kernel config that made a
difference because the card suddenly started working after that, even
after a kernel recompile of 2.6.16.18 using the same config.

On 6/3/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:

On Sat, Jun 03, 2006 at 11:15:57PM +0200, Woodoo People .pGa! wrote:
  Zapata Telephony Interface Registered on major 196
  Zaptel Version: 1.2.5 Echo Canceller: MG2
  Failed to initailize DAA, giving up...
  wcfxo: probe of :00:0e.0 failed with error -5

 these lines means, your x100p is not initialized - therefore cannot
 be used by zaptel. the problem below, reported by ztconfig.
 as the usage of zaptel device is following:
 modprobe zaptel
 modprobe module_of_card (like wcfxo)
 if it found, ztcfg
 if you see the device in /proc/zaptel/1 (or 2 or so)
 you can start asterisk and enjoy the device.

 you can believe me, this problem is in relation with irq sharing.
 (as i meet with that problem every time i have installed more than one
 card in a box - what i did more than 20 times)

In one at least one board the problem was resolved by adding the boot
parameter 'pci=noacpi' , IIRC.

Which is, indeed, something in the neighbourhood of IRQ assignments.

--
Tzafrir Cohen  sip:[EMAIL PROTECTED]
icq#16849755   iax:[EMAIL PROTECTED]
+972-50-7952406
[EMAIL PROTECTED]  http://www.xorcom.com
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Re: [Asterisk-Users] ADIT 600 = Asterisk Help

2006-06-03 Thread C F


The Adit is realy simple as all it is is a bridge, so you have one
interface that is virtualy cross connected to another interface within
the Adit.
If you want to use the Adit with asterisk you can put the cards into
the Adit (usualy FXS and/or FXO cards) and then connect the Adit to
Asterisk, in most cases the Adit is connected to Asterisk using a T1
cable that is connected to a T1 card (like from Digium), however you
can also use the CMG card from the Adit, but keep in mind that the CMG
card only works with FXS cards and not with FXO.


On 6/3/06, Bart Fisher [EMAIL PROTECTED] wrote:

I've been reading the Google searches trying to understand how to tie
together Adit 600 to Asterisk to provide 2 way service.  I'm about blind
from reading.

I assume, the answer is using MGCP between the boxes.  However, the examples
I found don't really explain fully enough to know how to modify examples to
work for me.

I'll have in the ADIT with T1's. There is a CMG and FXS card installed -
later I'd like to add a FXO card.

The goal would be able to route calls to and from ADIT from the T1's to
Asterisk and route some
Asterisk extensions to the FXS card.

If you have done this, would you mind posting or sending me your mgcp.conf
with some remarks explaining how  why and a your CMG config?

Thanks for the time

Bart



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Re: [Asterisk-Users] Size limitations of extensions.conf

2006-06-03 Thread C F


AFAIK 7000 lines of extesnsion.conf will not eat as much memory as an
AGI script half that long will.
Second, there is no reason that 1000 lines of code (IF you would be
adding one line for every possible area code in North America then it
would be around 800 lines, then give another 200 for the macro to
handle it) should slow down an asterisk in such a way that you should
consider going with AGI instead.

On 6/3/06, voiplist [EMAIL PROTECTED] wrote:

So what are the smart folks doing when it comes to retricting/allowing
which area/country codes can and can't be called?

AGI? We can go AGI but we are trying to avoid yet more calls to AGI
apps for obvious reasons.

So, is it smarter to use AGI or have it in the text file?

Thanks..

On 6/3/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
 On Sat, 2006-06-03 at 16:03 -0500, voiplist wrote:
  Can someone tell me the size (or any other) limitations for the 
extensions.conf?
 
  We have managed to keep our file pretty small thanks to AGI but we are
  about to setup a bunch of call restrictions based on area and country
  code.
 
  One line per area code in the US alone adds a LOT of text to this file.
 
  Is it a bad thing to have 5 or 6000 lines of text in your
  extensions.conf on a production system?
 
  Will it affect the performance?

 it adds memory and increases load time, it also causes asterisk to walk
 a longer tree each time it has to do something in that context at least
 rather than not ...


 --
 Trixter http://www.0xdecafbad.com Bret McDanel
 Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
 Utrecht NL +31 306 553058  US WA +1 360 207 0479
 US NY +1 516 687 5200  FreeWorldDialup: 635378
 http://www.trxtel.com we pay you to terminate calls with us!


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 Version: GnuPG v1.4.3 (GNU/Linux)

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 UoVBXbLKIPOL1TbXuFCvlo0=
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[Asterisk-Users] Sipura SPA-941 not available after Asterisk Freepbx upgrade

2006-06-03 Thread David K Parker

I'm experiencing a problem with a Sipura SPA-941 not available for incoming calls after Asterisk Freepbx upgrade. I can dial out with the phone gto any other internal or external ext. It is registered with the Asterisk server. When I dial the Sipura directly from any other extension, it goes directly to vm. I have other Sip softphones that are working fine. A sip debug when calling the Sipura indicates Extension 200 is not available to be called then goes to vm. This configuration had been working fine for over six months prior to the upgrade.
asterisk1*CLI sip show peer 200asterisk1*CLI * Name : 200 Secret : Set MD5Secret : Not set Context : from-internal Subscr.Cont. : Not set
Language : AMA flags : Unknown CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : [EMAIL PROTECTED] VM Extension : asterisk LastMsgsSent : 3
Call limit : 0 Dynamic : Yes Callerid : David 200 Expire : 12789 Insecure : no Nat : No ACL : No CanReinvite : No PromiscRedir : No
User=Phone : No Trust RPID : No Send RPID : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 192.168.1.174 Port 5060
Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 200 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (ulaw,alaw) Status : Unmonitored
Useragent : Sipura/SPA941-4.1.8 Reg. Contact : sip:[EMAIL PROTECTED]:5060
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Re: [Asterisk-Users] SIP Trunking

2006-06-03 Thread C F


It should work as is, just make usre that you have an extension
defined (or a catch all) for every DID you have with the provider so
that incoming works.

On 6/2/06, Steven Haldeman [EMAIL PROTECTED] wrote:


Hello,

I am attempting to figure out how to set up SIP trunking, between my company
and our SIP provider.  This is an expermintal project at this time.  The SIP
provider gave us a Signalling IP address and two Media IP addresses.  We
supplied them with the IP address of our Asterisk box.  When asked what our
Usernames and Passwords would be we were told that they were not needed for
a SIP trunk.  We can use what they call SIP lines that use username/password
however because of tarrifing the lines cost more per month than a trunk I
have been successfull in making a SIP Line work, but have no idea where to
start with a SIP Trunk.  We sill be using DID numbers on the SIP trunks.
Has anyone had any experiance with this type of configuration an example
would be extrememly helpfull.

I have search the internet for help and I may have seen a solution but just
was not certain what I was looking at, or how to implement as everything
that I have seen user a Username/Password combo.

Thank you in advance.

Thanks

Steven


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Re: [Asterisk-Users] Asterisk - Qsig

2006-06-03 Thread Josué Conti

Hello Michael, thank´s for help.
But what´s version asterisk you use? The qsig protocol supported for what version?

Best Regards

Josué
2006/6/3, Michael Konietzny [EMAIL PROTECTED]:
Hello Josué,yes i currently only switched switchtype in zapata.conf to the valueqsig. The only real PRI feature i've found out is the PRI_CAUSE
variable set on Hangup().Greetings,MichaelJosué Conti schrieb: Michael, thank´s for this attention. I go to test with equipment Siemens HiPath and features. I sending for you and the list an email with the results of the tests, ok? How was
 zapata.conf of its asterisk with qsig? You it only changed switchtype=euroisdn, for switchtype=qsig? Thank you for its attention. Greetings Josué 2006/6/3, Michael Konietzny 
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hello Josué, we're running Asterisk in combination of the T-Com Octopus E800 with
 QSig Protocoll. The protocoll itself is supported but some features are missing, or i didn't found out yet how to use them. I'm also interested in how to use qsig for determinating if other phones are available for calling and so on.
 Greetings, Michael Josué Conti schrieb:  Hello all, as good?  It would like to make a question, asterisk supports the protocol qsig,
  for interconnections in ISDN with equipment Siemens HiPath 4000 or  same Ericsson MD110, so that it could identify to the name and the  number of hosts and also to use some features of asterisk in the
  Siemens/Ericsson equipment.  Greetings  Josué   
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 mailto:[EMAIL PROTECTED] handy: 0176 / 24 79 8656 phone: 03529 / 527597 address:
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http://lists.digium.com/mailman/listinfo/asterisk-users--Mit freundlichen GrüßenMichael Konietznye_mail:[EMAIL PROTECTED]
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Re: [Asterisk-Users] SIP Trunking

2006-06-03 Thread Steven Haldeman

Thank you for your response.All that I get when I dial in is all circutes are busy and when I dial out 503 errors. Here are my configs. Any ideas would be greatly appreciated. The provider is using a Tekelec 9000 Class 5 switch if that is any help.The provider sat us up two accounts one that they are calling a line that uses authentication usernames and passwords as though we were terminating into a SIP phone. This account works but is not the prefered solution to our problem. The other account is what the provider calls a trunk account and it does not use usernames and passwords. This seems to be the solution for the provider and us. Thank you, Steven sip.conf
[inbound-trunk]type=friendcontext=incominginsecure=veryhost=xxx.xxx.xxx.xxxoutboundproxy=xxx.xxx.xxx.xxxfromdomain=xxx.xxx.xxx.xxxdefaultip=xxx.xxx.xxx.xxxdisallow=allallow=ulawnat=yescanreinvite=yesqualify=yes extension.conf[incoming]exten = NXXNXX,1,Answer()exten = NXXNXX,2,Background(greeting)exten = NXXNXX,3,SayDigits(${CALLERIDNUM})exten = NXXNXX,4,Dial(SIP/steven)exten = NXXNXX,5,Hangup()C F [EMAIL PROTECTED] wrote: It should work as is, just make usre that you have an extensiondefined (or a catch all) for every DID you have with the provider sothat incoming works.On 6/2/06, Steven Haldeman <[EMAIL PROTECTED]>wrote:
Hello, I am attempting to figure out how to set up SIP trunking, between my company and our SIP provider. This is an expermintal project at this time. The SIP provider gave us a Signalling IP address and two Media IP addresses. We supplied them with the IP address of our Asterisk box. When asked what our Usernames and Passwords would be we were told that they were not needed for a SIP trunk. We can use what they call SIP lines that use username/password however because of tarrifing the lines cost more per month than a trunk I have been successfull in making a SIP Line work, but have no idea where to start with a SIP Trunk. We sill be using DID numbers on the SIP trunks. Has anyone had any experiance with this type of configuration an example would be extrememly helpfull. I have search the internet for help and I may have seen a solution but just was not
certain what I was looking at, or how to implement as everything that I have seen user a Username/Password combo. Thank you in advance. Thanks Steven __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options
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[Asterisk-Users] Re: Sipura SPA-941 not available after Asterisk Freepbx upgrade

2006-06-03 Thread David K Parker

I finally had to give up on extension 200. I tried deleting/recreating and reloading sipura and asterisk but no luck. I had to go to a different extension for the line 1. line 2 never acted up. The new ext works fine.
On 6/3/06, David K Parker [EMAIL PROTECTED] wrote:
I'm experiencing a problem with a Sipura SPA-941 not available for incoming calls after Asterisk  Freepbx upgrade. I can dial out with the phone gto any other internal or external ext. It is registered with the Asterisk server. When I dial the Sipura directly from any other extension, it goes directly to vm. I have other Sip softphones that are working fine. A sip debug when calling the Sipura indicates Extension 200 is not available to be called then goes to vm. This configuration had been working fine for over six months prior to the upgrade.

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[Asterisk-Users] Asterisk 1.2.8 - WARNING[28769]: rtp.c:500 ast_rtp_read: Forcing Marker bit, because SSRC has changed

2006-06-03 Thread Erick Perez


While sending calls to a SIP provider, the following warning generates:

   -- Executing Dial(SIP/1000-c317,
SIP/[EMAIL PROTECTED]:5060|55|o) in new stack
   -- Called [EMAIL PROTECTED]:5060
   -- SIP/209.120.202.94:5060-0533 is making progress passing it to
SIP/1000-c317
   -- SIP/209.120.202.94:5060-0533 answered SIP/1000-c317
   -- Attempting native bridge of SIP/1000-c317 and
SIP/209.120.202.94:5060-0533
Jun  3 22:56:09 WARNING[28769]: rtp.c:500 ast_rtp_read: Forcing Marker
bit, because SSRC has changed

However the calls complete correctly.
I'm using 1.2.8 asterisk stable release.

what does that mean?

Thanks,


Erick.
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Re: [Asterisk-Users] Asterisk 1.2.8 - WARNING[28769]: rtp.c:500 ast_rtp_read: Forcing Marker bit, because SSRC has changed


- Erick Perez [EMAIL PROTECTED] wrote:

 Jun  3 22:56:09 WARNING[28769]: rtp.c:500 ast_rtp_read: Forcing
 Marker
 bit, because SSRC has changed
 
 However the calls complete correctly.
 I'm using 1.2.8 asterisk stable release.

It's a message that should not have been marked WARNING (or even generated at 
all unless the verbose level was adequately high). I've already changed this in 
Subversion and the next release will not generate this message unless 
requested. There is nothing to be concerned about.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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[Asterisk-Users] Meetme versus app_conference

2006-06-03 Thread Erick Perez


As stated here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe

A Meetme room uses Ulaw as the audio codec, so if the other channels
use different codecs, then * will transcode.

Does the app_conference application works the same way?
Or if i have SIP/g729 users and i create a conference with other users
also at g729 asterisk will not transcode (when using app_conference)?

Thanks,


--

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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Re: [Asterisk-Users] Meetme versus app_conference


- Erick Perez [EMAIL PROTECTED] wrote:

 Or if i have SIP/g729 users and i create a conference with other
 users
 also at g729 asterisk will not transcode (when using app_conference)?

It is not possible to mix conference audio together without converting it to an 
uncompressed form first. app_meetme in Asterisk 1.2.x certainly does do more 
transcoding (both inbound and outbound) than is absolutely needed, which 
app_conference does not do. However, app_meetme in SVN trunk (soon to be 
Asterisk 1.4) tries to minimize the amount of transcoding by avoiding the 
decoding of incoming audio from channels that are not speaking and by re-using 
the transcoded output for channels that share a format (codec). This should 
make it perform similarly to app_conference in many situations.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Meetme versus app_conference