[Asterisk-Users] DID Providers

2005-12-09 Thread Aaron Anderson

Gentelmen (and ladies too of course),

Just a quick question.

I run an internet provider here in Japan and we want to start offering 
US DIDs to some of our US military customers.


Does anyone have a link to some good information about DIDs and setting 
them up under asterisk?  Also, perhaps a few links to providers of DIDs 
in the US so I can get some rates for numbers?


Thanks in advance
Aaron
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RE: [Asterisk-Users] Asterisk on PPC chan_capi issue

2005-12-09 Thread David Waugh
Sorry Patrick,
I was mistaken here. 
The Diva Server for Linux drives currently only support Little Endian machines. 
Unfortunately the PPC based chipsets use Big Endian.

There is a discussion about this here:

http://www.cs.umass.edu/~verts/cs32/endian.html
Thanks
David

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Patrick
Sent: 08 December 2005 21:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk on PPC  chan_capi issue


On Thu, 2005-12-08 at 08:47 +, David Waugh wrote:
 Hello Patrick,
 
 I have an Eicon Diva PRI-30M card and use the Eicon Linux drivers with 
 chan_capi_cm.
 I am able to do ISDN to SIP calls with this.
 
 Have you tried using the Eicon drivers instead, rather than zaptel and zib 
 pri.
 
 Instruction for doing this can be found here:
 http://www.voip-info.org/wiki/view/Asterisk+Eicon+Diva+CAPI+ISDN
 
 I hope this help

Unfortunately not. The config I described below works fine on an Intel
server. The box showing the issue is a PPC box. The author of
chan_capi-cm doesn't think there is anything wrong with chan_capi-cm on
PPC and the way it behaves in the logfiles below. So the problem seems
to be Asterisk  PPC related but I have no idea how to track down the
issue or solve it.

Regards,
Patrick

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Patrick
 Sent: 06 December 2005 01:40
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Asterisk on PPC  chan_capi issue
 
 
 Hi all,
 
 I have a PPC box (IBM RS6000 43P-150, bigendian afaik) which runs Fedora
 Core 5 Test1 and zaptel, libpri and asterisk 1.2.0. I also installed
 chan_capi (0.6.1) so I can use my Eicon Diva Server BRI card. Asterisk
 was compiled with DEBUG=-g and DEBUG_THREADS = -DDUMP_SCHEDULER
 -DDEBUG_SCHEDULER-DDEBUG_THREADS -DDO_CRASH -DDETECT_DEADLOCKS. Next I
 did make clean, make valgrind, make install. Asterisk runs as user/group
 asterisk/asterisk.
 
 SIP -- SIP calls are fine, Calls from SIP out to the PSTN via
 CAPI/ISDN are fine too. ISDN/CAPI -- SIP calls don't work. Example
 output of the issue is below. Anyone have an idea how I fix this?
 
 Thanks and regards,
 Patrick
 
 
 chan_capi registers fine:
 **
  [chan_capi.so] = (Common ISDN API for Asterisk)
   == This box has 1 capi controller(s).
   == Reading config for BRI1
 -- ast_capi_pvt BRI1-pseudo-D (MSN1,MSN2,capi-in,0,2) (1,4,128)
 -- ast_capi_pvt BRI1 (MSN1,MSN2,capi-in,0,2) (1,4,128)
 -- ast_capi_pvt BRI1 (MSN1,MSN2,capi-in,0,2) (1,4,128)
 -- listening on contr1 CIPmask = 0x1fff03ff
   == Registered channel type 'CAPI' (Common ISDN API Driver ($Revision:
 1.115 $) )
   == Registered application 'capiCommand'
   == Registered custom function VANITYNUMBER
 
 Call from my GSM to a SIP phone (exten 1003) via ISDN/CAPI (MSN2):
 **
   == BRI1: Incoming call 'my GSM' - 'MSN2'
 
 -- Executing Macro(CAPI/BRI1/MSN2-0, stdexten|1003|SIP/1003)
 in new stack
 -- Executing Dial(CAPI/BRI1/MSN2-0, SIP/1003|10|TtwW) in new
 stack
 Dec  6 02:30:47 WARNING[28889]: channel.c:2494 ast_request: No
 translator path exists for channel type SIP (native 65535) to 0
 Dec  6 02:30:47 NOTICE[28889]: app_dial.c:1011 dial_exec_full: Unable to
 create channel of type 'SIP' (cause 0 - Unknown)
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Executing Goto(CAPI/BRI1/MSN2-0, s-CHANUNAVAIL|1) in new
 stack
 -- Goto (macro-stdexten,s-CHANUNAVAIL,1)
 -- Executing Goto(CAPI/BRI1/MSN2-0, s-NOANSWER|1) in new stack
 -- Goto (macro-stdexten,s-NOANSWER,1)
 -- Executing Answer(CAPI/BRI1/MSN2-0, ) in new stack
   == BRI1: Answering for 703241494
 -- Executing Wait(CAPI/BRI1/MSN2-0, 1) in new stack
 Dec  6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping
 incompatible voice frame on CAPI/BRI1/MSN2-0 of format alaw since our
 native format has changed to unknown
 Dec  6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping
 incompatible voice frame on CAPI/BRI1/MSN2-0 of format alaw since our
 native format has changed to unknown
 
 [snipped tons more of these]
 
 Dec  6 02:30:48 NOTICE[28889]: channel.c:1893 ast_read: Dropping
 incompatible voice frame on CAPI/BRI1/MSN2-0 of format alaw since our
 native format has changed to unknown
 -- Executing VoiceMail(CAPI/BRI1/MSN2, u1003) in new stack
 Dec  6 02:30:48 WARNING[28889]: channel.c:2313 set_format: Unable to
 find a codec translation path from unknown to gsm
 Dec  6 02:30:48 WARNING[28889]: file.c:820 ast_streamfile: Unable to
 open vm-theperson (format unknown): No such file or directory
   == BRI1: CAPI Hangingup
 CAPI INFO 0x3490: Normal call clearing
 
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[Asterisk-Users] PRI billing signalization

2005-12-09 Thread Tomislav Parčina
My local telephone provider on PRI lines gives billing signalling. Is there any 
way to use this signalization? I would like to store those information's in 
database (MySQL). Has anybody done something similar? So far I have export CDR 
that Asterisk generates, in MySQL. Those information I'll will use for double 
check of CDR and billing. 

Any information's are welcome.



--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
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[Asterisk-Users] Re: Is Polycom 500CS with P# 2201.11500.001 SIP capable?

2005-12-09 Thread Alphonse Ogulla
On 12/8/05, Alphonse Ogulla [EMAIL PROTECTED] wrote:
Greetings All,

I intend to buy a Polycom IP 500CS with part number 2201.11500.001 but
I'm not sure if it will work with Asterisk. Is this a SIP capable
phone? Moreover, what does CS stand for? Done some research at
http://www.voip-info.org/wiki/view/Polycom+Phones
 but the original info
on part numbers and supported protocols seems to have been erased.

Note that I currently have a Polycom IP300 SIP phone with part number
2201.11300.001 and it works perfectly with Asterisk. Please assist if
you have a phone similar to the one mentioned above.

-- 
Is the Polycom 500CS with part number 2201.11500.001 a SIP capable
phone? Will appreciate your expert opinion before I purchase it.
--
Thanks,
Alphonse

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[Asterisk-Users] Re: No application 'MeetMe' for extension

2005-12-09 Thread Evert Meulie

Found it! For some reason [EMAIL PROTECTED] had chosen to build itself without 
app_meetme.so!

After building this module by hand, all worked!  :-)

  Evert


Evert Meulie wrote:

Read before you reply...  ;-)

To be 100% clear on zaptel/ztdummy, here's the output of my lsmod:

[EMAIL PROTECTED] ~]# lsmod
Module  Size  Used by
md5 8001  1
ipv6  240097  16
autofs422085  0
i2c_dev14273  0
i2c_core   25921  1 i2c_dev
sunrpc139173  1
ztdummy 7748  0
wctdm  40640  0
wcfxo  16928  0
wcte11xp   30496  0
wct1xxp20768  0
wct4xxp57792  0
tor2   93472  0
zaptel196612  7 
ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2

crc_ccitt   6081  1 zaptel
microcode  11873  0
dm_mirror  28449  0
dm_mod 58949  1 dm_mirror
button 10449  0
battery12869  0
ac  8773  0
uhci_hcd   32729  0
ehci_hcd   31813  0
hw_random   9557  0
snd_azx20801  0
snd_hda_codec  75844  1 snd_azx
snd_pcm_oss52345  0
snd_mixer_oss  21825  1 snd_pcm_oss
snd_pcm91973  3 snd_azx,snd_hda_codec,snd_pcm_oss
snd_timer  27973  1 snd_pcm
snd56997  6 
snd_azx,snd_hda_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer

soundcore  12961  1 snd
snd_page_alloc 13641  2 snd_azx,snd_pcm
8139too27329  0
mii 8641  1 8139too
ext3  118729  2
jbd59481  1 ext3
ata_piix   13253  3
libata 47901  1 ata_piix
sd_mod 20545  4
scsi_mod  116429  2 libata,sd_mod



Kunal Parikh wrote:


Hi Evert,

Do you have the zaptel/ztdummy modules installed ?


Kunal

On 12/8/05, *Evert Meulie* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Hi all!

I have an [EMAIL PROTECTED] 2.1 setup here which is working 99% the way
it should. The only thing that does not work is Meetme/Conferences...

In the log-file I see:

Dec 8 11:51:28 WARNING[3288] pbx.c: No application 'MeetMe' for
extension (from-internal, 8125, 6)


This is when I dial 8125 from extension 125. 8125 is defined in the
meetme(-additional).conf

And before you ask: yes, ztdummy is loaded...


Who has any suggestions?  I'm stumped...   :-/


Regards,
Evert

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[Asterisk-Users] Re: ztdummy on FC4

2005-12-09 Thread Tony Mountifield
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote:
 I'm running asterisk on FC4.  All works fine, including musiconhold.
 I tried installing ztdummy as directed, since the documentation
 indicates that ztdummy is required for good music quality.
 
 However, installing ztdummy on FC4 causes moh to play very slow.
 If I remove ztdummy, all works okay again.

Check ztdummy.h and ztdummy.c and make doubly sure USE_RTC is defined.
It sounds like you are using a version compiled without USE_RTC on one
of the newer kernels that has fewer than 1000 jiffies per second.

 I used to use ztdummy before FC4, on a 2.4 kernel, but now I cannot.
 
 Do I really need it?

If you do not have any zaptel cards in the system,then ztdummy is
required for anything that requires local timing. That includes
MeetMe, IAX2 trunking and possibly a few other things.

 Will I need it when I implement conference calling?

Yes, if you use MeetMe to do it.

 Anyone have a similiar FC4 experience?

Sorry, only used FC1 and FC3.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Re: Core dumps since 1.2.0

2005-12-09 Thread Tony Mountifield
In article [EMAIL PROTECTED],
Ryan Laginski [EMAIL PROTECTED] wrote:
 -=-=-=-=-=-
 -=-=-=-=-=-
 
 Hi,
 Ever since upgrading to 1.2.0, Asterisk occasionally core dumps. I'm
 currently on 1.2.1 with the same problem.
 It crashes when an incoming call (zap) dials an extension. It will ring the
 extension short, then crash.
 
 Here is the backtrace:
 #0  0x40187e06 in mallopt () from /lib/libc.so.6
 (gdb) backtrace
 #0  0x40187e06 in mallopt () from /lib/libc.so.6
 #1  0x40186fb3 in malloc () from /lib/libc.so.6
 #2  0x4018ba60 in strdup () from /lib/libc.so.6
 #3  0x0805777a in ast_verbose (fmt=0x40246300 ) at logger.c:886
 #4  0x4069edc6 in skinny_call (ast=0x8189250, dest=0x49445554 Address
 0x49445554 out of bounds, timeout=0) at chan_skinny.c:1980
 #5  0x080631e0 in ast_call (chan=0x8189250, addr=0x49445554 Address
 0x49445554 out of bounds, timeout=1229215060) at channel.c:2547
 #6  0x407a6afe in dial_exec_full (chan=0x81879d0, data=0x8189250,
 peerflags=0xbd5f900c) at app_dial.c:1102
 #7  0x407a3dc5 in dial_exec (chan=0x49445554, data=0x49445554) at
 app_dial.c:1600
 #8  0x0808dfa5 in pbx_extension_helper (c=0x81879d0, con=0x49445554,
 context=0x8187b20 menu-bell, exten=0x8187c14 2, priority=2, label=0x0,
 callerid=0x0, action=0) at pbx.c:544

That looks like something is overflowing a text buffer. The address
0x49445554 appears to be a pointer that was overwritten with 'TUDI'.

 #9  0x0808eb4a in __ast_pbx_run (c=0x81879d0) at pbx.c:2220
 #10 0x0808c7b4 in ast_pbx_run (c=0x81879d0) at pbx.c:2546
 #11 0x406c7417 in ss_thread (data=0x81879d0) at chan_zap.c:6132
 #12 0x40025e51 in pthread_start_thread () from /lib/libpthread.so.0
 #13 0x401ed92a in clone () from /lib/libc.so.6
 
 Any ideas are welcome.

The first place I would look is in__ast_pbx_run(),and after that, in
pbx_extension_helper(). But it could be anywhere.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Aastra firmware 1.3.x. Solution to Far-End sound level issue

2005-12-09 Thread BennyBad








Hi list.



I just want to share this information with all the
Aastra IP phone users that has or am going to switch to FirmWare version 1.3.x.



Ive just installed a bunch of 480i phones
connected to a local Asterisk 1.0.9. Using the pre installed 1.2.x firmware the
sound quality and the sound level at the Far-End was perfect. Installing FW
1.3.x the sound quality remained the same but the sound level at the Far-End was
reduced by some 40%.



Contacting Aastra resulted in this very usefull
information.



Start quote;



The audio issue you mention is a known issue as a
result of some audio adjustments we've added in 1.3.

The audio properties have been adjusted slightly in the 1.3 firmware to
reduce side-tone and echo on the local and far-end equipment. In line with
these adjustments we have added some configurable parameters that allow
users to configure their own audio settings to best suit their needs. These
parameters are only configurable via the .cfg file via a TFTP server and use
the following syntax:

headset tx gain:
headset sidetone gain:
handset tx gain:
handset sidetone gain:
handsfree tx gain:

Each of these parameters can be adjusted by + / - 10 db

Example 1:
headset tx gain: -5 (reduce the headset transmit gain by 5 db)

Example 2:
handset tx gain: 10 (increase the handset transmit gain by 10 db)

Please try adjusting these parameter.



End quote;






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Re: [Asterisk-Users] Sip behind the NAT

2005-12-09 Thread Wilson Pickett
 i have an asterisk box behind the NAT ,when i try to
 send calls through Sip to the voip provider server the
 call is answered but in a one way calling,I hear  the
 voice of the other side just for 4 seconds and then
 stop but the call do not hangup.

SOmetimes this can be due to the client using silence suppression.
Make sure this function if OFF.
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[Asterisk-Users] SIP Canreinvite

2005-12-09 Thread Giordano Grandis








Hi all,

Im testing canreinvite = yes in my sip.conf
with snom190 and a Atcom320.Atcom320 seems support re-invite, but the snom190?

Does anyone known if this phone support it?



How I can be sure that it works?



Giordano 








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RE: [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel

2005-12-09 Thread asterisk-users
Dakota,

Looking at it objectively, Asterisk has many benefits over traditional PBX
systems, yet you should be aware of some of the limitations.

Benefits:
1. Open source / low-cost of ownership / operates on cheap PC hardware. You
get voicemail, IVR, hunt-groups etc. without additional fees. Last I checked
those are all expensive add-ons in the Nortel world. There aren't expensive
licenses per user/handset either. 

2. Flexibility - you can configure Asterisk to handle calls to a microscopic
degree of precision. This is just not possible with traditional PBX systems
which are inherently proprietary. Asterisk also makes it easier to present
data to callers from CRM, Billing, Order Tracking systems etc. using
text-to-speech, automated-speech recognition and/or DTMF recognition. 

3. Flexibility again - It really is much more flexible than anything else!!

4. Supports multiple VoIP protocols - SIP, IAX, H323, (and skinny to a
degree) and supports connection of a broad spectrum of third party handsets
- e.g. Cisco, Siemens, Sipura, etc. IAX is a proprietary protocol for
Asterisk but it has some benefits over SIP (supposedly - my experience has
been a little different) and perhaps more importantly is gaining popularity
among VoIP service providers.



Limitations:

1. Digium PSTN interface boards are not as cheap as they could be and
haven't been around long enough for us to have meaningful data on how
reliable they are.

2. Complexity. Asterisk is powerful but it is complicated - which is it You
will need to spend a few weeks solidly learning about Asterisk and playing
with it in a test environment before even thinking about trying to install
it in a production environment. Clearly your time has a cost to your
employer - thus this may be perceived as problem with Asterisk. You can of
course buy in the services of an Asterisk consultant to help set things up -
but ideally you want to have someone on site with some degree of knowledge
about Asterisk's capabilities. If your business has basic telephony
requirements, doesn't need fancy features and wants to minimize the need for
on-site technical expertise to support Asterisk, then a Mitel/Nortel
solution MIGHT make sense. IMHO - the present level of
complexity/flexibility is the biggest strength and weakness to Asterisk.

3. Asterisk is a work in progress. Yes it's pretty stables and yes it's
being used in very large production systems from what one hears on this
list. However it's a moving target with new releases appearing frequently.
On a positive note that's great if you want new features and bug fixes - but
it can also be a pain if you want a nice stable, low-maintenance system.

4. Cost savings aren't necessarily as great at they first seem. You ideally
want to have redundancy on your Asterisk set up. To support 75 users you
probably want to have a couple of decent Dual-proc Pentium Xeon servers.
Sure you can build these cheap - but if your company is like mine you'll
probably buy from Dell/HP etc. which can make that a not-insignificant
investment. Then you'll need 2x PSTN interface cards for each machine.
Depending on your PSTN lines there this can cost anywhere from $800 - $3000
per card. So overall you can be talking perhaps upwards of $10,000 for the
hardware to support your asterisk installation. Handsets would obviously
cost more though you have the flexibility to choose any pretty SIP/IAX
handsets you like.


Hope these observations help. 

N




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dakota
Sent: Tuesday, December 06, 2005 5:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel

How does Asterisk compare to Nortel, NorthStar and Mitel PBX systems?
For a medium size company not growing past 75 extensions, would you
recommend Asterisk? 

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Re: [Asterisk-Users] Call simulators

2005-12-09 Thread Lenz



Hi Rob,
you could build a simple Perl or Python script to create incoming calls
using callfiles. We have used such a strategy and it seems to be working.
l.


On Thu, 08 Dec 2005 14:15:50 +0100, Rob Hillis [EMAIL PROTECTED]  
wrote:


I'm currently starting development of an add-on to a program designed to  
be used in a call-centre type environment that will interface very  
closely with Asterisk - quite possibly to the point that the add-on  
itself will be a softphone as well.


In order to test this application properly, I find myself needing to  
generate a constant volume of calls to a queue.  I can do this by  
dialling from the two test extensions I have set up on my system, but it  
would seem a better way of doing this would be to have an external  
application randomly generate calls at a certain volume.


My budget is not big - this is a project for a non-profit volunteer  
organisation I do a lot of work with so I would obviously prefer  
something open source.  The ability to randomly generate caller ID and  
intermittently suppress caller ID would be a *very* useful addition.


Does anyone know of any software that would fit this bill?  If such  
software doesn't exist, or is beyond my capacity to afford, what other  
options might I have?  My test rig is my home PABX - a very small setup  
running with three ATAs and two VoIP trunks.  It would seem that  
simulating a trunk would be the best way of doing this, but again, I  
don't know what is available.


Any help would be gratefully received.
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--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it

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Re: [Asterisk-Users] Bristuff / Junghanns / Customer Service

2005-12-09 Thread Frederic Steinfels

Tobias Jönsson wrote:


On Thu, 17 Nov 2005, Frederic Steinfels wrote:

Last January I told KPJ that I can still not use my Simens Gigagaset 
cordless phones and sent him some bug reports. He promised me to fix 
this bug several times but nothing happened. The problem is that the 
phone is displaying the word Störung (english probably out of 
order) within days of using it requiring modules to be 
unloaded/loaded and asterisk restarted.



Are you sure the problem isn't on the Siemens' side? I have a Siemens 
Gigaset SX303isdn connected to my asterisk (bristuff-0.2.0-RC8p) and 
the Gigaset keeps running out of TEIs after some while (since TEI:s 
lost by the terminal equipment will not be reused until asterisk is 
restarted). Scheduled asterisk restarts every week seems to solve the 
problem. With other ISDN phones this problem does not occur.


The phones are working perfectly when connected directly. Even if they 
were faulty, bristuff acting as NT must be tolerant and not kill all 
buses if one client is acting strangely. Anyway I offered serveral times 
to debug this and due to my help with gdb, junghanns was able to fix 
some null pointer references (core dumps) when hanging up. At least 
Asterisk is no longer crashing completely under these circumstances but 
the actual bug was never fixed.


Frederic

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[Asterisk-Users] /dev/zap/ctl or /dev/zapctl cause ztdummy in init.d failed

2005-12-09 Thread L




Hi list,

I m wondering is there a bug regarding the zaptel/ztdummy? or it is
just my misconfiguration?
As the log shown zaptel is refering to the /dev/zap/ctl when it suppose
to refer to /dev/zapctl as i m concering.

I m using
1) zaptel-1.2.1
2) kernel 2.6.12-1.1381_FC3
3) no zap's PCI hardware

On the second try I started the modprobe manually and the system shown
the statement ("line 0: Unable to open master device '/dev/zap/ctl'").
But when i crosscheck using lsmod, ztdummy is actually loaded into the
kernel.
And this line somehow cause init.d/zaptel failed to load the later half
of the script.

[EMAIL PROTECTED] ~]# modprobe zaptel
[EMAIL PROTECTED] ~]# lsmod | grep z
Module Size Used by
zaptel 208132 0
crc_ccitt 2113 1 zaptel
dm_zero 2113 0
dm_mod 57333 6 dm_snapshot,dm_zero,dm_mirror
[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

FATAL: Error running install command for ztdummy
[EMAIL PROTECTED] ~]# lsmod | grep z
Module Size Used by
ztdummy 3924 0
zaptel 208132 1 ztdummy
crc_ccitt 2113 1 zaptel
dm_zero 2113 0
dm_mod 57333 6 dm_snapshot,dm_zero,dm_mirror

Any idea?

-L-

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[Asterisk-Users] Hangup after dialing

2005-12-09 Thread René Enskat [Teamware GmbH]



i updated to actual
sVN but now when i call with my phone i get a hangup when the clal should be
ringing.
with the branch all
is fine.

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[Asterisk-Users] Queue routing - calls return to agent which previously handled call

2005-12-09 Thread Hilton Williams

Hi

Is there a way to get incoming calls to go to the same agent that handled 
them previously, based on the Caller ID?  This would be great for support / 
helpdesk, since the caller doesn't have to explain the whole problem to each 
agent.


Does anyone know?

We're using [EMAIL PROTECTED] 1.5, with Asterisk 1.0.9, but I'd be interested to 
know even if it's in a more recent version of Asterisk.


Regards
Hilton Williams



Datatex Dynamics CC
Web site http://www.datatex.co.za/
Email to [EMAIL PROTECTED]
Tel +27215924033
Fax +27215924077

The use of the Datatex e-mail facility is not permitted
for the distribution of chain letters or offensive email
of any nature whatsoever. Datatex hereby distances itself
from and accepts no liability in respect of the
unauthorised use of its e-mail facility or the sending of
e-mail communications for other than strictly business
purposes. Datatex furthermore disclaims liability for any
unauthorised instruction for which permission was not
granted. Any recipient of an unacceptable communication,
a chain letter or offensive material of any nature is
requested to report it to [EMAIL PROTECTED] 



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[Asterisk-Users] CIDNUM CIDNAME

2005-12-09 Thread René Enskat [Teamware GmbH]



Does the CIDNUM and
CIDNAME is not any longer working?
How do i get the
parts from the CALLERID?

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[Asterisk-Users] Re: /dev/zap/ctl or /dev/zapctl cause ztdummy in init.d failed

2005-12-09 Thread Tony Mountifield
In article [EMAIL PROTECTED], L [EMAIL PROTECTED] wrote:
 -=-=-=-=-=-
 Hi list,
 
 I m wondering is there a bug regarding the zaptel/ztdummy? or it is just my 
 misconfiguration?
 As the log shown zaptel is refering to the /dev/zap/ctl when it suppose to 
 refer to
 /dev/zapctl as i m concering.
 
 I m using
 1) zaptel-1.2.1
 2) kernel 2.6.12-1.1381_FC3
 3) no zap's PCI hardware
 
 On the second try I started the modprobe manually and the system shown the 
 statement (line
 0: Unable to open master device '/dev/zap/ctl').

Have you followed the instructions in zaptel/README.udev?

Cheers
Tony
-- 
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Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Re: /dev/zap/ctl or /dev/zapctl cause ztdummy in init.d failed - SOLVED

2005-12-09 Thread L




Hi Tony,

I have miss the README.udev part and this is user mistake.
Thanx for pointing me to the right direction.

Regards
-L-

Tony Mountifield wrote:

  In article [EMAIL PROTECTED], L [EMAIL PROTECTED] wrote:
  
  
-=-=-=-=-=-
Hi list,

I m wondering is there a bug regarding the zaptel/ztdummy? or it is just my misconfiguration?
As the log shown zaptel is refering to the /dev/zap/ctl when it suppose to refer to
/dev/zapctl as i m concering.

I m using
1) zaptel-1.2.1
2) kernel 2.6.12-1.1381_FC3
3) no zap's PCI hardware

On the second try I started the modprobe manually and the system shown the statement ("line
0: Unable to open master device '/dev/zap/ctl'").

  
  
Have you followed the instructions in zaptel/README.udev?

Cheers
Tony
  



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Re: [Asterisk-Users] /dev/zap/ctl or /dev/zapctl cause ztdummy in init.d failed

2005-12-09 Thread Steve Ringwald
Are you running udev? If so, you need to follow the directions in 
README.udev...


http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3


   Help!: line 146: Unable to open master device '/dev/zap/ctl'

You are probably running udev and don't know it.. were you paying 
attention during the make? If you were, you would have seen this fly by:

 Dynamic filesystem detected -- not creating device nodes
 If you are running udev, read README.udev

If you didn't, try doing a:

[EMAIL PROTECTED] zaptel]# make devices


In your zaptel directory and see if it comes up.. If it does, view the 
suggested README.udev.


*Run udevstart to re-read the new configuration and to create the zap 
nodes.*


*Note:* My Fedora Core 3 install which came with a 2.6.9 kernel did just 
this. If you follow the directions in README.udev, you will fix this.




L wrote:

Hi list,

I m wondering is there a bug regarding the zaptel/ztdummy? or it is 
just my misconfiguration?
As the log shown zaptel is refering to the /dev/zap/ctl when it 
suppose to refer to /dev/zapctl as i m concering.


I m using
1) zaptel-1.2.1
2) kernel 2.6.12-1.1381_FC3
3) no zap's PCI hardware

On the second try I started the modprobe manually and the system shown 
the statement (line 0: Unable to open master device '/dev/zap/ctl').
But when i crosscheck using lsmod, ztdummy is actually loaded into the 
kernel.
And this line somehow cause init.d/zaptel failed to load the later 
half of the script.



[EMAIL PROTECTED] ~]#* modprobe zaptel
[EMAIL PROTECTED] ~]#* lsmod | grep z
Module  Size  Used by
zaptel208132  0
crc_ccitt   2113  1 zaptel
dm_zero 2113  0
dm_mod 57333  6 dm_snapshot,dm_zero,dm_mirror
[EMAIL PROTECTED] ~]#* modprobe ztdummy
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

FATAL: Error running install command for ztdummy
[EMAIL PROTECTED] ~]#* lsmod | grep z
Module  Size  Used by
ztdummy 3924  0
zaptel208132  1 ztdummy
crc_ccitt   2113  1 zaptel
dm_zero 2113  0
dm_mod 57333  6 dm_snapshot,dm_zero,dm_mirror

 
Any idea?


-L-
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Re: [Asterisk-Users] Asterisk Dial Failover

2005-12-09 Thread Philipp von Klitzing
Hi!

 I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
 lookups are broken. If I issue a series of Dial commands, such as
 this: 
  
 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
 exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)

How about you use ChanIsAvail() before each dial statement? Or try to 
Dial with 1 sec timeout or no timeout? Also check ${DIALSTATUS}.

Cheers, Philipp


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[Asterisk-Users] connection between asterisk and cisco

2005-12-09 Thread muhammad usman
HI!

how are you people. i am a newbie in asterisk and
voip.
i need your help.

the scenerio is like this.

1.all local SIP users will be connected to asterisk
via IP.

2.PSTN will be connected to AS5300.pstn will give us a
local prefix like 333. so any one calling at
333 will go to my as5300.

3.now i want if someone calls via PSTN to a number
333 this should go to my some sip user e.g john
(connect to asterisk via ip). but only to john.

4.now when john dials to any number outside 333 range
, it should be dialed to the destination via
AS5300(which is connected to PSTN). and destination
should see that it is called by a number 333.

5.now if all this scenerio is possible, how the
asterisk server and As5300 will talk to each other.
what protocol can be used between them.
and what physical connection i.e like ethernet or E-1
connection between as5300 and asterisk server.


6.which billing radius server you recommend, and what
kind of cards will be required in a5300.


thanks a lot for reading this.
and thanks for reply in advance.


any other suggestions are also welcome.

regards







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RE: [Asterisk-Users] Asterisk on PPC chan_capi issue

2005-12-09 Thread Armin Schindler
On Fri, 9 Dec 2005, David Waugh wrote:
 Sorry Patrick, I was mistaken here. The Diva Server for Linux drives 
 currently only support Little Endian machines. Unfortunately the PPC based 
 chipsets use Big Endian.

The melware.net drivers (part of kernels 2.6) do support Big-Endian.

Armin

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Patrick
 Sent: 08 December 2005 21:18
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Asterisk on PPC  chan_capi issue
 
 
 On Thu, 2005-12-08 at 08:47 +, David Waugh wrote:
  Hello Patrick,
  
  I have an Eicon Diva PRI-30M card and use the Eicon Linux drivers with 
  chan_capi_cm.
  I am able to do ISDN to SIP calls with this.
  
  Have you tried using the Eicon drivers instead, rather than zaptel and zib 
  pri.
  
  Instruction for doing this can be found here:
  http://www.voip-info.org/wiki/view/Asterisk+Eicon+Diva+CAPI+ISDN
  
  I hope this help
 
 Unfortunately not. The config I described below works fine on an Intel
 server. The box showing the issue is a PPC box. The author of
 chan_capi-cm doesn't think there is anything wrong with chan_capi-cm on
 PPC and the way it behaves in the logfiles below. So the problem seems
 to be Asterisk  PPC related but I have no idea how to track down the
 issue or solve it.
 
 Regards,
 Patrick
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] Behalf Of Patrick
  Sent: 06 December 2005 01:40
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] Asterisk on PPC  chan_capi issue
  
  
  Hi all,
  
  I have a PPC box (IBM RS6000 43P-150, bigendian afaik) which runs Fedora
  Core 5 Test1 and zaptel, libpri and asterisk 1.2.0. I also installed
  chan_capi (0.6.1) so I can use my Eicon Diva Server BRI card. Asterisk
  was compiled with DEBUG=-g and DEBUG_THREADS = -DDUMP_SCHEDULER
  -DDEBUG_SCHEDULER-DDEBUG_THREADS -DDO_CRASH -DDETECT_DEADLOCKS. Next I
  did make clean, make valgrind, make install. Asterisk runs as user/group
  asterisk/asterisk.
  
  SIP -- SIP calls are fine, Calls from SIP out to the PSTN via
  CAPI/ISDN are fine too. ISDN/CAPI -- SIP calls don't work. Example
  output of the issue is below. Anyone have an idea how I fix this?
  
  Thanks and regards,
  Patrick
  
  
  chan_capi registers fine:
  **
   [chan_capi.so] = (Common ISDN API for Asterisk)
== This box has 1 capi controller(s).
== Reading config for BRI1
  -- ast_capi_pvt BRI1-pseudo-D (MSN1,MSN2,capi-in,0,2) (1,4,128)
  -- ast_capi_pvt BRI1 (MSN1,MSN2,capi-in,0,2) (1,4,128)
  -- ast_capi_pvt BRI1 (MSN1,MSN2,capi-in,0,2) (1,4,128)
  -- listening on contr1 CIPmask = 0x1fff03ff
== Registered channel type 'CAPI' (Common ISDN API Driver ($Revision:
  1.115 $) )
== Registered application 'capiCommand'
== Registered custom function VANITYNUMBER
  
  Call from my GSM to a SIP phone (exten 1003) via ISDN/CAPI (MSN2):
  **
== BRI1: Incoming call 'my GSM' - 'MSN2'
  
  -- Executing Macro(CAPI/BRI1/MSN2-0, stdexten|1003|SIP/1003)
  in new stack
  -- Executing Dial(CAPI/BRI1/MSN2-0, SIP/1003|10|TtwW) in new
  stack
  Dec  6 02:30:47 WARNING[28889]: channel.c:2494 ast_request: No
  translator path exists for channel type SIP (native 65535) to 0
  Dec  6 02:30:47 NOTICE[28889]: app_dial.c:1011 dial_exec_full: Unable to
  create channel of type 'SIP' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
  -- Executing Goto(CAPI/BRI1/MSN2-0, s-CHANUNAVAIL|1) in new
  stack
  -- Goto (macro-stdexten,s-CHANUNAVAIL,1)
  -- Executing Goto(CAPI/BRI1/MSN2-0, s-NOANSWER|1) in new stack
  -- Goto (macro-stdexten,s-NOANSWER,1)
  -- Executing Answer(CAPI/BRI1/MSN2-0, ) in new stack
== BRI1: Answering for 703241494
  -- Executing Wait(CAPI/BRI1/MSN2-0, 1) in new stack
  Dec  6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping
  incompatible voice frame on CAPI/BRI1/MSN2-0 of format alaw since our
  native format has changed to unknown
  Dec  6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping
  incompatible voice frame on CAPI/BRI1/MSN2-0 of format alaw since our
  native format has changed to unknown
  
  [snipped tons more of these]
  
  Dec  6 02:30:48 NOTICE[28889]: channel.c:1893 ast_read: Dropping
  incompatible voice frame on CAPI/BRI1/MSN2-0 of format alaw since our
  native format has changed to unknown
  -- Executing VoiceMail(CAPI/BRI1/MSN2, u1003) in new stack
  Dec  6 02:30:48 WARNING[28889]: channel.c:2313 set_format: Unable to
  find a codec translation path from unknown to gsm
  Dec  6 02:30:48 WARNING[28889]: file.c:820 ast_streamfile: Unable to
  open vm-theperson (format unknown): No such file or directory
== BRI1: CAPI Hangingup
  CAPI INFO 0x3490: Normal call 

Re: [Asterisk-Users] Asterisk Dial Failover

2005-12-09 Thread burke
Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes. One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for sub-second
failover. I think there is a article on voip-info.org about this, but
don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan

 All,

 I have an Asterisk system that sends PSTN calls to an OpenSER system to be
 routed. I have a command like this in my extensions.conf:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)

 There's actually two OpenSER systems for redundancy. I'm trying to find a
 way to have Asterisk attempt to route the call to one OpenSER system, and
 if it's down, fallback to another.

 Any first thoughts on how to achieve this?

 I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups
 are broken. If I issue a series of Dial commands, such as this:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
 exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)

 ... what seems to happen is that when proxy1 is down, Asterisk waits the
 full 20 seconds before returning control. Also, This 20s includes the time
 is takes for the other end to answer, so if I put a small value of say 5s
 in there, the dial command will probably give up before someone answers at
 the other end. Neither is workable.

 Asterisk SHOULD be able to distinguish between a TRYING and no response.
 In the event it gets no TRYING response to a dial command within a
 specified timeout it should return control and flag an error. If on the
 other hand it does get a TRYING response (and maybe a RINGING too) it
 should continue to wait until the 20s has expired.

 I can't use dynamic DNS (ie putting two A records for a hostname in DNS)
 because Asterisk reads the extensions.conf on startup and also seems to
 cache what the host maps to on startup. Subsequent calls to the host
 always return the same IP address.

 But... in general... how have people implemented this?

 Help appreciated!
 Doug







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Re: [Asterisk-Users] Porting a phone number to a voip provider

2005-12-09 Thread Robert Webb


On Fri, 09 Dec 2005 00:36:18 -0500
 Matthew matthew@zeut.net wrote:
Hello, has anyone taken their cell phone number and 
ported it over to a voip provider?  If so, what voip 
provider and what was your experience? 
Matt




Matt,

  I have done this. I had a cell number with ATT 
Wireless and first ported it to Broadvox Direct. There 
service was ok but ended up not fitting my needs as trying 
to run their Mediatrix box into my Asterisk box was just 
not working too well.


  I have since ported it away from Broadvox Direct to a 
Voicepulse Connect Account. I am running it straight into 
Asterisk now over an IAX connection and it has been 
working fine for me. Not a lot of calls come in on it, so 
I cannot really tell you what the real up time is on it. I 
just know that no one has ever told me that they tried on 
that number and could not get me.


Robert
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[Asterisk-Users] Re: ztdummy on FC4

2005-12-09 Thread Jim Duda
 Check ztdummy.h and ztdummy.c and make doubly sure USE_RTC is defined.
 It sounds like you are using a version compiled without USE_RTC on one
 of the newer kernels that has fewer than 1000 jiffies per second.


I'm using zaptel from CVS (cvs.digium.com).  I just did an update (many 
files
out-of-date), however, USE_RTC is NOT to be found in any source file.  I 
could
add it, however, doesn't appear to be used anywhere.

Jim

Tony Mountifield [EMAIL PROTECTED] wrote in message 
news:[EMAIL PROTECTED]
 In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] 
 wrote:
 I'm running asterisk on FC4.  All works fine, including musiconhold.
 I tried installing ztdummy as directed, since the documentation
 indicates that ztdummy is required for good music quality.

 However, installing ztdummy on FC4 causes moh to play very slow.
 If I remove ztdummy, all works okay again.

 Check ztdummy.h and ztdummy.c and make doubly sure USE_RTC is defined.
 It sounds like you are using a version compiled without USE_RTC on one
 of the newer kernels that has fewer than 1000 jiffies per second.

 I used to use ztdummy before FC4, on a 2.4 kernel, but now I cannot.

 Do I really need it?

 If you do not have any zaptel cards in the system,then ztdummy is
 required for anything that requires local timing. That includes
 MeetMe, IAX2 trunking and possibly a few other things.

 Will I need it when I implement conference calling?

 Yes, if you use MeetMe to do it.

 Anyone have a similiar FC4 experience?

 Sorry, only used FC1 and FC3.

 Cheers
 Tony
 -- 
 Tony Mountifield
 Work: [EMAIL PROTECTED] - http://www.softins.co.uk
 Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Queue routing - calls return to agent which previously handled call

2005-12-09 Thread Lenz

Hi,
I don't think it is impossible, though not yet supported by Asterisk  
out-of-the-box. You could have a general queue plus a queue per each  
agent, and you would route the call to each agent based on the caller*id.  
This might end up spoiling the advantage of a queue, meaning that you  
might have three agents sitting idle and a fourth with three calls queued.  
It would be better to have different agents automatically open a case  
ticket based on the caller*id, so each agent sees the problem.

Yours
l.


On Fri, 09 Dec 2005 11:40:03 +0100, Hilton Williams [EMAIL PROTECTED]  
wrote:



Hi

Is there a way to get incoming calls to go to the same agent that  
handled them previously, based on the Caller ID?  This would be great  
for support / helpdesk, since the caller doesn't have to explain the  
whole problem to each agent.


Does anyone know?

We're using [EMAIL PROTECTED] 1.5, with Asterisk 1.0.9, but I'd be  
interested to know even if it's in a more recent version of Asterisk.


Regards
Hilton Williams



Datatex Dynamics CC
Web site http://www.datatex.co.za/
Email to [EMAIL PROTECTED]
Tel +27215924033
Fax +27215924077

The use of the Datatex e-mail facility is not permitted
for the distribution of chain letters or offensive email
of any nature whatsoever. Datatex hereby distances itself
from and accepts no liability in respect of the
unauthorised use of its e-mail facility or the sending of
e-mail communications for other than strictly business
purposes. Datatex furthermore disclaims liability for any
unauthorised instruction for which permission was not
granted. Any recipient of an unacceptable communication,
a chain letter or offensive material of any nature is
requested to report it to [EMAIL PROTECTED]




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Re: [Asterisk-Users] Porting a phone number to a voip provider

2005-12-09 Thread Rich Adamson
  Hello, has anyone taken their cell phone number and 
 ported it over to a voip provider?  If so, what voip 
 provider and what was your experience? 
  Matt
  
 
 Matt,
 
I have done this. I had a cell number with ATT 
 Wireless and first ported it to Broadvox Direct. There 
 service was ok but ended up not fitting my needs as trying 
 to run their Mediatrix box into my Asterisk box was just 
 not working too well.
 
I have since ported it away from Broadvox Direct to a 
 Voicepulse Connect Account. I am running it straight into 
 Asterisk now over an IAX connection and it has been 
 working fine for me. Not a lot of calls come in on it, so 
 I cannot really tell you what the real up time is on it. I 
 just know that no one has ever told me that they tried on 
 that number and could not get me.

Just an FYI... not all cell numbers are portable.


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RE: [Asterisk-Users] Asterisk Dial Failover

2005-12-09 Thread Jonathan k. Creasy
I chose this method and have been happy with the results. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second
failover. I think there is a article on voip-info.org about this, but
don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan

 All,

 I have an Asterisk system that sends PSTN calls to an OpenSER system
to be
 routed. I have a command like this in my extensions.conf:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)

 There's actually two OpenSER systems for redundancy. I'm trying to
find a
 way to have Asterisk attempt to route the call to one OpenSER system,
and
 if it's down, fallback to another.

 Any first thoughts on how to achieve this?

 I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups
 are broken. If I issue a series of Dial commands, such as this:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
 exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)

 ... what seems to happen is that when proxy1 is down, Asterisk waits
the
 full 20 seconds before returning control. Also, This 20s includes the
time
 is takes for the other end to answer, so if I put a small value of say
5s
 in there, the dial command will probably give up before someone
answers at
 the other end. Neither is workable.

 Asterisk SHOULD be able to distinguish between a TRYING and no
response.
 In the event it gets no TRYING response to a dial command within a
 specified timeout it should return control and flag an error. If on
the
 other hand it does get a TRYING response (and maybe a RINGING too) it
 should continue to wait until the 20s has expired.

 I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)
 because Asterisk reads the extensions.conf on startup and also seems
to
 cache what the host maps to on startup. Subsequent calls to the host
 always return the same IP address.

 But... in general... how have people implemented this?

 Help appreciated!
 Doug







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[Asterisk-Users] Change time when * is running

2005-12-09 Thread Julian Lyndon-Smith
We've just seen that one of our servers is an hour out (it reckons that 
it's 15:02 instead of 14:02).


Can I change the time when * is running ? I don't want to try just in 
case it causes * some grief.


Julian.

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Re: [Asterisk-Users] Porting a phone number to a voip provider

2005-12-09 Thread Matthew

Rich Adamson wrote:

Just an FYI... not all cell numbers are portable.


Do you have any more information on this?  I read somewhere that 
sometimes you can port a number to a VoIP provider but not be able to 
port it back to the PSTN because not all PSTN providers will take 
numbers from VoIP providers.  Is this what you are talking about?


Matt

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Re: [Asterisk-Users] Queue routing - calls return to agent which previously handled call

2005-12-09 Thread Rob Lith
An elegant wat to do this would be to have the caller ID and agent the call was sent to stored in mysql (Asterisk can do systems calls to this) and when calls come in do a quick check to the database, if it's matched put it through to the same agent, if the agent is busy it can revert to the general queue - it there is no match then put it straight in the general queue. We could look at doing this for/with you. Email me - 
[EMAIL PROTECTED]RobOn 12/9/05, Lenz [EMAIL PROTECTED]
 wrote:Hi,I don't think it is impossible, though not yet supported by Asterisk
out-of-the-box. You could have a general queue plus a queue per eachagent, and you would route the call to each agent based on the caller*id.This might end up spoiling the advantage of a queue, meaning that you
might have three agents sitting idle and a fourth with three calls queued.It would be better to have different agents automatically open a caseticket based on the caller*id, so each agent sees the problem.
Yoursl.On Fri, 09 Dec 2005 11:40:03 +0100, Hilton Williams [EMAIL PROTECTED]wrote: Hi Is there a way to get incoming calls to go to the same agent that
 handled them previously, based on the Caller ID?This would be great for support / helpdesk, since the caller doesn't have to explain the whole problem to each agent. Does anyone know?
 We're using [EMAIL PROTECTED] 1.5, with Asterisk 1.0.9, but I'd be interested to know even if it's in a more recent version of Asterisk. Regards Hilton Williams
 Datatex Dynamics CC Web site http://www.datatex.co.za/ Email to [EMAIL PROTECTED] Tel +27215924033
 Fax +27215924077 The use of the Datatex e-mail facility is not permitted for the distribution of chain letters or offensive email of any nature whatsoever. Datatex hereby distances itself
 from and accepts no liability in respect of the unauthorised use of its e-mail facility or the sending of e-mail communications for other than strictly business purposes. Datatex furthermore disclaims liability for any
 unauthorised instruction for which permission was not granted. Any recipient of an unacceptable communication, a chain letter or offensive material of any nature is requested to report it to 
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Re: [Asterisk-Users] Polycom 501 remapping keys

2005-12-09 Thread Matthew
There has been a fair amount of converstaion about this, but I'm not 
sure anyone really has this working.  I had exactly the same problem 
that the button got remapped to a volume up function.  The only button 
remapping I got working was to map the Transfer button to the # key so 
that when you hit Transfer it started and Asterisk based transfer.


I would love to hear from someone who has this working.

Matthew O'Connor



[EMAIL PROTECTED] wrote:

I've tried to configure the services-key on my Polycom 501 to run a SpeedDial-entry in 
[MACADRESS]-directory.xml (which would call a asterisk-extension that starts SayUnixTime) but i 
have not been able to accomplish my goal. Whe configuring the SpeedDial-function in sip.cfg 
VolUp is started when i press the Services-Key.

Also some other possible functions listed under 4.6.1.15 in the SIP 1.6 Administrator Guide fail. Some of them were working with the expected function, some where not giving any response at all but some where starting totally different functions, e.g. configuring Redial as the function starts Settings, function Messages starts Redial, SpeedDialMenu starts VolUp, VolUp starts Line1 :-[ 


I've seen that other failed as well 
(http://lists.digium.com/pipermail/asterisk-users/2005-October/130129.html) - 
anyone ever got this working? Maybe with BootROM 3.0/3.1? Or should i downgrade 
to 1.5 where there was a ipmid-file for remapping-info...?

I'm running Firmware 1.6.2.0041/BootROM 2.6.2.0032

regards
Christian
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Re: [Asterisk-Users] Change time when * is running

2005-12-09 Thread Kevin P. Fleming

Julian Lyndon-Smith wrote:
We've just seen that one of our servers is an hour out (it reckons that 
it's 15:02 instead of 14:02).


Can I change the time when * is running ? I don't want to try just in 
case it causes * some grief.


It can cause some repercussions. I wouldn't recommend changing the time 
backwards by such a large amount while Asterisk is running with active 
calls... but your mileage may vary :-) (And before people jump in and 
say that this already happens when DST shifts occur... that is very 
different, as the actual clock tick counter does _NOT_ change, only the 
userspace representation of 'wall clock time' does)


Certainly it would be safer to wait until a time when there is no call 
activity and change it at that point.

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[Asterisk-Users] Low Layer Compatibility (LLC) not forwarded?

2005-12-09 Thread Lars Poschitzki
Hello *-users,

this is my first mail and here I have my first big problem for you...

I use Asterisk with the Bristuff-Patches and a Digium TE405P plus a 
quadBRI-card (european ISDN).
Everything I tried (calls with speech via BRI, SIP and POTS) was succesful 
except the current task. I try to establish a data call (V.110) originating in 
a BRI terminal going to an E1-ISDN-Router.
The E1-station checks whether an LLC-information is contained in the 
SETUP-Message and then accepts the call.

Now I can see while I debug the channel (Asterisk CLI - pri debug span #) that 
the LLC is received by the PBX but not sent to the PRI... Sad
The result is that the PRI-station rejects the call with Cause: Incompatible 
destination.


Here's a trace of the incoming (BRI) line:


asterisk1*CLI bri debug span 2
asterisk1*CLI Enabled debugging on span 2

asterisk1*CLI
Protocol Discriminator: Q.931 (8)  len=31
Call Ref: len= 1 (reference 125/0x7D) (Originator)
Message type: SETUP (5)
[04 02 88 90]
Bearer Capability (len= 4) [ Ext: 1  Q.931 Std: 0  Info transfer capability: 
Unrestricted digital information (8)
 Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
 Ext: 0  User information layer 1: Unknown (24)
[18 01 83]
Channel ID (len= 3) [ Ext: 1  IntID: Implicit, Other Spare: 0, Preferred Dchan: 0
   ChanSel: Any channel selectedNo channel selected
   ]
[70 0a 80 35 31 32 38 30 30 35 30 32]
Called Number (len=12) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown 
Number Plan (0) '512800502' ]
[7c 06 88 90 21 45 20 bb]
IE: Low-layer Compatibility (len = 8)
-- Making new call for cr 125
-- Processing Q.931 Call Setup
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 112 (cs0, Called Party Number)
-- Processing IE 124 (cs0, Low-layer Compatibility)
Protocol Discriminator: Q.931 (8)  len=7
Call Ref: len= 1 (reference 253/0xFD) (Terminator)
Message type: CALL PROCEEDING (2)
[18 01 8a]
Channel ID (len= 3) [ Ext: 1  IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0
   ChanSel: B2 channel
   ]
Protocol Discriminator: Q.931 (8)  len=8
Call Ref: len= 1 (reference 253/0xFD) (Terminator)
Message type: ALERTING (1)






The outgoing (PRI) line shows:


asterisk1*CLI pri debug span 5
asterisk1*CLI Enabled debugging on span 5
...
...
...
asterisk1*CLI
-- Accepting data call from '' to '512800502' on channel 0/2, span 2
...
...
...
-- Making new call for cr 32770
-- Requested transfer capability: 0x08 - DIGITAL
asterisk1*CLI

Protocol Discriminator: Q.931 (8)  len=27
Call Ref: len= 2 (reference 2/0x2) (Originator)
Message type: SETUP (5)
[04 02 88 90]
Bearer Capability (len= 4) [ Ext: 1  Q.931 Std: 0  Info transfer capability: 
Unrestricted digital information (8)
 Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
 Ext: 0  User information layer 1: Unknown (24)
[18 03 a1 83 81]
Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0
   ChanSel: Reserved
  Ext: 1  Coding: 0   Number Specified   Channel Type: 3
  Ext: 1  Channel: 1 ]
[6c 02 00 c3]
Calling Number (len= 4) [ Ext: 0  TON: Unknown Number Type (0)  NPI: Unknown 
Number Plan (0)
  Presentation: Number not available (67) '' ]
[70 06 c1 31 33 2d 32 37]
Called Number (len= 8) [ Ext: 1  TON: Subscriber Number (4)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '13-27' ]
[a1]
Sending Complete (len= 1)
  -- Called 13-27
asterisk1*CLI

Protocol Discriminator: Q.931 (8)  len=9
Call Ref: len= 2 (reference 32770/0x8002) (Terminator)
Message type: RELEASE COMPLETE (90)
[08 02 80 d8]
Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: User 
(0)
 Ext: 1  Cause: Incompatible destination (88), class = Invalid 
message (5) ]
 Processing IE 8 (cs0, Cause)
  -- Channel 0/1, span 5 got hangup


Do you have any ideas if Asterisk supports LLC at all or what the solution 
could be?

Thanks in advance,
Lars

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Re: [Asterisk-Users] Change time when * is running

2005-12-09 Thread Tzafrir Cohen
On Fri, Dec 09, 2005 at 02:02:53PM +, Julian Lyndon-Smith wrote:
 We've just seen that one of our servers is an hour out (it reckons that 
 it's 15:02 instead of 14:02).
 
 Can I change the time when * is running ? I don't want to try just in 
 case it causes * some grief.

keep clocks in sync with ntp . Or set the system clock with time.

However if this is exactly an hour, I suggest that you simply look at it
from an incorrect timezone.

Could you please provide the output of:

  ls -l /etc/localtime

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
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Re: [Asterisk-Users] Change time when * is running

2005-12-09 Thread Gavin Hamill

Julian Lyndon-Smith wrote:

Kevin P. Fleming wrote:

Can I change the time when * is running ? I don't want to try just in 
case it causes * some grief.



It can cause some repercussions. I wouldn't recommend changing the 
time backwards by such a large amount while Asterisk is running with 
active calls...



Julian, can you check and make sure that your system is configured to 
use the correct timezone? Usually just typing


# ls -l /etc/localtime

will be enough - to ensure that the symlink is pointing at the right 
area of the world, since that will automatically keep the clock up to 
date with daylight savings etc.


Cheers,
Gavin.

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Re: [Asterisk-Users] Low Layer Compatibility (LLC) not forwarded?

2005-12-09 Thread Klaus Darilion

Hi Lars!

I had similar problems when I tried to forward an UMTS video call 
(H.324M). You can read the problems and mabye where to fix it on

http://bugs.digium.com/view.php?id=3891

klaus

Lars Poschitzki wrote:

Hello *-users,

this is my first mail and here I have my first big problem for you...

I use Asterisk with the Bristuff-Patches and a Digium TE405P plus a 
quadBRI-card (european ISDN).
Everything I tried (calls with speech via BRI, SIP and POTS) was succesful 
except the current task. I try to establish a data call (V.110) originating in 
a BRI terminal going to an E1-ISDN-Router.
The E1-station checks whether an LLC-information is contained in the 
SETUP-Message and then accepts the call.

Now I can see while I debug the channel (Asterisk CLI - pri debug span #) that 
the LLC is received by the PBX but not sent to the PRI... Sad
The result is that the PRI-station rejects the call with Cause: Incompatible 
destination.


Here's a trace of the incoming (BRI) line:


asterisk1*CLI bri debug span 2
asterisk1*CLI Enabled debugging on span 2

asterisk1*CLI
Protocol Discriminator: Q.931 (8)  len=31
Call Ref: len= 1 (reference 125/0x7D) (Originator)
Message type: SETUP (5)
[04 02 88 90]
Bearer Capability (len= 4) [ Ext: 1  Q.931 Std: 0  Info transfer capability: 
Unrestricted digital information (8)
 Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
 Ext: 0  User information layer 1: Unknown (24)
[18 01 83]
Channel ID (len= 3) [ Ext: 1  IntID: Implicit, Other Spare: 0, Preferred Dchan: 0
   ChanSel: Any channel selectedNo channel selected
   ]
[70 0a 80 35 31 32 38 30 30 35 30 32]
Called Number (len=12) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown 
Number Plan (0) '512800502' ]
[7c 06 88 90 21 45 20 bb]
IE: Low-layer Compatibility (len = 8)
-- Making new call for cr 125
-- Processing Q.931 Call Setup
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 112 (cs0, Called Party Number)
-- Processing IE 124 (cs0, Low-layer Compatibility)
Protocol Discriminator: Q.931 (8)  len=7
Call Ref: len= 1 (reference 253/0xFD) (Terminator)
Message type: CALL PROCEEDING (2)
[18 01 8a]
Channel ID (len= 3) [ Ext: 1  IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0
   ChanSel: B2 channel
   ]
Protocol Discriminator: Q.931 (8)  len=8
Call Ref: len= 1 (reference 253/0xFD) (Terminator)
Message type: ALERTING (1)






The outgoing (PRI) line shows:


asterisk1*CLI pri debug span 5
asterisk1*CLI Enabled debugging on span 5
...
...
...
asterisk1*CLI
-- Accepting data call from '' to '512800502' on channel 0/2, span 2
...
...
...
-- Making new call for cr 32770
-- Requested transfer capability: 0x08 - DIGITAL
asterisk1*CLI

Protocol Discriminator: Q.931 (8)  len=27
Call Ref: len= 2 (reference 2/0x2) (Originator)
Message type: SETUP (5)
[04 02 88 90]
Bearer Capability (len= 4) [ Ext: 1  Q.931 Std: 0  Info transfer capability: 
Unrestricted digital information (8)
 Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
 Ext: 0  User information layer 1: Unknown (24)
[18 03 a1 83 81]
Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0
   ChanSel: Reserved
  Ext: 1  Coding: 0   Number Specified   Channel Type: 3
  Ext: 1  Channel: 1 ]
[6c 02 00 c3]
Calling Number (len= 4) [ Ext: 0  TON: Unknown Number Type (0)  NPI: Unknown 
Number Plan (0)
  Presentation: Number not available (67) '' ]
[70 06 c1 31 33 2d 32 37]
Called Number (len= 8) [ Ext: 1  TON: Subscriber Number (4)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '13-27' ]
[a1]
Sending Complete (len= 1)
  -- Called 13-27
asterisk1*CLI

Protocol Discriminator: Q.931 (8)  len=9
Call Ref: len= 2 (reference 32770/0x8002) (Terminator)
Message type: RELEASE COMPLETE (90)
[08 02 80 d8]
Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: User 
(0)
 Ext: 1  Cause: Incompatible destination (88), class = Invalid 
message (5) ]
 Processing IE 8 (cs0, Cause)
  -- Channel 0/1, span 5 got hangup


Do you have any ideas if Asterisk supports LLC at all or what the solution 
could be?

Thanks in advance,
Lars



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Re: [Asterisk-Users] Change time when * is running

2005-12-09 Thread Julian Lyndon-Smith

Hmm,

ls -l /etc/localtime
-rw-r--r--  1 root root 1323 Nov 25 12:43 /etc/localtime

there's no symlink that I can see. This is CentOS 4.2

Julian


Gavin Hamill wrote:


Julian Lyndon-Smith wrote:

Kevin P. Fleming wrote:

Can I change the time when * is running ? I don't want to try just 
in case it causes * some grief.




It can cause some repercussions. I wouldn't recommend changing the 
time backwards by such a large amount while Asterisk is running with 
active calls...




Julian, can you check and make sure that your system is configured to 
use the correct timezone? Usually just typing


# ls -l /etc/localtime

will be enough - to ensure that the symlink is pointing at the right 
area of the world, since that will automatically keep the clock up to 
date with daylight savings etc.


Cheers,
Gavin.

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Re: [Asterisk-Users] Change time when * is running

2005-12-09 Thread Julian Lyndon-Smith

Yeah,

I was going to change it tonight :) I'll wait until then.

Julian.

Kevin P. Fleming wrote:


Julian Lyndon-Smith wrote:

We've just seen that one of our servers is an hour out (it reckons 
that it's 15:02 instead of 14:02).


Can I change the time when * is running ? I don't want to try just in 
case it causes * some grief.



It can cause some repercussions. I wouldn't recommend changing the 
time backwards by such a large amount while Asterisk is running with 
active calls... but your mileage may vary :-) (And before people jump 
in and say that this already happens when DST shifts occur... that is 
very different, as the actual clock tick counter does _NOT_ change, 
only the userspace representation of 'wall clock time' does)


Certainly it would be safer to wait until a time when there is no call 
activity and change it at that point.

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Re: [Asterisk-Users] Change time when * is running

2005-12-09 Thread Gavin Hamill

Julian Lyndon-Smith wrote:


Hmm,

ls -l /etc/localtime
-rw-r--r--  1 root root 1323 Nov 25 12:43 /etc/localtime

there's no symlink that I can see. This is CentOS 4.2


OK, I just had a look in /usr/share/zoneinfo and the only files which 
were 1323 bytes were for UK + Ireland, so if that's where you are, then 
you're all set - looks like you will indeed need a clock jump - best 
schedule that in a quiet time, or just do it and blame BT :)


Cheers,
Gavin.

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[Asterisk-Users] Re: ztdummy on FC4

2005-12-09 Thread Tony Mountifield
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote:
  Check ztdummy.h and ztdummy.c and make doubly sure USE_RTC is defined.
  It sounds like you are using a version compiled without USE_RTC on one
  of the newer kernels that has fewer than 1000 jiffies per second.
 
 
 I'm using zaptel from CVS (cvs.digium.com).  I just did an update (many
 files out-of-date), however, USE_RTC is NOT to be found in any source
 file.  I could add it, however, doesn't appear to be used anywhere.

That sound like you might have a sticky tag keeping you on an older
branch. Could you post the output of cvs status ztdummy.c ? Thanks.

Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Why Won't Asterisk REINVITE?

2005-12-09 Thread Julian J. M.
Try removing the Answer() before the Dial... e.g.:

[spa2100]

exten = _X.,1,NoOp(SIP Call from SPA2100 to ${EXTEN})
exten = _X.,2,Dial(SIP/netvoice-102)
exten = _X.,3,Hangup

Regards
   Julian J. M.


On 12/9/05, George Pajari [EMAIL PROTECTED] wrote:
 Eric ManxPower Wieling wrote:

  T/t/H/h and other options to Dial require Asterisk to stay in the RTP
  stream.

 Understood but already checked as not being the cause. Thanks for the
 suggestion, though.

 Here is our entire extensions.conf context:

 [spa2100]

 exten = _X.,1,NoOp(SIP Call from SPA2100 to ${EXTEN})
 exten = _X.,2,Answer
 exten = _X.,3,Wait(2)
 exten = _X.,4,Dial(SIP/netvoice-102)
 exten = _X.,5,Hangup

 where

 [netvoice-102]
 accountcode=netvoice-102
 callerid=NETVOICE COMMS 604 484 8647
 username=netvoice-102
 type=friend
 host=dynamic
 dtmfmode=rfc2833
 nat=no
 qualify=no
 mailbox=102
 context = netvoice-internal
 canreinvite=yes
 disallow=all
 allow=ulaw

 Here is a sip show channels during a call:

 aa.bb.cc.39netvoice-1  7f6a484c36f  00103/0   ulaw
 aa.bb.cc.40nvc.test.a  6cfe5077-2f  00103/00102   ulaw

 --
 George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
 Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
   www.netvoice.ca  www.ip-centrex.ca
   www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca

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RE: [Asterisk-Users] Asterisk Dial Failover

2005-12-09 Thread Adam Robins
What are you using to terminate the PSTN calls and do the SIP
transcoding? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second failover. I think there is a article on voip-info.org about
this, but don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan

 All,

 I have an Asterisk system that sends PSTN calls to an OpenSER system
to be
 routed. I have a command like this in my extensions.conf:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)

 There's actually two OpenSER systems for redundancy. I'm trying to
find a
 way to have Asterisk attempt to route the call to one OpenSER system,
and
 if it's down, fallback to another.

 Any first thoughts on how to achieve this?

 I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups
 are broken. If I issue a series of Dial commands, such as this:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
 exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)

 ... what seems to happen is that when proxy1 is down, Asterisk waits
the
 full 20 seconds before returning control. Also, This 20s includes the
time
 is takes for the other end to answer, so if I put a small value of say
5s
 in there, the dial command will probably give up before someone
answers at
 the other end. Neither is workable.

 Asterisk SHOULD be able to distinguish between a TRYING and no
response.
 In the event it gets no TRYING response to a dial command within a 
 specified timeout it should return control and flag an error. If on
the
 other hand it does get a TRYING response (and maybe a RINGING too) it 
 should continue to wait until the 20s has expired.

 I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)
 because Asterisk reads the extensions.conf on startup and also seems
to
 cache what the host maps to on startup. Subsequent calls to the host 
 always return the same IP address.

 But... in general... how have people implemented this?

 Help appreciated!
 Doug







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This transmission is sent in trust, for the sole purpose of delivery to the 
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reproduction or dissemination of this transmission is strictly prohibited. If 
you are not the intended recipient, please immediately notify the sender by 
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[Asterisk-Users] Phone Information

2005-12-09 Thread James Horn
On the CM, there is away to get the Device Information, Network Configuration, etc. by httping to the phones IP address. Is there away to do this via Astericks?
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Re: [Asterisk-Users] New GSM 1-8 ports Gateway / Terminal for sale (with SMS Feature and Many more)

2005-12-09 Thread Rusty Dekema
If it's an electronic device, which this certainly is, and if it works on 100-240V, it will almost certainly work at either 50 or 60Hz. It probably gets converted to DC anyway but even if not, there wouldn't be much point in manufacturing a 100-240V power supply if it wouldn't work on both 50 and 60Hz. 
-RustyOn 12/8/05, Brian Fertig 
[EMAIL PROTECTED] wrote:
Well up until I saw the 100-220V and 50HZ I was sold..But if you dontsupport 60HZ it will never work in North America.Well it could but itwould be a pain in the ass../bOn Thu, 2005-12-08 at 23:41 +0800, Sam Tam wrote:
 The long waited Ultimate GSM Gateway is finally out. This time we have managed to source a new patch of brand NEW GSM Gateway at prices that is only 50% of what the market rate. And with the SMS Function and many more...
 For purchase please email gsm AT cyper-telecom.net. We accept paypal and bank transfer.
 Postage is not included. Please notice we have also got the standard Dual Band GSM gateway for £60 per unit.
 Introduction: Cyber-Telecom Fixed Wireless GSM Gateway Devices integrated GSM and CDMA technologies. Features: Security features including terminal lock, SIM lock, Carrier Lock and District Lock
 Supports least cost routing based phone # dialled Network Management through SMS: can configure FWT parameters and query FWT parameters Parameter management: parameters can be modified either by phone or via NMS software
 Billing signal support by providing reverse polarity signals Models: GSM-TRI-SMS-01 RJ11 interface 1 GSM/CDMA interface £99 per unit GSM-TRI-SMS-04
 4 RJ11 interfaces 4 GSM interfaces £399 per unit GSM-TRI-SMS-08 8 RJ11 interfaces 8 GSM interfaces £799 per unit Technical Specifications:

 Working spectrumGSM900/GSM1800/GSM1900MHz CDMA800/CDMA1900MHz Power AC 110-220V/50Hz Temperature -20 ℃ - 40℃ Humidity10%-95% ___
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http://lists.digium.com/mailman/listinfo/asterisk-users--_.._Brian FertigData/Telecom EngineerIT AdministratorPlanet Telecom, IncTampa, FL Office
o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164SIP URI:[EMAIL PROTECTED]
This email was scanned by:Mcafee GroupShield CONFIDENTIAL DISCLAMER 
All information provided in this email is considered confidentialand proprietary of Planet Telecom, Inc. and Telecenter Inc.Use of this information by anyone other than the recipient orsender will be considered in breach of agreement.
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Re: [Asterisk-Users] Porting a phone number to a voip provider

2005-12-09 Thread Rich Adamson
  Just an FYI... not all cell numbers are portable.
 
 Do you have any more information on this?  I read somewhere that 
 sometimes you can port a number to a VoIP provider but not be able to 
 port it back to the PSTN because not all PSTN providers will take 
 numbers from VoIP providers.  Is this what you are talking about?

No. Some cell providers do not have the capability to selectively let
a telephone number move to another carrier/telco/itsp. They may have
technical plans to allow/support it, but the hardware/software necessary
to allow/support is not yet in place. Same is true with some smaller
telcos.

I'd also have to guess that come cell providers have probably taken a
stand that says they aren't going to do it, regardless of what the 
regulatory folks do/say.

So, the issue with number portability really starts with the company that
currently provides the cell number to you. (Sometimes they don't like
to volunter that info for obvious reasons.) If your number is portable,
then it should remain portable regardless of where you move it to, and
you should be able to move it as many times as you'd like.

If you move your number to the xyz itsp and don't like their service,
then pick another company and ask them to move the number for you. For
them to move it, however, requires an acknowledgement from the previous
company in most cases. In some (rather rare) cases, the previous company
will sometimes drag their feet or refuse to acknowledge the transfer. There
are regulatory escalation processes to address that problem, but usually
it takes a considerable amount of time to get it done.

In some cases, even the larger cell providers play games as they don't want
to lose their numbers, customers, etc. In other cases, some small itsp's
don't have a clue how to support portability or even how to accomplish the
transfer. As you can probably guess, a lot of this has to do with capabilities
to support SS7 (either directly or through another provider), access to shared
databases, etc.


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RE: [Asterisk-Users] Asterisk Dial Failover

2005-12-09 Thread John Cianfarani
Ryan/Jonathan,

Couple quick questions regarding your setup?
Do you operate this in a strictly master/slave setup? 
Do you have anything(mon/ha's internal status/monitor options) that
actually monitors the asterisk process (to determine if it is hung). Or
is it only with total box failure to you fail over?
Do you use something to sync config/vm/cdr? Rsync/unison?

Thanks
John

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second
failover. I think there is a article on voip-info.org about this, but
don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan

 All,

 I have an Asterisk system that sends PSTN calls to an OpenSER system
to be
 routed. I have a command like this in my extensions.conf:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)

 There's actually two OpenSER systems for redundancy. I'm trying to
find a
 way to have Asterisk attempt to route the call to one OpenSER system,
and
 if it's down, fallback to another.

 Any first thoughts on how to achieve this?

 I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups
 are broken. If I issue a series of Dial commands, such as this:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
 exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)

 ... what seems to happen is that when proxy1 is down, Asterisk waits
the
 full 20 seconds before returning control. Also, This 20s includes the
time
 is takes for the other end to answer, so if I put a small value of say
5s
 in there, the dial command will probably give up before someone
answers at
 the other end. Neither is workable.

 Asterisk SHOULD be able to distinguish between a TRYING and no
response.
 In the event it gets no TRYING response to a dial command within a
 specified timeout it should return control and flag an error. If on
the
 other hand it does get a TRYING response (and maybe a RINGING too) it
 should continue to wait until the 20s has expired.

 I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)
 because Asterisk reads the extensions.conf on startup and also seems
to
 cache what the host maps to on startup. Subsequent calls to the host
 always return the same IP address.

 But... in general... how have people implemented this?

 Help appreciated!
 Doug







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RE: [Asterisk-Users] Asterisk Dial Failover

2005-12-09 Thread Douglas Garstang
Adam,

An Audicodes Mediant 2000 gateway with a couple of PRI's. 
Why?

Doug.

-Original Message-
From: Adam Robins [mailto:[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


What are you using to terminate the PSTN calls and do the SIP
transcoding? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second failover. I think there is a article on voip-info.org about
this, but don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan

 All,

 I have an Asterisk system that sends PSTN calls to an OpenSER system
to be
 routed. I have a command like this in my extensions.conf:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)

 There's actually two OpenSER systems for redundancy. I'm trying to
find a
 way to have Asterisk attempt to route the call to one OpenSER system,
and
 if it's down, fallback to another.

 Any first thoughts on how to achieve this?

 I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups
 are broken. If I issue a series of Dial commands, such as this:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
 exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)

 ... what seems to happen is that when proxy1 is down, Asterisk waits
the
 full 20 seconds before returning control. Also, This 20s includes the
time
 is takes for the other end to answer, so if I put a small value of say
5s
 in there, the dial command will probably give up before someone
answers at
 the other end. Neither is workable.

 Asterisk SHOULD be able to distinguish between a TRYING and no
response.
 In the event it gets no TRYING response to a dial command within a 
 specified timeout it should return control and flag an error. If on
the
 other hand it does get a TRYING response (and maybe a RINGING too) it 
 should continue to wait until the 20s has expired.

 I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)
 because Asterisk reads the extensions.conf on startup and also seems
to
 cache what the host maps to on startup. Subsequent calls to the host 
 always return the same IP address.

 But... in general... how have people implemented this?

 Help appreciated!
 Doug







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The contents of this email message and any attachments are confidential and are 
intended solely for addressee. The information may also be legally privileged. 
This transmission is sent in trust, for the sole purpose of delivery to the 
intended recipient. If you have received this transmission in error, any use, 
reproduction or dissemination of this transmission is strictly prohibited. If 
you are not the intended recipient, please immediately notify the sender by 
reply email and delete this message and its attachments, if any.


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[Asterisk-Users] Wait for X rings before answering?

2005-12-09 Thread Robert La Ferla
How do I set up extensions.conf to wait for x rings (ringing all 
extensions) before answering?  I'm trying to mimic a regular answering 
machine on an multiple analog phone system.  Currently, Asterisk picks 
up after 1 ring and just tries to dial one extension.  I want all 
extensions to ring.


[incoming]
exten = s,1,Dial(SIP/myext,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()

Also, I couldn't find documentation on the r option for Dial().

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[Asterisk-Users] Teliax experiences

2005-12-09 Thread Rolf Brusletto
Howdy  - This is my first post on the list, and from what I've seen of * I'm
very impressed. I had a question regarding everybodys experience with Teliax
or Broadvoice. I setup a Teliax trunk this morning, and had calls going out
it in about 5 minutes(Had to get more coffee). Has anybody had any problems
with them, outages, issues with dids etc??


Thanks, 

Rolf Brusletto
Denver, Co. 

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RE: [Asterisk-Users] Asterisk Dial Failover

2005-12-09 Thread Douglas Garstang
Yes, that's a great question. I'm wondering the same thing. Can these heartbeat 
apps monitor applications as well as network connectivity? The heartbeat 
utility at www.linux-ha.org talks about monitoring some standard apps like web 
servers and such but isn't clear about other apps... like Asterisk or SER.

-Original Message-
From: John Cianfarani [mailto:[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 8:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


Ryan/Jonathan,

Couple quick questions regarding your setup?
Do you operate this in a strictly master/slave setup? 
Do you have anything(mon/ha's internal status/monitor options) that
actually monitors the asterisk process (to determine if it is hung). Or
is it only with total box failure to you fail over?
Do you use something to sync config/vm/cdr? Rsync/unison?

Thanks
John

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second
failover. I think there is a article on voip-info.org about this, but
don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan

 All,

 I have an Asterisk system that sends PSTN calls to an OpenSER system
to be
 routed. I have a command like this in my extensions.conf:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)

 There's actually two OpenSER systems for redundancy. I'm trying to
find a
 way to have Asterisk attempt to route the call to one OpenSER system,
and
 if it's down, fallback to another.

 Any first thoughts on how to achieve this?

 I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups
 are broken. If I issue a series of Dial commands, such as this:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
 exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)

 ... what seems to happen is that when proxy1 is down, Asterisk waits
the
 full 20 seconds before returning control. Also, This 20s includes the
time
 is takes for the other end to answer, so if I put a small value of say
5s
 in there, the dial command will probably give up before someone
answers at
 the other end. Neither is workable.

 Asterisk SHOULD be able to distinguish between a TRYING and no
response.
 In the event it gets no TRYING response to a dial command within a
 specified timeout it should return control and flag an error. If on
the
 other hand it does get a TRYING response (and maybe a RINGING too) it
 should continue to wait until the 20s has expired.

 I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)
 because Asterisk reads the extensions.conf on startup and also seems
to
 cache what the host maps to on startup. Subsequent calls to the host
 always return the same IP address.

 But... in general... how have people implemented this?

 Help appreciated!
 Doug







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[Asterisk-Users] X100 clone

2005-12-09 Thread Vladimir Montealegre

Wath brands of modem or chip's work with asterisk?

Intel 537EP
Ambient MD3200
Motorola 62802

and i mix the 3 types of modem in 1 pcpbx? or how i do to mannage tree phone
lines?

thnks in advance

Vladimir 


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[Asterisk-Users] Echo PSTN [EMAIL PROTECTED] 2.0 Digium TDM11B DSL

2005-12-09 Thread David K Parker
I have a Digium TDM11B, I'm fighting an issue with with echo on the PSTN side. I run [EMAIL PROTECTED] 2.0. I have an analog phone on the FXS channel 1 and Telco on the FXS channel 4. I also have a coupe of softphones, 1 iax2, the other sip, and a LinkSys Sipura 941. I use a VOIP provider for long distance. I'm experiencing echo on all calls on any phone for calls going out over the PSTN, but no echo at all on Long Distance calls with my VOIP provider or Internal calls. I think its safe to say that echo is occuring on the PSTN side on channel 4. I've followed the trouble shooting provedures on 
voip-info.org for echo cancellation, even calling the local CO using ztmonitor to adjust rx  tx gain. The only thing I haven't tried yet is installing shielded cable. I use Verizon DSL for Internet and have the appropriate filter for my PSTN on channel 4. I'm beginning to wonder if the problem is due to DSL. Has anyone else had this experience.
Thanks,David
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Re: [Asterisk-Users] Asterisk on PPC chan_capi issue

2005-12-09 Thread Jason Williams


  chan_capi registers fine:  **
 [chan_capi.so] = (Common ISDN API for Asterisk)  == This box has 1 capi controller(s).  == Reading config for BRI1  -- ast_capi_pvt BRI1-pseudo-D (MSN1,MSN2,capi-in,0,2) (1,4,128)
  -- ast_capi_pvt BRI1 (MSN1,MSN2,capi-in,0,2) (1,4,128)  -- ast_capi_pvt BRI1 (MSN1,MSN2,capi-in,0,2) (1,4,128)  -- listening on contr1 CIPmask = 0x1fff03ff
  == Registered channel type 'CAPI' (Common ISDN API Driver ($Revision:  1.115 $) )  == Registered application 'capiCommand'  == Registered custom function VANITYNUMBER
   Call from my GSM to a SIP phone (exten 1003) via ISDN/CAPI (MSN2):  **  == BRI1: Incoming call 'my GSM' - 'MSN2'
   -- Executing Macro(CAPI/BRI1/MSN2-0, stdexten|1003|SIP/1003)  in new stack  -- Executing Dial(CAPI/BRI1/MSN2-0, SIP/1003|10|TtwW) in new
  stack  Dec6 02:30:47 WARNING[28889]: channel.c:2494 ast_request: No  translator path exists for channel type SIP (native 65535) to 0  Dec6 02:30:47 NOTICE[28889]: app_dial.c:1011 dial_exec_full: Unable to
  create channel of type 'SIP' (cause 0 - Unknown)  == Everyone is busy/congested at this time (1:0/0/1)

Looks like a codec problem when making calls to the SIP phone, ensure your sip phone has Alaw enabled in sip.conf, and supports the g711alaw codec. In its config


Jason



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Re: [Asterisk-Users] Porting a phone number to a voip provider

2005-12-09 Thread Rich Adamson
   Just an FYI... not all cell numbers are portable.
  
  Do you have any more information on this?  I read somewhere that 
  sometimes you can port a number to a VoIP provider but not be able to 
  port it back to the PSTN because not all PSTN providers will take 
  numbers from VoIP providers.  Is this what you are talking about?
 
 No. Some cell providers do not have the capability to selectively let
 a telephone number move to another carrier/telco/itsp. They may have
 technical plans to allow/support it, but the hardware/software necessary
 to allow/support is not yet in place. Same is true with some smaller
 telcos.
 
 I'd also have to guess that come cell providers have probably taken a
 stand that says they aren't going to do it, regardless of what the 
 regulatory folks do/say.
 
 So, the issue with number portability really starts with the company that
 currently provides the cell number to you. (Sometimes they don't like
 to volunter that info for obvious reasons.) If your number is portable,
 then it should remain portable regardless of where you move it to, and
 you should be able to move it as many times as you'd like.
 
 If you move your number to the xyz itsp and don't like their service,
 then pick another company and ask them to move the number for you. For
 them to move it, however, requires an acknowledgement from the previous
 company in most cases. In some (rather rare) cases, the previous company
 will sometimes drag their feet or refuse to acknowledge the transfer. There
 are regulatory escalation processes to address that problem, but usually
 it takes a considerable amount of time to get it done.
 
 In some cases, even the larger cell providers play games as they don't want
 to lose their numbers, customers, etc. In other cases, some small itsp's
 don't have a clue how to support portability or even how to accomplish the
 transfer. As you can probably guess, a lot of this has to do with capabilities
 to support SS7 (either directly or through another provider), access to shared
 databases, etc.

Oh, and I forgot to mention that DID numbers from itsp's are frequently
considered non-portable. Most of that is because the itsp's have contracted
with other companies to get those numbers, and the itsp cannot represent
those as portable since they really have no management control over them.

For example, teliax.com offers DID's from a large number of area codes and
exchanges, but they have contracted with others to get access to them. It's
not up to teliax to say whether those are portable or not since they are
not the end office responsible for those numbers. (They might know however.)


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RE: [Asterisk-Users] Asterisk Dial Failover

2005-12-09 Thread Adam Robins
Doug, 

We currently are using Digium TE410P boards directly into each Asterisk
server.  I've been researching various gateways, up to DS3 capacity, to
convert PRI to SIP and then allocate the SIP among multiple Asterisk
servers.  I've looked at Cisco AS5400 (), Lucent APX 1000 ($$$), and
Quintum Tenor CMS ($$).

Thanks,
Adam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Friday, December 09, 2005 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

Adam,

An Audicodes Mediant 2000 gateway with a couple of PRI's. 
Why?

Doug.

-Original Message-
From: Adam Robins [mailto:[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


What are you using to terminate the PSTN calls and do the SIP
transcoding? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second failover. I think there is a article on voip-info.org about
this, but don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan

 All,

 I have an Asterisk system that sends PSTN calls to an OpenSER system
to be
 routed. I have a command like this in my extensions.conf:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)

 There's actually two OpenSER systems for redundancy. I'm trying to
find a
 way to have Asterisk attempt to route the call to one OpenSER system,
and
 if it's down, fallback to another.

 Any first thoughts on how to achieve this?

 I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups
 are broken. If I issue a series of Dial commands, such as this:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
 exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)

 ... what seems to happen is that when proxy1 is down, Asterisk waits
the
 full 20 seconds before returning control. Also, This 20s includes the
time
 is takes for the other end to answer, so if I put a small value of say
5s
 in there, the dial command will probably give up before someone
answers at
 the other end. Neither is workable.

 Asterisk SHOULD be able to distinguish between a TRYING and no
response.
 In the event it gets no TRYING response to a dial command within a 
 specified timeout it should return control and flag an error. If on
the
 other hand it does get a TRYING response (and maybe a RINGING too) it 
 should continue to wait until the 20s has expired.

 I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)
 because Asterisk reads the extensions.conf on startup and also seems
to
 cache what the host maps to on startup. Subsequent calls to the host 
 always return the same IP address.

 But... in general... how have people implemented this?

 Help appreciated!
 Doug







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transmission is strictly prohibited. If you are not the intended
recipient, please 

Re: [Asterisk-Users] Teliax experiences

2005-12-09 Thread Rich Adamson

 Howdy  - This is my first post on the list, and from what I've seen of * I'm
 very impressed. I had a question regarding everybodys experience with Teliax
 or Broadvoice. I setup a Teliax trunk this morning, and had calls going out
 it in about 5 minutes(Had to get more coffee). Has anybody had any problems
 with them, outages, issues with dids etc??

They have been very good for us over the last six months. You might
consider searching the list archives for things like this since there
have been a lot of similar postings (not just teliax) over the last
year or so.

Very few outages, good to excellent support, pretty solid calls. They
are apparently a little behind in their asterisk code as some things
like iax trunking with jitterbuffer isn't supported.


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RE: [Asterisk-Users] Asterisk Dial Failover

2005-12-09 Thread Ashley Wright
Hi I use a allied telesyn at-vp730
Works quite well

ash

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: 09 December 2005 16:01
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

Doug, 

We currently are using Digium TE410P boards directly into each Asterisk
server.  I've been researching various gateways, up to DS3 capacity, to
convert PRI to SIP and then allocate the SIP among multiple Asterisk
servers.  I've looked at Cisco AS5400 (), Lucent APX 1000 ($$$), and
Quintum Tenor CMS ($$).

Thanks,
Adam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Friday, December 09, 2005 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

Adam,

An Audicodes Mediant 2000 gateway with a couple of PRI's. 
Why?

Doug.

-Original Message-
From: Adam Robins [mailto:[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


What are you using to terminate the PSTN calls and do the SIP
transcoding? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second failover. I think there is a article on voip-info.org about
this, but don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan

 All,

 I have an Asterisk system that sends PSTN calls to an OpenSER system
to be
 routed. I have a command like this in my extensions.conf:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)

 There's actually two OpenSER systems for redundancy. I'm trying to
find a
 way to have Asterisk attempt to route the call to one OpenSER system,
and
 if it's down, fallback to another.

 Any first thoughts on how to achieve this?

 I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups
 are broken. If I issue a series of Dial commands, such as this:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
 exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)

 ... what seems to happen is that when proxy1 is down, Asterisk waits
the
 full 20 seconds before returning control. Also, This 20s includes the
time
 is takes for the other end to answer, so if I put a small value of say
5s
 in there, the dial command will probably give up before someone
answers at
 the other end. Neither is workable.

 Asterisk SHOULD be able to distinguish between a TRYING and no
response.
 In the event it gets no TRYING response to a dial command within a 
 specified timeout it should return control and flag an error. If on
the
 other hand it does get a TRYING response (and maybe a RINGING too) it 
 should continue to wait until the 20s has expired.

 I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)
 because Asterisk reads the extensions.conf on startup and also seems
to
 cache what the host maps to on startup. Subsequent calls to the host 
 always return the same IP address.

 But... in general... how have people implemented this?

 Help appreciated!
 Doug







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RE: [Asterisk-Users] Asterisk Dial Failover

2005-12-09 Thread Douglas Garstang
I think the Audiocodes boxes run at about $19,000 each.

-Original Message-
From: Adam Robins [mailto:[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 9:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


Doug, 

We currently are using Digium TE410P boards directly into each Asterisk
server.  I've been researching various gateways, up to DS3 capacity, to
convert PRI to SIP and then allocate the SIP among multiple Asterisk
servers.  I've looked at Cisco AS5400 (), Lucent APX 1000 ($$$), and
Quintum Tenor CMS ($$).

Thanks,
Adam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Friday, December 09, 2005 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

Adam,

An Audicodes Mediant 2000 gateway with a couple of PRI's. 
Why?

Doug.

-Original Message-
From: Adam Robins [mailto:[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


What are you using to terminate the PSTN calls and do the SIP
transcoding? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second failover. I think there is a article on voip-info.org about
this, but don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan

 All,

 I have an Asterisk system that sends PSTN calls to an OpenSER system
to be
 routed. I have a command like this in my extensions.conf:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)

 There's actually two OpenSER systems for redundancy. I'm trying to
find a
 way to have Asterisk attempt to route the call to one OpenSER system,
and
 if it's down, fallback to another.

 Any first thoughts on how to achieve this?

 I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups
 are broken. If I issue a series of Dial commands, such as this:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
 exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)

 ... what seems to happen is that when proxy1 is down, Asterisk waits
the
 full 20 seconds before returning control. Also, This 20s includes the
time
 is takes for the other end to answer, so if I put a small value of say
5s
 in there, the dial command will probably give up before someone
answers at
 the other end. Neither is workable.

 Asterisk SHOULD be able to distinguish between a TRYING and no
response.
 In the event it gets no TRYING response to a dial command within a 
 specified timeout it should return control and flag an error. If on
the
 other hand it does get a TRYING response (and maybe a RINGING too) it 
 should continue to wait until the 20s has expired.

 I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)
 because Asterisk reads the extensions.conf on startup and also seems
to
 cache what the host maps to on startup. Subsequent calls to the host 
 always return the same IP address.

 But... in general... how have people implemented this?

 Help appreciated!
 Doug







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Re: [Asterisk-Users] Phone Information

2005-12-09 Thread C F
Can you please explain?
Whats CM?
Whats Astericks?

On 12/9/05, James Horn [EMAIL PROTECTED] wrote:
 On the CM, there is away to get the Device Information, Network
 Configuration, etc. by httping to the phones IP address. Is there away to do
 this via Astericks?
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[Asterisk-Users] Asteriskguru Queue Statistics version 0.7 released

2005-12-09 Thread Zoa


Hello,

After a long period of inactivity we are proud to bring you a new 
version of the Queue Statistics.


Main changes in this version are:

- Fixed a nasty bug where calls can't longer than 999 seconds.
- Added the possibility to see reports for all queues.
- some code cleanups

In the next version we will try to support mysql and pgsql. (currently 
we only have pgsql).


More information and a free download link is available from:  
http://www.asteriskguru.com/tools/queue_stats.php


Zoa.




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Re: [Asterisk-Users] CIDNUM CIDNAME

2005-12-09 Thread C F
${CALLERID(type)}
In the CLI:
type show functions to get a list of functions
type show function callerid to get a list of types
In any case read /usr/src/asterisk/docs/README.variables

On 12/9/05, René Enskat [Teamware GmbH] [EMAIL PROTECTED] wrote:

 Does the CIDNUM and CIDNAME is not any longer working?
 How do i get the parts from the CALLERID?
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[Asterisk-Users] Re: ztdummy on FC4

2005-12-09 Thread Jim Duda
I did cvs update -A, which brought in new files.

make clean
make
make install
make config
/etc/rc.d/init.d/zaptel restart
lsmod | grep ztdummy

Ztdummy is loaded.

/etc/rc.d/init.d/asterisk restart

lsmod | grep ztdummy
ztdummy 7816  0
wcfxo  17440  0
wcte11xp   30496  0
wct1xxp21408  0
wct4xxp   108352  0
tor2   94112  0
zaptel195076  12 ztdummy,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2

I don't see the ztdummy module used, but I did last night.

Musiconhold isn't slow like before, but certainly appears cleaner without
ztdummy loaded.  But since ztdummy isn't even loaded, I don't think it has
any effect.

I'm still missing something ... Any ideas?

Thanks so much for your help Tony.

Jim

Tony Mountifield [EMAIL PROTECTED] wrote in message 
news:[EMAIL PROTECTED]
 In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] 
 wrote:
  Check ztdummy.h and ztdummy.c and make doubly sure USE_RTC is defined.
  It sounds like you are using a version compiled without USE_RTC on one
  of the newer kernels that has fewer than 1000 jiffies per second.
 

 I'm using zaptel from CVS (cvs.digium.com).  I just did an update (many
 files out-of-date), however, USE_RTC is NOT to be found in any source
 file.  I could add it, however, doesn't appear to be used anywhere.

 That sound like you might have a sticky tag keeping you on an older
 branch. Could you post the output of cvs status ztdummy.c ? Thanks.

 Tony
 -- 
 Tony Mountifield
 Work: [EMAIL PROTECTED] - http://www.softins.co.uk
 Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [Asterisk-Users] Asterisk Dial Failover

2005-12-09 Thread burke
WEll I personally have not implemented a Linux-HA cluster mainly because I
don't have the resources to do so. I study Asterisk purley as a hobby
(nerd.. yeahI know) because it is an awesome OSS product. Anyways, after
some searching around I think it would not be TOO difficult to implement a
resource failover using a combination of Linux HA (linux-ha.org), Mon
(www.kernel.org/software/mon) and SIPp (sipp.sourceforge.net).

Linux HA supports failover on resources or machine based
failures(inherently if a machine goes down all the resources will be gone
too). However Linux HA just provides the interface to say which machine is
the Active machine in the cluster for a specific resource (aka
Asterisk). That is where Mon would come in. Mon montiors services on a
machine and can be configured to react if a service fails the periodic
test. But Mon does not have a per-defined monitor script for Asterisk.
That is where SIPp would come in. You could create a Mon script that calls
SIPp and looks for the return code after a number of calls through the
server. SIPp will return a 0 if all the calls succeeded. That way in your
Mon script if SIPp returned anything other than a 0 then register it as a
failure. Once a failure occurs you can configure Mon to switch the active
Asterisk server using the Linux HA functionality.

Like I said, there is turn key way of doing this, but it looks like a good
little project for the wiki? Maybe I'll start working on this in my spare
time, I just need to get some time to play with the different components.

There are a few more logicistical things that would have to be taken care,
mainly anything file realted, but that could be alleviated with some kind
of remote mounted filesystem.

Hope this helps,
Ryan

 Yes, that's a great question. I'm wondering the same thing. Can these
 heartbeat apps monitor applications as well as network connectivity? The
 heartbeat utility at www.linux-ha.org talks about monitoring some standard
 apps like web servers and such but isn't clear about other apps... like
 Asterisk or SER.

 -Original Message-
 From: John Cianfarani [mailto:[EMAIL PROTECTED]
 Sent: Friday, December 09, 2005 8:10 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Asterisk Dial Failover


 Ryan/Jonathan,

 Couple quick questions regarding your setup?
 Do you operate this in a strictly master/slave setup?
 Do you have anything(mon/ha's internal status/monitor options) that
 actually monitors the asterisk process (to determine if it is hung). Or
 is it only with total box failure to you fail over?
 Do you use something to sync config/vm/cdr? Rsync/unison?

 Thanks
 John

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
 k. Creasy
 Sent: Friday, December 09, 2005 8:45 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Asterisk Dial Failover

 I chose this method and have been happy with the results.

 -Jonathan

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 [EMAIL PROTECTED]
 Sent: Friday, December 09, 2005 7:51 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Asterisk Dial Failover

 Your other option is to setup the OpenSER boxes in a truly redundant
 configuration using Linux HA (www.linux-ha.org). That way you setup all
 your PSTN calls to forward to one shared virtual IP between the boxes.
 One
 of the boxes is the Master, the other is the Slave. There is a heartbeat
 between the boxes that goes at a configurable rate. If the Master fails
 then the Slave will take over and it can even be configured for
 sub-second
 failover. I think there is a article on voip-info.org about this, but
 don't have time to look it up.

 Good luck and let us know what you choose to do.

 Ryan

 All,

 I have an Asterisk system that sends PSTN calls to an OpenSER system
 to be
 routed. I have a command like this in my extensions.conf:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)

 There's actually two OpenSER systems for redundancy. I'm trying to
 find a
 way to have Asterisk attempt to route the call to one OpenSER system,
 and
 if it's down, fallback to another.

 Any first thoughts on how to achieve this?

 I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
 lookups
 are broken. If I issue a series of Dial commands, such as this:

 exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
 exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)

 ... what seems to happen is that when proxy1 is down, Asterisk waits
 the
 full 20 seconds before returning control. Also, This 20s includes the
 time
 is takes for the other end to answer, so if I put a small value of say
 5s
 in there, the dial command will probably give up before someone
 answers at
 the other end. Neither is workable.

 Asterisk SHOULD be able to distinguish between a TRYING and no
 response.
 In 

[Asterisk-Users] a few questions

2005-12-09 Thread Stas Khromoy

we are beginning to test asterisk for our office
one of the features of the current phone system that is very heavily 
used is overhead paging


now i came accross this post
http://forums.digium.com/viewtopic.php?t=2844highlight=features

that basically says it is not possible with asterisk.

let's hope that i am not understanding it right, since i am new to the 
telephony.


can any one help me out and explain it to the unfortunate ? :))

second question is as follows:

when you access voice mail
the default msg is 'welcome to comedian mail'
is there any way to get rid of this par of the greeting ?


thanks


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Re: [Asterisk-Users] Wait for X rings before answering?

2005-12-09 Thread Derek Whitten
Robert La Ferla wrote:
 How do I set up extensions.conf to wait for x rings (ringing all
 extensions) before answering?  I'm trying to mimic a regular answering
 machine on an multiple analog phone system.  Currently, Asterisk picks
 up after 1 ring and just tries to dial one extension.  I want all
 extensions to ring.
 
 [incoming]
 exten = s,1,Dial(SIP/myext,25,t,r)
 exten = s,2,Voicemail(myext)
 exten = s,3,Hangup()
 
 Also, I couldn't find documentation on the r option for Dial().
 
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[incoming]
exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()
-- 
.


-BEGIN GEEK CODE BLOCK-
Version: 3.1
GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w--
PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y
 --END GEEK CODE BLOCK--


.


signature.asc
Description: OpenPGP digital signature
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[Asterisk-Users] Re: ztdummy on FC4

2005-12-09 Thread Jim Duda
Sorry, forgot to follow directions :-)

linux cvs status ztdummy.c
===
File: ztdummy.c Status: Up-to-date

   Working revision:1.15
   Repository revision: 1.15/usr/cvsroot/zaptel/ztdummy.c,v
   Sticky Tag:  (none)
   Sticky Date: (none)
   Sticky Options:  (none)

Tony Mountifield [EMAIL PROTECTED] wrote in message 
news:[EMAIL PROTECTED]
 In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] 
 wrote:
  Check ztdummy.h and ztdummy.c and make doubly sure USE_RTC is defined.
  It sounds like you are using a version compiled without USE_RTC on one
  of the newer kernels that has fewer than 1000 jiffies per second.
 

 I'm using zaptel from CVS (cvs.digium.com).  I just did an update (many
 files out-of-date), however, USE_RTC is NOT to be found in any source
 file.  I could add it, however, doesn't appear to be used anywhere.

 That sound like you might have a sticky tag keeping you on an older
 branch. Could you post the output of cvs status ztdummy.c ? Thanks.

 Tony
 -- 
 Tony Mountifield
 Work: [EMAIL PROTECTED] - http://www.softins.co.uk
 Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Teliax experiences

2005-12-09 Thread John Reynolds
I use Teliax. I think the sound quality is really very good. I get
about an 80ms ping with them, but a 20ms ping to Junction Networks.
Some how calls to/form Teliax sound better.

With Junction Networks I get great customer service, with Teliax I get
Okay to good customer service (depending on who is responding).

I also use nufone.net  and get good call quality, but no real human
interaction. Fortunately, it just works as advertised. I did have a
800# provisioning problem, that a human corrected for me, it just took
a little time.

JR


On 12/9/05, Rich Adamson [EMAIL PROTECTED] wrote:

  Howdy  - This is my first post on the list, and from what I've seen of * I'm
  very impressed. I had a question regarding everybodys experience with Teliax
  or Broadvoice. I setup a Teliax trunk this morning, and had calls going out
  it in about 5 minutes(Had to get more coffee). Has anybody had any problems
  with them, outages, issues with dids etc??

 They have been very good for us over the last six months. You might
 consider searching the list archives for things like this since there
 have been a lot of similar postings (not just teliax) over the last
 year or so.

 Very few outages, good to excellent support, pretty solid calls. They
 are apparently a little behind in their asterisk code as some things
 like iax trunking with jitterbuffer isn't supported.


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Re: [Asterisk-Users] Change time when * is running

2005-12-09 Thread Remco Barende

On Fri, 9 Dec 2005, Tzafrir Cohen wrote:


On Fri, Dec 09, 2005 at 02:02:53PM +, Julian Lyndon-Smith wrote:

We've just seen that one of our servers is an hour out (it reckons that
it's 15:02 instead of 14:02).

Can I change the time when * is running ? I don't want to try just in
case it causes * some grief.


keep clocks in sync with ntp . Or set the system clock with time.


I do run ntp on my servers, when ntp corrects the time I start getting 
error messages on the * console about music on hold events occurring in 
the past.


It doesn't seem to cause any problems though so I just ignore it.

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RE: [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel

2005-12-09 Thread O'Connor, Jonathan
Not sure I completely agree with all of these.

 Looking at it objectively, Asterisk has many benefits over 
 traditional PBX systems, yet you should be aware of some of 
 the limitations.
 
 Benefits:
 1. Open source / low-cost of ownership / operates on cheap PC 
 hardware. You get voicemail, IVR, hunt-groups etc. without 
 additional fees. Last I checked those are all expensive 
 add-ons in the Nortel world. There aren't expensive licenses 
 per user/handset either. 

You get what you pay for, yes it operates on cheap hardware, but if you
go that route you risk loss whatever system you run.  

 2. Flexibility - you can configure Asterisk to handle calls 
 to a microscopic degree of precision. This is just not 
 possible with traditional PBX systems which are inherently 
 proprietary. Asterisk also makes it easier to present data to 
 callers from CRM, Billing, Order Tracking systems etc. using 
 text-to-speech, automated-speech recognition and/or DTMF recognition. 

I would have to agree mostly...  The Definity ECS we have also has a
level of detail and ability that is close to (and in some areas exceeds)
Asterisk.  Of course that's a 24000 handset capable system so I would
hope it does :)

 3. Flexibility again - It really is much more flexible than 
 anything else!!

If you consider cost yes, otherwise you have to take a strong look at
some of the VoIP offerings out there.  I don't want to sound like a huge
Avaya fan, but their newer IP stuff is being designed from a whole new
perspective.

 4. Supports multiple VoIP protocols - SIP, IAX, H323, (and skinny to a
 degree) and supports connection of a broad spectrum of third 
 party handsets
 - e.g. Cisco, Siemens, Sipura, etc. IAX is a proprietary 
 protocol for Asterisk but it has some benefits over SIP 
 (supposedly - my experience has been a little different) and 
 perhaps more importantly is gaining popularity among VoIP 
 service providers.

This I love about it.  I use Atcom AT320 phones here for home users with
cable/DSL and only have to have one firewall port open for them, its
beautiful in its simplicity.  Internally we use Polycoms running SIP and
Ciscos Plus a few ATAs and softphones running whatever the user
prefers!

 1. Digium PSTN interface boards are not as cheap as they 
 could be and haven't been around long enough for us to have 
 meaningful data on how reliable they are.

I agree they havent been around that long, however I have never spent
more then $600 on a single port T1 card and that's both cheaper then the
ones for my traditional PBX and other manifacturers I have seen.  They
have to make a profit, and I cant see that sort of card with this small
a market compared to other devices being able to come down much more...

 2. Complexity. Asterisk is powerful but it is complicated - 
 which is it You will need to spend a few weeks solidly 
 learning about Asterisk and playing with it in a test 
 environment before even thinking about trying to install it 
 in a production environment. Clearly your time has a cost to 
 your employer - thus this may be perceived as problem with 
 Asterisk. You can of course buy in the services of an 
 Asterisk consultant to help set things up - but ideally you 
 want to have someone on site with some degree of knowledge 
 about Asterisk's capabilities. If your business has basic 
 telephony requirements, doesn't need fancy features and wants 
 to minimize the need for on-site technical expertise to 
 support Asterisk, then a Mitel/Nortel solution MIGHT make 
 sense. IMHO - the present level of complexity/flexibility is 
 the biggest strength and weakness to Asterisk.

Agree 100%, however its not alone here  I have an Avaya Definity, a
Nortel and a Vodavi switch in this company to run...  In the end the
Avaya is slightly easier to manage then Asterisk but not much, and
both are FAR easier then the other two.

That said, Asterisk is the glue that bonds them, in that each one is
connected to an Asterisk server with a T1 card and we have 4 digit
dialing throughout our enterprise because of it, over IAX trunks.

 3. Asterisk is a work in progress. Yes it's pretty stables 
 and yes it's being used in very large production systems from 
 what one hears on this list. However it's a moving target 
 with new releases appearing frequently.
 On a positive note that's great if you want new features and 
 bug fixes - but it can also be a pain if you want a nice 
 stable, low-maintenance system.

stable, low-maintenance  I wish my Avaya Voicemail was  The Audix
LX is the worst thing they have ever made.  My Vodavi system is a piece
of crap that makes me want to go postal every time I try and get it to
do the simplest thing  Last software upgrade they sent us disabled
all of the Hold/Mute etc... Buttons on the handsets!


 4. Cost savings aren't necessarily as great at they first 
 seem. You ideally want to have redundancy on your Asterisk 
 set up. To support 75 users you probably want to have a 
 

RE: [Asterisk-Users] a few questions

2005-12-09 Thread Kerry Garrison
Overhead paging is totally possible, there are several articles available on
how to do it. But you cannot have multiple zones today unless you use a sip
device that has autoanswer. 

Easiet way to remove that message is to replace the file with one that only
has a split second of silence.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stas Khromoy
Sent: Friday, December 09, 2005 8:21 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] a few questions

we are beginning to test asterisk for our office one of the features of the
current phone system that is very heavily used is overhead paging

now i came accross this post
http://forums.digium.com/viewtopic.php?t=2844highlight=features

that basically says it is not possible with asterisk.

let's hope that i am not understanding it right, since i am new to the
telephony.

can any one help me out and explain it to the unfortunate ? :))

second question is as follows:

when you access voice mail
the default msg is 'welcome to comedian mail'
is there any way to get rid of this par of the greeting ?


thanks


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Re: [Asterisk-Users] Wait for X rings before answering?

2005-12-09 Thread Robert La Ferla

Derek Whitten wrote:

[incoming]
exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()
  
Thanks.  This will call/ring multiple extensions but what about waiting 
for X rings before going to voicemail?  How do I do that?



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RE: [Asterisk-Users] a few questions

2005-12-09 Thread Kerry Garrison
That article is about shared call appearance. I have this working using
Grandstream GXP-2000's. It's a great new feature.
-Kerry


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stas Khromoy
Sent: Friday, December 09, 2005 8:21 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] a few questions

we are beginning to test asterisk for our office one of the features of the
current phone system that is very heavily used is overhead paging

now i came accross this post
http://forums.digium.com/viewtopic.php?t=2844highlight=features

that basically says it is not possible with asterisk.

let's hope that i am not understanding it right, since i am new to the
telephony.

can any one help me out and explain it to the unfortunate ? :))

second question is as follows:

when you access voice mail
the default msg is 'welcome to comedian mail'
is there any way to get rid of this par of the greeting ?


thanks


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RE: [Asterisk-Users] Wait for X rings before answering?

2005-12-09 Thread Kerry Garrison
Create a ring group, put all the extensions into the ring group, set your
dialplan to go to the ring group first and then failover to a voicemail
extension.
-Kerry
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten
Sent: Friday, December 09, 2005 8:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Wait for X rings before answering?

Robert La Ferla wrote:
 How do I set up extensions.conf to wait for x rings (ringing all
 extensions) before answering?  I'm trying to mimic a regular answering 
 machine on an multiple analog phone system.  Currently, Asterisk picks 
 up after 1 ring and just tries to dial one extension.  I want all 
 extensions to ring.
 
 [incoming]
 exten = s,1,Dial(SIP/myext,25,t,r)
 exten = s,2,Voicemail(myext)
 exten = s,3,Hangup()
 
 Also, I couldn't find documentation on the r option for Dial().
 
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[incoming]
exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()
--
.


-BEGIN GEEK CODE BLOCK-
Version: 3.1
GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w--
PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y
 --END GEEK CODE BLOCK--


.


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Re: [Asterisk-Users] Phone Information

2005-12-09 Thread Time Bandit
 Can you please explain?
 Whats CM?
I think this is for (Cisco) Call Manager

 Whats Astericks?
Maybe it's in the same part as Atérisk (only the french-speaking
will laugh this one)
;)
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RE: [Asterisk-Users] Phone Information

2005-12-09 Thread Lawrence Jovellanos
He maybe referring to Cisco's Call Manager

-Original Message-
From: C F [mailto:[EMAIL PROTECTED] 
Sent: Friday, December 09, 2005 11:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Phone Information

Can you please explain?
Whats CM?
Whats Astericks?

On 12/9/05, James Horn [EMAIL PROTECTED] wrote:
 On the CM, there is away to get the Device Information, Network
 Configuration, etc. by httping to the phones IP address. Is there away to
do
 this via Astericks?
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[Asterisk-Users] t38 support in latest asterisk release

2005-12-09 Thread Erick Perez
Hi guys, given 1.2.1 is out. How is the t38/fax support going on?

also, can someone point me to proven brands/configs with ip fax capable
machines? Fax machines with a lan port (i heard of them but havent
found them online).
or a fax machine plugged to a converter that actually works for heavy faxing? like fax machines with sipura?

Thanks for your commentsErick.


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[Asterisk-Users] Re: ztdummy on FC4

2005-12-09 Thread Tony Mountifield
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote:
 I did cvs update -A, which brought in new files.
 
 make clean
 make
 make install
 make config
 /etc/rc.d/init.d/zaptel restart
 lsmod | grep ztdummy
 
 Ztdummy is loaded.
 
 /etc/rc.d/init.d/asterisk restart
 
 lsmod | grep ztdummy
 ztdummy 7816  0
 wcfxo  17440  0
 wcte11xp   30496  0
 wct1xxp21408  0
 wct4xxp   108352  0
 tor2   94112  0
 zaptel195076  12 ztdummy,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2
 
 I don't see the ztdummy module used, but I did last night.
 
 Musiconhold isn't slow like before, but certainly appears cleaner without
 ztdummy loaded.  But since ztdummy isn't even loaded, I don't think it has
 any effect.

Now you're confusing me, or confused. In what you just quoted, ztdummy is
the first in the list, just above wcfxo, and also listed as a caller of
the zaptel module.

However, firstly, do you have any Digium cards in the system? If not, you
should not have any modules loaded except for ztdummy and zaptel. tor2 and
the ones beginning with w are not required, and should be commented out
in your file /etc/rc.d/init.d/zaptel.

Look in that file for the MODULES and RMODULES lines, and set them to:

MODULES=ztdummy
RMODULES=ztdummy

This is assuming you don't have Digium cards installed. If you do, those
lines should list only the required drivers, and not ztdummy. You should
never load ztdummy if you have a Digium card.

 I'm still missing something ... Any ideas?

I'm still concerned you can't find reference to USE_RTC. In ztdummy.c
there should be some lines like the following, fairly near to top:

/*
 * NOTE: (only applies to kernel 2.6)
 * If using an i386 architecture without a PC real-time clock,
 * the #define USE_RTC should be commented out.
 */
#if defined(__i386__)
#if LINUX_VERSION_CODE = VERSION_CODE(2,6,13)
#define USE_RTC
#else
#if 0
#define USE_RTC
#endif
#endif
#endif

And then there are #ifdef lines further down based on USE_RTC.

If you can't find those, then you *definitely* do not have the current version.

Also, you should edit the above section to change the #if 0 to #if 1
(the #if 0 was a panic reaction to a mistake, and unfortunately has never
been undone).

 Thanks so much for your help Tony.

You're welcome.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Wait for X rings before answering?

2005-12-09 Thread Dave Cotton
On Fri, 2005-12-09 at 11:41 -0500, Robert La Ferla wrote:
 Derek Whitten wrote:
  [incoming]
  exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
  exten = s,2,Voicemail(myext)
  exten = s,3,Hangup()

 Thanks.  This will call/ring multiple extensions but what about waiting 
 for X rings before going to voicemail?  How do I do that?

What do you think the 25 does?

Maybe it's a time or something.

-- 
Dave Cotton [EMAIL PROTECTED]

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[Asterisk-Users] testers needed for channel.c jitter buffer (better known as SIP jitter buffer)

2005-12-09 Thread Zoa


Hey ho,

About a week ago i uploaded a channel.c jitter buffer on mantis, this is 
the patch many of you have been waiting for. It's supposed to be rock 
stable (stability wise), but needs some more audio quality testing.
This channel.c jitter buffer implementation is a channel independent 
jitter buffer, currently only implemented for SIP (but easy to add to 
all protocols as long as they provide timestamps).


When testing, please post the results on mantis, don't forget to mention 
the jb options and ATA you used.


Code and comments can be found on:
http://bugs.digium.com/view.php?id=3854

Cheers,

Zoa.
www.asteriskguru.com
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Re: [Asterisk-Users] Wait for X rings before answering?

2005-12-09 Thread José Luis Gómez
If you need more rings, increase 25 (that is in seconds, more or less).
Regards.

El vie, 09-12-2005 a las 11:41 -0500, Robert La Ferla escribió:
 Derek Whitten wrote:
  [incoming]
  exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
  exten = s,2,Voicemail(myext)
  exten = s,3,Hangup()

 Thanks.  This will call/ring multiple extensions but what about waiting 
 for X rings before going to voicemail?  How do I do that?
 
 
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Qualis Information Technology
Av. Rivadavia 2553, PB Of. 43 EP
0342-4565684 int 102
Bs. As. 011-51990896
www.qualis.com.ar
Soporte 0810-8880022
Santa Fe - Argentina

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Re: [Asterisk-Users] DID Providers

2005-12-09 Thread Rehan Ahmed
Dear Aaron,

You can check out www.didx.org US did's start at 10 cents a number per month. uk start 30 cents per month.

You get 2 free did's to test the didx services.

If you can PROVIDE us Japan did's that would be super cool.

Rehan
On 12/9/05, Aaron Anderson [EMAIL PROTECTED] wrote:
Gentelmen (and ladies too of course),Just a quick question.I run an internet provider here in Japan and we want to start offering
US DIDs to some of our US military customers.Does anyone have a link to some good information about DIDs and settingthem up under asterisk?Also, perhaps a few links to providers of DIDsin the US so I can get some rates for numbers?
Thanks in advanceAaron___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Rehan Ahmed AllahWala
http://www.SuperTec.com - Tommrow's Technology, Today.http://www.didx.net - DID Number Exchange and Peering Service.
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RE: [Asterisk-Users] Nortel Meridian Option81C to TE405P

2005-12-09 Thread Schochet, Wes



Yes, I have other PRIs. problem is that there are 
100 different fields to fill in on the M1, but only 20 on the zap/asterisk 
side! 

I was able to get this 
going.Afewpoints:

1. I am using the Sangoma Card - works 
great.
2. I have the M1 
set for USR and the Asterisk set for NET
3. The D-Channel messages on the Asterisk side 
are generated by the Asterisk application (in hindsight - duh!). The 
d-channel won't come up unless the application is 
running.

I had the clocking on the M1 set for "External" on the 
ADAN / D-Channel. The clocking on the Sangoma has two choices: "Master" 
and "Normal". I picked "Master" because I figured the M1 would be the 
"Slave" from a clocking standpoint. This generated a ton of errors and 
took me a long time to figure out. (Why the hell can't Nortel give 
you an inline description of an error message?After all of 
these years, you still get DTA0021 as an error message!) I changed the 
clock on the Sangoma to Normal and the span came right 
up.

I may have also had a physical layer problem - maybe a 
bad DB15 to RJ48 converter.

Now I'm in dial plan and MeetMe hell - but I'll get by 
that too!



From: Joe Pukepail [mailto:[EMAIL PROTECTED] 
Sent: Thursday, December 08, 2005 9:44 PMTo: Schochet, 
WesSubject: Re: [Asterisk-Users] Nortel Meridian Option81C to 
TE405P
yup, is this the only PRI you have coming into your nortel? we 
already had 2 other PRI's coming in so pretty much just copied the config 
settings from one of the other PRIs. What does it show if you stat the 
d-channel. (I think it is in load 96: stat dch d channel 
number). Everything else setup on the nortel? Have a clock 
source setup and everything? 
On 12/8/05, Schochet, 
Wes [EMAIL PROTECTED] 
wrote: 

  Joe, are 
  you running PRI to your Opt 11? I have a 61 and I can;t get my d-channel 
  to come up to save my life!
  
  
  From: Joe Pukepail [mailto:[EMAIL PROTECTED]] 
  Sent: Thursday, December 08, 2005 9:21 AM To: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Nortel Meridian Option81C to 
  TE405P
  
  
  You can't use a ethernet crossover cable, make sure you are using a T1 
  crossover cable. (you will definately need to use a T1 crossover cable). 
  
  
  I'm running a Nortel Option 11 and Asterisk connected in this manner. 
  
  
  On 12/8/05, 
  Steve Totaro  
  [EMAIL PROTECTED] wrote: 
  
He said that he is using a 
crossover but for some reason I think thecrossover may be the 
problem.Try making a new one.Cross pin one with 
four and two with five.Also try a straight through 
cable.Your configs look fine on the asterisk side although I 
am not real cluefullon the Meridian.One question, was the 
Meridian ever hooked up to the PSTN? 
Thanks,Steve This might be an obvious 
question, but should you be using a crossover  
cable? Information on setting up Nortel to TDM card links 
can be found at:  http://www.pham.org/asterisk/asterisk-meridian-a1.pdf 
Regards, -- Anthony Rodgers Business Systems 
Analyst District of North Vancouver Web: http://www.dnv.org RSS 
Feed: http://www.dnv.org/rss.asp On Dec 
6, 2005, at 2:59 PM, Anish Basu wrote:   Hi, 
  I am having problems connecting a Nortel Meridian Option 
81C PBX tomy  Asterisk 1.20 server. We are using the TE405P 
card with onecrossover  PRI   T1 cable 
connecting the two systems. The lights on the back of the  
TE405P  are green and zttool shows that the span is okay. Calls 
cannot be  made and  'pri show span 1' shows the 
d-channel as down. If anyone has any   experience  
with this, suggestions and tips are greatly appreciatd. If 
wecannot  get  this resolved within the next few 
days, we are willing to pay  consulting   fees for 
help. The config files are as listed below. Thanks forany  
help  in advance.
zaptel.conf  ---  loadzone = us  
 defaultzone=us  span=1,0,0,esf,b8zs  
bchan=1-23  dchan=24   
zapata.conf  ---  [trunkgroups] 
  [channels]   language=en  
switchtype=5ess  context=from-pbx  
signalling=pri_net  group=1  callgroup=1 
 pickupgroup=1  channel = 1-23   
usecallerid=yes  hidecallerid=no  
callwaiting=yes  callwaitingcallerid=yes  
threewaycalling=yes  transfer=yes  
canpark=yes  cancallforward=yes   
callreturn=yes  echocancel=yes  
echocancelwhenbridged=yes  rxgain=0.0  
txgain=0.0  faxdetect=both  
musiconhold=default   Nortel configuration: 
b-channel,d-channel, and route data block  
---  
REQ prt  TYPE adan dch 10   ADAN DCH 10 
  CTYP MSDL  GRP 3  DNUM 2  
PORT 0  DES VresaBridge  USR PRI  DCHL 
101  OTBF 32  PARM RS422 DTE  DRAT 64KC 
  CLOK EXT  IFC ESS5  SIDE USR 
 CNEG 1  RLS ID 1  RCAP ND2  MBGA 
NO  OVLR NO  OVLS NO  T200 3 
 T203 10   N200 3  N201 260  K 
7ROUT 1   

Re: [Asterisk-Users] Phone Information

2005-12-09 Thread Jeffery Chen
hehe, :-)

On 12/10/05, Lawrence Jovellanos [EMAIL PROTECTED] wrote:
He maybe referring to Cisco's Call Manager-Original Message-From: C F [mailto:
[EMAIL PROTECTED]]Sent: Friday, December 09, 2005 11:09 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Phone InformationCan you please explain?Whats CM?
Whats Astericks?On 12/9/05, James Horn [EMAIL PROTECTED] wrote: On the CM, there is away to get the Device Information, Network Configuration, etc. by httping to the phones IP address. Is there away to
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RE: [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel

2005-12-09 Thread Schochet, Wes
The other thing I'll say about my PBX is that there is no comparison between
my Nortel i2004 and any  SIP phone I've seen.  Yes, the cost is slightly
more, but for an instrument that I interact with constantly - there is no
SIP device to compare.  I know there will be eventually, but not now!

-Original Message-
From: O'Connor, Jonathan [mailto:[EMAIL PROTECTED] 
Sent: Friday, December 09, 2005 10:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel

Not sure I completely agree with all of these.

 Looking at it objectively, Asterisk has many benefits over traditional 
 PBX systems, yet you should be aware of some of the limitations.
 
 Benefits:
 1. Open source / low-cost of ownership / operates on cheap PC 
 hardware. You get voicemail, IVR, hunt-groups etc. without additional 
 fees. Last I checked those are all expensive add-ons in the Nortel 
 world. There aren't expensive licenses per user/handset either.

You get what you pay for, yes it operates on cheap hardware, but if you go
that route you risk loss whatever system you run.  

 2. Flexibility - you can configure Asterisk to handle calls to a 
 microscopic degree of precision. This is just not possible with 
 traditional PBX systems which are inherently proprietary. Asterisk 
 also makes it easier to present data to callers from CRM, Billing, 
 Order Tracking systems etc. using text-to-speech, automated-speech 
 recognition and/or DTMF recognition.

I would have to agree mostly...  The Definity ECS we have also has a level
of detail and ability that is close to (and in some areas exceeds) Asterisk.
Of course that's a 24000 handset capable system so I would hope it does :)

 3. Flexibility again - It really is much more flexible than anything 
 else!!

If you consider cost yes, otherwise you have to take a strong look at some
of the VoIP offerings out there.  I don't want to sound like a huge Avaya
fan, but their newer IP stuff is being designed from a whole new
perspective.

 4. Supports multiple VoIP protocols - SIP, IAX, H323, (and skinny to a
 degree) and supports connection of a broad spectrum of third party 
 handsets
 - e.g. Cisco, Siemens, Sipura, etc. IAX is a proprietary protocol for 
 Asterisk but it has some benefits over SIP (supposedly - my experience 
 has been a little different) and perhaps more importantly is gaining 
 popularity among VoIP service providers.

This I love about it.  I use Atcom AT320 phones here for home users with
cable/DSL and only have to have one firewall port open for them, its
beautiful in its simplicity.  Internally we use Polycoms running SIP and
Ciscos Plus a few ATAs and softphones running whatever the user prefers!

 1. Digium PSTN interface boards are not as cheap as they could be and 
 haven't been around long enough for us to have meaningful data on how 
 reliable they are.

I agree they havent been around that long, however I have never spent more
then $600 on a single port T1 card and that's both cheaper then the ones for
my traditional PBX and other manifacturers I have seen.  They have to make a
profit, and I cant see that sort of card with this small a market compared
to other devices being able to come down much more...

 2. Complexity. Asterisk is powerful but it is complicated - which is 
 it You will need to spend a few weeks solidly learning about Asterisk 
 and playing with it in a test environment before even thinking about 
 trying to install it in a production environment. Clearly your time 
 has a cost to your employer - thus this may be perceived as problem 
 with Asterisk. You can of course buy in the services of an Asterisk 
 consultant to help set things up - but ideally you want to have 
 someone on site with some degree of knowledge about Asterisk's 
 capabilities. If your business has basic telephony requirements, 
 doesn't need fancy features and wants to minimize the need for on-site 
 technical expertise to support Asterisk, then a Mitel/Nortel solution 
 MIGHT make sense. IMHO - the present level of complexity/flexibility 
 is the biggest strength and weakness to Asterisk.

Agree 100%, however its not alone here  I have an Avaya Definity, a
Nortel and a Vodavi switch in this company to run...  In the end the Avaya
is slightly easier to manage then Asterisk but not much, and both are FAR
easier then the other two.

That said, Asterisk is the glue that bonds them, in that each one is
connected to an Asterisk server with a T1 card and we have 4 digit dialing
throughout our enterprise because of it, over IAX trunks.

 3. Asterisk is a work in progress. Yes it's pretty stables and yes 
 it's being used in very large production systems from what one hears 
 on this list. However it's a moving target with new releases appearing 
 frequently.
 On a positive note that's great if you want new features and bug fixes 
 - but it can also be a pain if you want a nice stable, 

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 56

2005-12-09 Thread James Horn
CM is the Cisco Call Manager and Astericks is the Asterisk Software.


-- Forwarded message --From:C F [EMAIL PROTECTED]
To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate:Fri, 9 Dec 2005 11:08:48 -0500Subject:Re: [Asterisk-Users] Phone Information
Can you please explain?Whats CM?Whats Astericks?


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Re: [Asterisk-Users] Teliax experiences

2005-12-09 Thread Chris
   I've been using them for about 3 months and haven't experienced any 
problems with them.
My only problem has been with my ISP.   I get two calls going at the same 
time and the ISP boggs down.


Regards,

Chris

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

Sent: Friday, December 09, 2005 9:16 AM
Subject: [Asterisk-Users] Teliax experiences


Howdy  - This is my first post on the list, and from what I've seen of * 
I'm
very impressed. I had a question regarding everybodys experience with 
Teliax
or Broadvoice. I setup a Teliax trunk this morning, and had calls going 
out
it in about 5 minutes(Had to get more coffee). Has anybody had any 
problems

with them, outages, issues with dids etc??


Thanks,

Rolf Brusletto
Denver, Co.

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Re: [Asterisk-Users] a few questions

2005-12-09 Thread C F
On 12/9/05, Stas Khromoy [EMAIL PROTECTED] wrote:
 we are beginning to test asterisk for our office
 one of the features of the current phone system that is very heavily
 used is overhead paging

Overhead paging can be done with asteirsk in anyway you want, you can
even do mutilple zones, all zones, or whatever you want.


 now i came accross this post
 http://forums.digium.com/viewtopic.php?t=2844highlight=features


I cound't find *anything* on that page that has to do with paging.

 that basically says it is not possible with asterisk.


Exactly where on that page???

 let's hope that i am not understanding it right, since i am new to the
 telephony.

 can any one help me out and explain it to the unfortunate ? :))

 second question is as follows:

 when you access voice mail
 the default msg is 'welcome to comedian mail'
 is there any way to get rid of this par of the greeting ?


Yeah, just rerecord that massage.


 thanks


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Re: [Asterisk-Users] a few questions

2005-12-09 Thread C F
On 12/9/05, Kerry Garrison [EMAIL PROTECTED] wrote:
 Overhead paging is totally possible, there are several articles available on
 how to do it. But you cannot have multiple zones today unless you use a sip
 device that has autoanswer.


Why can mutilple zones not be done?, why do I need
a sip device at all for the paging? any of the follwing (and I'm sure
more) will do, even for multiple zones:
* PC Sound Card
* Digum hardware
* any type of ata type gateway (SIP/h323 or whatever else that will
interface with an analog port), even one without auto answer



 Easiet way to remove that message is to replace the file with one that only
 has a split second of silence.


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Stas Khromoy
 Sent: Friday, December 09, 2005 8:21 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] a few questions

 we are beginning to test asterisk for our office one of the features of the
 current phone system that is very heavily used is overhead paging

 now i came accross this post
 http://forums.digium.com/viewtopic.php?t=2844highlight=features

 that basically says it is not possible with asterisk.

 let's hope that i am not understanding it right, since i am new to the
 telephony.

 can any one help me out and explain it to the unfortunate ? :))

 second question is as follows:

 when you access voice mail
 the default msg is 'welcome to comedian mail'
 is there any way to get rid of this par of the greeting ?


 thanks


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Re: [Asterisk-Users] Porting a phone number to a voip provider

2005-12-09 Thread Austin Denyer

On Fri,  9 Dec 2005 09:46:39 -0600
Rich Adamson  [EMAIL PROTECTED] wrote:

Just an FYI... not all cell numbers are portable.
   
   Do you have any more information on this?  I read somewhere that 
   sometimes you can port a number to a VoIP provider but not be
   able to port it back to the PSTN because not all PSTN providers
   will take numbers from VoIP providers.  Is this what you are
   talking about?
  
  No. Some cell providers do not have the capability to selectively
  let a telephone number move to another carrier/telco/itsp. They may
  have technical plans to allow/support it, but the hardware/software
  necessary to allow/support is not yet in place. Same is true with
  some smaller telcos.

Another 'game' is that the regulations do not require portability
within a company.  For example, Boost Mobile is the pre-paid arm of
Nextel.  As such, there is no regulatory requirement for portability
between Boost Mobile and Nextel, and they will not do it.  However, you
could port the number from Boost Mobile to Cingular, and then from
Cingular to Nextel.

Regards,
Ozz.


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RE: [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel

2005-12-09 Thread Colin Anderson
FWIW, we have replaced a Mitel 3300 with Asterisk - 170 users, mixed SIP/IAX
and cell (GSM gateway). The feature set that Asterisk brings to the table is
as good as or (more often) far better than the 3300 at a far, far cheaper
cost. I am doing stuff that my users quite frankly find amazing, and does
not have a direct equivalent in the Mitel world, for example, a single DID
for each user that is the extension number, the cell number, and the fax
number. I can project the extension locally, remotely through SIP/IAX or
remotely through the PSTN. Can't do that with a 3300. 

Because Asterisk exposes a standard API, and uses PSTN and VoIP standards,
the potential applications are only limited by your imagination. Some
vendors can do something along these lines, most can't. Some only dream of
the stuff that is possible with Asterisk. 

Personally, I find Asterisk way easier to configure than the 3300. Setting
up an extension in the 3300 with a DID is an exercise in frustration. Say
what you will about a monolithic extensions.conf, it's still light years
better than the 3300. Under Asterisk, I can have an extension running with a
DID in under 5 minutes. On the Mitel, takes us a couple of hours. 

As to reliability, scalability, and quality, here is the downside of
Asterisk and where the 3300 (and any other traditional PBX) wins, hands
down. It is *so* easy to create a lousy Asterisk install that just plain
won't work. It is exceedingly hard to create a stable and scalable Asterisk
install. Because there are so many variables involved that draw from many
disciplines (Linux admin, Linux coding, network admin, telephony,
general-purpose hardware hands on experience, I could go on) that you have
to have great IT judo in order to make it work the way you want. A 3300, by
contrast is dead simple: Take the system out of the box, rack it up, plug
everything in, start adding extensions (but there the frustration begins !
). If you can get Asterisk to behave in this manner in anything other than a
simple SOHO or test scenario then, sir, my hat's off to you. 

Support is also an interesting study in contrasts with either side has it's
strengths and weaknesses. An issue with a Mitel box, for example, is either
documented to death and dealt with in short order or is outright ignored
with Mitel protesting that the issue doesn't exist in the first place.
Issues with Asterisk, you are very much on your own and you better damn well
know what you are doing because formal docs are loosey-goosey and Asterisk
itself is such a moving target (although, kudos to ADP they have done an
amazing job so far). On the other hand, you post to this list and 9 times
out of 10 you will get a response, such was the case I had last week with a
vexing Dial() problem and I posted it to the list, and got an authoritative
answer that worked perfectly in 10 minutes, far faster than calling up Mitel
support and waiting in queue. This underscores the passion of the people who
use Asterisk. You will *not* find that same kind of passion when you call up
Mitel and talk to support (assuming you even get to talk to a person)

To mitigate Asterisk-specific issues in a roll out, it is best to go in baby
steps and test, test, test. The guy that does an Asterisk roll out along the
lines of: OK the Asterisk box is up, today your are using your Meridian,
tomorrow you are using Asterisk is either a genius or a fool. 


My 2c

-Original Message-
From: Dakota [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 06, 2005 6:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel

How does Asterisk compare to Nortel, NorthStar and Mitel PBX systems?
For a medium size company not growing past 75 extensions, would you
recommend Asterisk?

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Re: [Asterisk-Users] Porting a phone number to a voip provider

2005-12-09 Thread Brian Capouch

Rich Adamson wrote:

Just an FYI... not all cell numbers are portable.


Do you have any more information on this?  I read somewhere that 
sometimes you can port a number to a VoIP provider but not be able to 
port it back to the PSTN because not all PSTN providers will take 
numbers from VoIP providers.  Is this what you are talking about?



No. Some cell providers do not have the capability to selectively let
a telephone number move to another carrier/telco/itsp. They may have
technical plans to allow/support it, but the hardware/software necessary
to allow/support is not yet in place. Same is true with some smaller
telcos.

I'd also have to guess that come cell providers have probably taken a
stand that says they aren't going to do it, regardless of what the 
regulatory folks do/say.


So, the issue with number portability really starts with the company that
currently provides the cell number to you. (Sometimes they don't like
to volunter that info for obvious reasons.) If your number is portable,
then it should remain portable regardless of where you move it to, and
you should be able to move it as many times as you'd like.

If you move your number to the xyz itsp and don't like their service,
then pick another company and ask them to move the number for you. For
them to move it, however, requires an acknowledgement from the previous
company in most cases. In some (rather rare) cases, the previous company
will sometimes drag their feet or refuse to acknowledge the transfer. There
are regulatory escalation processes to address that problem, but usually
it takes a considerable amount of time to get it done.



I'd like to add, not only considerable time, but if you are dealing with 
a company whose approach to stopping churn is aggressive, then you may 
find yourself caught in a perpetual loop of problems trying to port the 
number.


For instance, I spent the better part of a year trying to port a Qwest 
8XX DID number.  The new carrier would email me to say, The transfer 
has been refused *again*, and then Qwest's folks would say, Oh, you 
had a call from someone on that number, so the balance is non-zero.


Finally I called them and paid the account 100% on the spot with a 
credit card, and then immediately called and told the new carrier to try 
the port.


Guess what?  You happened to pay the account on the exact minute a new 
billing period started, so by the time you got hold of the new carrier 
there was a balance again.


That was where I gave up, let them have the number, and am now fighting 
them about that final balance with the state PUC.  They have sent the 
balance to a collection agency and of course if they file against my 
credit record it will then be my word against theirs in that domain.


They are rude as can be once they realize you are going to give up and 
quit paying them, and they also seem to know which state regulatory 
agencies (like mine) are staffed with incompetents who won't do anything 
even if you do file a report.


Overall, huge PIA, and I'll never give Qwest a dime of my money again as 
long as I live.


B.
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Re: [Asterisk-Users] Wait for X rings before answering?

2005-12-09 Thread Robert La Ferla
I realize that it's a timeout but what's implicit in that is that 
Asterisk can't detect # of rings just the amount of time spent ringing?  
I have been looking at the reference manual on asteriskguru.com.  They 
say it's a timeout but they don't indicate the units.  Is it 
milliseconds, microseconds or seconds?



Dave Cotton wrote:

On Fri, 2005-12-09 at 11:41 -0500, Robert La Ferla wrote:
  

Derek Whitten wrote:


[incoming]
exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()
  
  
Thanks.  This will call/ring multiple extensions but what about waiting 
for X rings before going to voicemail?  How do I do that?



What do you think the 25 does?

Maybe it's a time or something.

  


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Re: [Asterisk-Users] Asterisk and Adtran TA 750 Channel Bank -- odd behavior (help!)

2005-12-09 Thread Gaurav Naik

Update:

I've determined that the problem is DTMF 9.  I cannot get to  
extension 6950 on the Nortel.  The 9 is totally skipped.  However, if  
I dial 69501, I get connected to extension 6501.  What is so special  
about DTMF 9?


Thanks


On Dec 8, 2005, at 7:40 PM, Gaurav Naik wrote:


I'm having a strange problem with an analog line connected to an  
Adtran Channel Bank.  It seems as tho, I cannot make outgoing calls  
out of the PBX the analog lines are connected to.  I'll explain...


The channel bank has a few analog lines (loop start) coming in to  
the FXO cards from a Nortel Meridien Option 81c.  I have no  
administrative/technical control over the Nortel, so I have to  
believe that the lines are configured correctly (nor are they  
willing to setup a PRI).


As a start, I terminated one of the analog lines into an analog  
phone and was able to successfully make and receive calls.  Next, I  
connected this same line to a X100P and had no problems with  
Asterisk (other than disconnect supervision).


Finally, I connected the line to the channel bank (channel 1 of the  
T1).  The T1 between TE110P and the Channel Bank is up -- no  
errors.  The TE110P is the master, with the 750 as the slave.  I  
could dial a 4-digit extension number and connect to any station on  
the Nortel switch (albeit with some echo, but no static).  However,  
dialing outside of the Nortel doesn't work.


A '9' has to be dialed in order to get an outside line.  First, I  
started with a SIP phone, and setup my dial plan accordingly.   
Stuck a few No-Ops in there and watched on the console.  Asterisk  
was correctly dialing 91800 on the FXO port, however, the  
Nortel kept ringing extension 1800. Next, I tried adding waits  
(w9w1800XXX), but I still keep getting extension 1800. I even  
tried longer and longer waits. I turned on zap debugging from the  
asterisk CLI, and could see that DTMF 9 was being sent.  Hmm.   
Again, with X100P, I could dial outside no problem -- using the  
SAME dial plan.


In order to make debugging this problem easier, I hooked up the  
analog phone to one of the FXS ports, and had Asterisk do a native  
bridge of the two ZAP lines.  Fine, so now I'm hearing the dialtone  
from the Nortel on my analog set.  I dial '9', there a slight  
silence (500ms or so) and I get a dial tone.  Correct behavior.   
Next, I dial 1800extension 1800 is ringing.  That didn't  
work.  I cycled thru about 20 different configurations of  
zapata.conf and zaptel.conf.  Still doesn't work.  I'm beginning to  
wonder if there is something wrong with the Adtran.


For a change, I decided to try incoming calls.  Asterisk gets the  
ring and answers (although it does report something about the line  
ringing in the wrong state), however, it cannot decode the DTMF  
digits the caller is dialing.  I dial 813 and Asterisk thinks I  
dialed 83. My dial plan to setup to report an invalid extension and  
gives the user another chance to enter the extension.  The second  
try always works.  (I did this about 5 times, with no problem.)   
Hmm.  So it gets the wrong DTMF digits the first time, every time.   
I checked the Digit/Response timeouts...they are set to 5 (for both  
cases).  I even tried the relaxdtmf directive, but to know avail.   
Again, I wasn't having these problems on X100P.  The only  
difference is that the X100P is running Asterisk v1.0.9 and is in  
an older machine.


I've gone through every configuration directive possible (disabling  
usecallerid, callprogress, echocancel, etc. etc. ).  I've even  
tried dialing 9 three times.  Then I pulled out a multi-meter and  
made sure that Tip and Ring weren't reversed (although that  
shouldn't make a difference).  I checked zttool for IRQ misses and  
it reported none.


The dialtone sounds fine (no hiss, pops, or static), but after  
checking the asterisk/zaptel configuration 25 times, I'm beginning  
to think its a wiring problem.  Its possible that the 750 isn't  
grounded correctly..that is probably my next step.


The relevant portions of my current configuration follow.  Any  
assistance, or hints and tips for debugging this problem are  
appreciated.


Thanks in advance,
--
Gaurav Naik
...apologizing for the long e-mail.

***
System Config
--
Dell Poweredge 1550 server
Digium TE110P (master clock)
Adtran TA 750 (slave, and two 4-port FXO cards, and 1 4-port FXS card)
Asterisk/Zapata v1.2
RHEL v4.0 Advanced Server
Polycom/Grandstream SIP Phones
a few plain old analog phones

/etc/zaptel.conf
--
span=1,0,0,esf,b8zs
fxsls=1
fxols=9
loadzone = us
defaultzone=us


/etc/asterisk/zapata.conf (this is version 34325, the simplest one).
--
signalling=fxo_ls
language=en
context=from-analog-phone
channel = 9

signalling=fxs_ls
language=en
context=from-outside
channel = 1

Adtran TA 750
--
FXO Loop Start (Time Slot 1)
TX Attenuation 0.0
RX Attenuation 0.0

All other FXO ports are disabled.  (i've tried it with all the FXO  
ports enabled as 

[Asterisk-Users] Re: ztdummy on FC4

2005-12-09 Thread Jim Duda
Tony,

I downloaded fresh versions of asterisk 1.2.1 and zaptel 1.2.1 from digium.
I now have USE_RTC in the zaptel files.  I recompiled and installed both.

I updated /etc/rc.d/init.d/zaptel and removed the other modules as I don't
have any Digium cards in my system.

lsmod results show that ztdummy is loaded.

ztdummy 7816  0
zaptel195076  5 ztdummy
crc_ccitt   6400  1 zaptel

My final issue is that ztdummy reports that it's unused (7816 0).  I can 
rmmod
ztdummy which means that * isn't using it.  Am I incorrect here?  I would 
assume
that ztdummy would have a non zero value if * was using it.

Jim

Tony Mountifield [EMAIL PROTECTED] wrote in message 
news:[EMAIL PROTECTED]
 In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] 
 wrote:
 I did cvs update -A, which brought in new files.

 make clean
 make
 make install
 make config
 /etc/rc.d/init.d/zaptel restart
 lsmod | grep ztdummy

 Ztdummy is loaded.

 /etc/rc.d/init.d/asterisk restart

 lsmod | grep ztdummy
 ztdummy 7816  0
 wcfxo  17440  0
 wcte11xp   30496  0
 wct1xxp21408  0
 wct4xxp   108352  0
 tor2   94112  0
 zaptel195076  12 
 ztdummy,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2

 I don't see the ztdummy module used, but I did last night.

 Musiconhold isn't slow like before, but certainly appears cleaner without
 ztdummy loaded.  But since ztdummy isn't even loaded, I don't think it 
 has
 any effect.

 Now you're confusing me, or confused. In what you just quoted, ztdummy is
 the first in the list, just above wcfxo, and also listed as a caller of
 the zaptel module.

 However, firstly, do you have any Digium cards in the system? If not, you
 should not have any modules loaded except for ztdummy and zaptel. tor2 and
 the ones beginning with w are not required, and should be commented out
 in your file /etc/rc.d/init.d/zaptel.

 Look in that file for the MODULES and RMODULES lines, and set them to:

 MODULES=ztdummy
 RMODULES=ztdummy

 This is assuming you don't have Digium cards installed. If you do, those
 lines should list only the required drivers, and not ztdummy. You should
 never load ztdummy if you have a Digium card.

 I'm still missing something ... Any ideas?

 I'm still concerned you can't find reference to USE_RTC. In ztdummy.c
 there should be some lines like the following, fairly near to top:

 /*
 * NOTE: (only applies to kernel 2.6)
 * If using an i386 architecture without a PC real-time clock,
 * the #define USE_RTC should be commented out.
 */
 #if defined(__i386__)
 #if LINUX_VERSION_CODE = VERSION_CODE(2,6,13)
 #define USE_RTC
 #else
 #if 0
 #define USE_RTC
 #endif
 #endif
 #endif

 And then there are #ifdef lines further down based on USE_RTC.

 If you can't find those, then you *definitely* do not have the current 
 version.

 Also, you should edit the above section to change the #if 0 to #if 1
 (the #if 0 was a panic reaction to a mistake, and unfortunately has 
 never
 been undone).

 Thanks so much for your help Tony.

 You're welcome.

 Cheers
 Tony
 -- 
 Tony Mountifield
 Work: [EMAIL PROTECTED] - http://www.softins.co.uk
 Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 56

2005-12-09 Thread C F
I'm trying to figure out why you changed the subject?
Anyhow, thirdlane makes something called asterisk PBX Manager. There
is also another tool but it doesn't work (AFAIK) over http, amongst
others that tool can:
* Show you all the contexts
* Show you all the extensions, and the DP that drive them
* Show every sinlge configuration file that asterisk, or your linux
system might use
* Allow you edit contexts
* Allow you to edit extensions
* Allow you to edit every single configuration file on your entire system
* Allow you to add contexts
* Allow you to add extensions
* Allow you to add configuration files
* Allow you to delete contexts
* Allow you to delete extensions
* Allow you to delete items within any configuration file on your entire system.

That tool is called vi


P.S. If you find anything that can do all of the above over http
please post them (but for Webmin file manager). Thank You

On 12/9/05, James Horn [EMAIL PROTECTED] wrote:
 CM is the Cisco Call Manager and Astericks is the Asterisk Software.



  -- Forwarded message --
  From: C F [EMAIL PROTECTED] 
  To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
  Date: Fri, 9 Dec 2005 11:08:48 -0500
  Subject: Re: [Asterisk-Users] Phone Information
  Can you please explain?
  Whats CM?
  Whats Astericks?




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Re: [Asterisk-Users] a few questions

2005-12-09 Thread Jerry Jones

hehe
I just installed * with a T1 span to and Adit600 with 2fxs and 1fxo

The 8 fxo ports were for zone pageing

works great

should work with any fxo device and an existing page system

On Dec 9, 2005, at 11:34 AM, C F wrote:

Overhead paging is totally possible, there are several articles  
available on
how to do it. But you cannot have multiple zones today unless you  
use a sip

device that has autoanswer.



Why can mutilple zones not be done?, why do I need
a sip device at all for the paging? any of the follwing (and I'm sure
more) will do, even for multiple zones:
* PC Sound Card
* Digum hardware
* any type of ata type gateway (SIP/h323 or whatever else that will
interface with an analog port), even one without auto answer



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Re: [Asterisk-Users] Wait for X rings before answering?

2005-12-09 Thread Derek Whitten
Robert La Ferla wrote:
 Derek Whitten wrote:
 
 [incoming]
 exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
 exten = s,2,Voicemail(myext)
 exten = s,3,Hangup()
   
 
 Thanks.  This will call/ring multiple extensions but what about waiting
 for X rings before going to voicemail?  How do I do that?
 
 
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that's what the '25' is..

so currently your dialplan goes to VM after 25 sec..

you could also declare some variables too..

PHONE1=SIP/myext
PHONE2=SIP/myext1
PHONE3=SIP/myext2


;Timeout Variables
VERYSHORTTIMEOUT=10
SHORTTIMEOUT=20
DEFTIMEOUT=30
MEDTIMEOUT=40
LONGTIMEOUT=70

[incoming]
exten = s,1,Dial(SIP/${PHONE1}${PHONE2}${PHONE3},${MEDTIMEOUT},Ttr)
exten = s,2,Voicemail(u)
exten = s,102,Voicemail(b)

-- 
.


-BEGIN GEEK CODE BLOCK-
Version: 3.1
GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w--
PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y
 --END GEEK CODE BLOCK--


.


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Re: [Asterisk-Users] Wait for X rings before answering?

2005-12-09 Thread Derek Whitten
Robert La Ferla wrote:
 Derek Whitten wrote:
 
 [incoming]
 exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
 exten = s,2,Voicemail(myext)
 exten = s,3,Hangup()
   
 
 Thanks.  This will call/ring multiple extensions but what about waiting
 for X rings before going to voicemail?  How do I do that?
 
 
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that's what the '25' is..

so currently your dialplan goes to VM after 25 sec..

you could also declare some variables too..

PHONE1=SIP/myext
PHONE2=SIP/myext1
PHONE3=SIP/myext2


;Timeout Variables
VERYSHORTTIMEOUT=10
SHORTTIMEOUT=20
DEFTIMEOUT=30
MEDTIMEOUT=40
LONGTIMEOUT=70

[incoming]
exten = s,1,Dial(SIP/${PHONE1}${PHONE2}${PHONE3},${MEDTIMEOUT},Ttr)
exten = s,2,Voicemail(u)
exten = s,102,Voicemail(b)

-- 
.


-BEGIN GEEK CODE BLOCK-
Version: 3.1
GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w--
PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y
 --END GEEK CODE BLOCK--


.


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