[asterisk-users] Newcomer Question

2007-09-20 Thread Guenther Sohler
Hallo Group!

My Name is Guenther Sohler and I registred to this group, because
I think asterisk could be interesting for me.

I have got a small server at home running linux.
It does NAT and a Firewall. There is an intranet with my home PC
and a hardware SIP phone.

This SIP phone registers at mujtelefon.cz

Now I got another account at sipgate.at

My idea is following:
I want to be reachable at both providers(numbers) at the same time.
And If I call someone, calls to austria shall use sipgate, whereas
calls to czech shall use mujtelefon.

So far, i read through the tutorial at digium.com and got the impression,
that asterisk might be able to do this and makes phoning very conveniant.

YEsterday evening i managed to get a compiled version of asterisk running
on my server. 

I just font yet have a complete idea, what is to be changed

* get asterisk running on my server as phone central(registrar)
* which firewall settings ? before/after nat
* my hardware phone registers with asterisk at my server
* which files to i have to change ? dialplan, sip.conf?


How do I achieve this ?

Thanx in advance!

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Re: [asterisk-users] Newcomer Question

2007-09-20 Thread Jan De Coster
hi,

and first off all ... welcome!
now it would be handy if you provide us with the name of your phone for 
ex 'a linksys spa942' or somthing

kr,

Jan de Coster

Guenther Sohler wrote:
 Hallo Group!

 My Name is Guenther Sohler and I registred to this group, because
 I think asterisk could be interesting for me.

 I have got a small server at home running linux.
 It does NAT and a Firewall. There is an intranet with my home PC
 and a hardware SIP phone.

 This SIP phone registers at mujtelefon.cz

 Now I got another account at sipgate.at

 My idea is following:
 I want to be reachable at both providers(numbers) at the same time.
 And If I call someone, calls to austria shall use sipgate, whereas
 calls to czech shall use mujtelefon.

 So far, i read through the tutorial at digium.com and got the impression,
 that asterisk might be able to do this and makes phoning very conveniant.

 YEsterday evening i managed to get a compiled version of asterisk running
 on my server. 

 I just font yet have a complete idea, what is to be changed

 * get asterisk running on my server as phone central(registrar)
 * which firewall settings ? before/after nat
 * my hardware phone registers with asterisk at my server
 * which files to i have to change ? dialplan, sip.conf?


 How do I achieve this ?

 Thanx in advance!

   

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Re: [asterisk-users] Newcomer Question

2007-09-20 Thread Guenther Sohler
My Phone identifies as

USer-Agent: ALL7950 02.09.23

I suppose its AllNet 7950

Hope this helps :)



 Original-Nachricht 
 Datum: Thu, 20 Sep 2007 08:36:59 +0200
 Von: Jan De Coster [EMAIL PROTECTED]
 An: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Betreff: Re: [asterisk-users] Newcomer Question

 hi,
 
 and first off all ... welcome!
 now it would be handy if you provide us with the name of your phone for 
 ex 'a linksys spa942' or somthing
 
 kr,
 
 Jan de Coster
 
 Guenther Sohler wrote:
  Hallo Group!
 
  My Name is Guenther Sohler and I registred to this group, because
  I think asterisk could be interesting for me.
 
  I have got a small server at home running linux.
  It does NAT and a Firewall. There is an intranet with my home PC
  and a hardware SIP phone.
 
  This SIP phone registers at mujtelefon.cz
 
  Now I got another account at sipgate.at
 
  My idea is following:
  I want to be reachable at both providers(numbers) at the same time.
  And If I call someone, calls to austria shall use sipgate, whereas
  calls to czech shall use mujtelefon.
 
  So far, i read through the tutorial at digium.com and got the
 impression,
  that asterisk might be able to do this and makes phoning very
 conveniant.
 
  YEsterday evening i managed to get a compiled version of asterisk
 running
  on my server. 
 
  I just font yet have a complete idea, what is to be changed
 
  * get asterisk running on my server as phone central(registrar)
  * which firewall settings ? before/after nat
  * my hardware phone registers with asterisk at my server
  * which files to i have to change ? dialplan, sip.conf?
 
 
  How do I achieve this ?
 
  Thanx in advance!
 

 
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[asterisk-users] asterisk crash and core dump: format_mp3.so

2007-09-20 Thread Vieri
Hi,

My Asterisk server process irregularly segfaults, ie.
it usually works fine (is stable) when there's low
traffic but repeatedly crashes during morning hours
when there are more calls.

I gdb'ed the core dump files and found that the
culprit may be format_mp3. So I disabled MOH today and
will see if that's the cause.
I know that mp3 files are known to cause * crashes but
what I don't understand is why it doesn't *always*
crash (ie. why doesn't it crash even when there's low
traffic? I mean, if the offending code is in the mp3
format then it should *always* crash, right?).

I'm pasting the bt below to see if someone has any
suggestions on this ML. I'd rather wait a little
before bothering the devel list... 

Core was generated by `/usr/sbin/asterisk -f -U
asterisk -G asterisk -vvvg'.
Program terminated with signal 11, Segmentation fault.
#0  0xb7b41954 in ?? () from
/usr/lib/asterisk/modules/format_mp3.so

(gdb) bt
#0  0xb7b41954 in ?? () from
/usr/lib/asterisk/modules/format_mp3.so
#1  0xb7ea6ff4 in ?? () from /lib/libc.so.6
#2  0xb7ea8120 in ?? () from /lib/libc.so.6
#3  0x001a in ?? ()
#4  0x in ?? ()

(gdb) bt full
#0  0xb7b41954 in ?? () from
/usr/lib/asterisk/modules/format_mp3.so
No symbol table info available.
#1  0xb7ea6ff4 in ?? () from /lib/libc.so.6
No symbol table info available.
#2  0xb7ea8120 in ?? () from /lib/libc.so.6
No symbol table info available.
#3  0x001a in ?? ()
No symbol table info available.
#4  0x in ?? ()
No symbol table info available.
(gdb)

(gdb) thread apply all bt
... not posting because too long here ...

Thanks,

Vieri



  

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[asterisk-users] loop detected

2007-09-20 Thread Pezhman Lali
I have an asterisk 1.4, that was working properly, 
but from  last week, without any changing in the config of asterisk, all of 
calls,fall in loop detected error.

there is two strange actions:
1-the first call after restarting the asterisk, is done successfully .
2-no packet , was sent to the destination, during this error.

plz help me
best
Mani




  

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Re: [asterisk-users] asterisk crash and core dump: format_mp3.so

2007-09-20 Thread Atis Lezdins
On Thursday 20 September 2007 11:34:44 Vieri wrote:
 My Asterisk server process irregularly segfaults, ie.
 it usually works fine (is stable) when there's low
 traffic but repeatedly crashes during morning hours
 when there are more calls.

 I gdb'ed the core dump files and found that the
 culprit may be format_mp3. So I disabled MOH today and
 will see if that's the cause.
 I know that mp3 files are known to cause * crashes but
 what I don't understand is why it doesn't *always*
 crash (ie. why doesn't it crash even when there's low
 traffic? I mean, if the offending code is in the mp3
 format then it should *always* crash, right?).

We also experienced this problem on 1.2, but i'm not sure that this is 
registered in bug database. You should check bugs.digium.com and if it's 
still valid for 1.4, you should post your backtraces there.

As solution - we refused from using format_mp3 at all - actually it has almost 
no benefits. If your MOH is in MP3s - you will get them decoded (and 
translated to necessary codec) on-the-fly for every call, so more 
performance. You can convert all your MOH to native channell formats of 
asterisk, and put all those files (one for each format/MOH combination) in 
MOH directory - asterisk will pick up one with less translation.

Regards,
Atis


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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835

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Re: [asterisk-users] asterisk crash and core dump: format_mp3.so

2007-09-20 Thread Vieri

--- Atis Lezdins [EMAIL PROTECTED] wrote:

 We also experienced this problem on 1.2, but i'm not
 sure that this is 
 registered in bug database. You should check
 bugs.digium.com and if it's 
 still valid for 1.4, you should post your backtraces
 there.

Actually, I'm using 1.2.21.1 so since 1.2 will only
receive security fixes I don't think I'll post a bug
report.

 As solution - convert all your MOH to native
 channell formats of 
 asterisk, and put all those files (one for each
 format/MOH combination) in 
 MOH directory - asterisk will pick up one with less
 translation.

Thanks Atis. Will try that out.



   

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Re: [asterisk-users] Newcomer Question

2007-09-20 Thread Anselm Martin Hoffmeister
Am Donnerstag, den 20.09.2007, 08:30 +0200 schrieb Guenther Sohler:
 Hallo Group!
 
 My Name is Guenther Sohler and I registred to this group, because
 I think asterisk could be interesting for me.

Hi Guenther, this place probably is the right one. Welcome!

 I have got a small server at home running linux.
 It does NAT and a Firewall. There is an intranet with my home PC
 and a hardware SIP phone.
 
 This SIP phone registers at mujtelefon.cz
 
 Now I got another account at sipgate.at
 
 My idea is following:
 I want to be reachable at both providers(numbers) at the same time.
 And If I call someone, calls to austria shall use sipgate, whereas
 calls to czech shall use mujtelefon.

This is possible, and it does not require too difficult steps.

First question though is wether your server has an external IP (e.g.
does the internet routing) or there is a router in between (you wrote
the server does NAT, but I already saw double- and even triple-NAT
configurations - I have to mention that). Both will work, but _not_
having NAT in between might be one trouble source less - so if you run
Asterisk on a machine with a globally valid and routable IP, you are
better off.

Your firewall should accept incoming TCP on port 5060 and incoming UDP
on all the ports RTP uses (like 1 to 2) - I rarely bother
firewalling incoming UDP packets on high ports, but you should check
that. If your phone works behind the router, the UDP requirement
probably is already sorted.

Basically, you will have to edit a few configuration files. I will give
some examples based on one of my asterisk configs, but you really should
read about those files and check wether everything is OK - I will try to
adapt to your situation, but do not blame me if I mistype or just
mis-think something.

In sip.conf, you will need to list the providers and the phones you are
going to use. I assume you will have your allnet and perhaps a few
softphones - you will probably want more than one phone some day ;-)

8 sip.conf (with example data indicated)
[general]
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
musicclass=default
language=en
; well, no idea if there are czech audio files readily available.
; I personally use language=de, of course.
dtmfmode=rfc2833
sipdebug=no

register = 1234567:[EMAIL PROTECTED]:5060/004311234567
; put your sip id (1234567), password (4321) and your
; phone number (004311234567) here
register = 123321321:[EMAIL PROTECTED]:5060/12

[sipgateat]
host=sipgate.at
secret=4321
username=1234567
fromuser=1234567
fromdomain=sipgate.at
srvlookup=yes
context=sipgateat-in
canreinvite=no
nat=no
; perhaps this needs to be set to yes
; insecure=very
; perhaps this needs to be activated - try it.
type=friend
qualify=yes

[otherprovider]
host=otherprovider.example.org
secret=abcd
username=123321321
fromuser=123321321
fromdomain=otherprovider.example.org
srvlookup=yes
context=otherprovider-in
canreinvite=no
nat=no
type=friend
qualify=yes

; stanza for SIP clients
[sip01]
mailbox=01
callerid=11
type=friend
username=sip01
secret=LaBananaLoca
; replace with the secret for your telephone, username should
; always be the same as the [stanza] name to avoid trouble
context=sipclient
host=dynamic
nat=yes

[sip02]
mailbox=01
callerid=12
type=friend
username=sip01
secret=AyayayDiosMio
context=sipclient
host=dynamic
nat=yes

8

so much for the sip.conf. This allows for two accounts with providers,
and two SIP phones (wether hard- or softphone does not matter, of
course :-) 

You will also need to setup an extensions.conf, somehow like this

8 extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
;; all of those have been like this in my conf for ages, I do not
;; even know what exactly those are good for.


; context where sipclient outgoing calls are handled
[sipclient]
; let 11 and 12 be internal numbers
exten = 11,1,Dial(SIP/sip01,60)
exten = 11,2,Hangup()
exten = 12,1,Dial(SIP/sip02,60)
exten = 12,2,Hangup()
; Outward calls. If a country prefix is present _and_ it is Austria,
; use sipgate.at
exten = _0043.,1,Dial(SIP/[EMAIL PROTECTED],60)
exten = _0043.,2,Hangup()
; Outward calls with country prefix for Czech Republic go through
; your other provider
exten = _00420.,1,Dial(SIP/[EMAIL PROTECTED],60)
exten = _00420.,2,Hangup()
; All other non-international calls go through otherprovider -
; three digit minimum here, shorter numbers treated as internal
exten = _0[1-9].,1,Dial(SIP/[EMAIL PROTECTED],60)
exten = _0[1-9].,2,Hangup()
exten = _[1-9][0-9].,1,Dial(SIP/[EMAIL PROTECTED],60)
exten = _[1-9][0-9].,2,Hangup

; add stuff for voicemail call-in here

; context for incoming calls through sipgate

[sipgateat-in]
exten = 004311234567,1,Dial(SIP/sip01SIP/sip02,60)
exten = 004311234567,2,Hangup()

[otherprovider-in]
exten = 12,1,Dial(SIP/sip01SIP/sip02,60)
exten = 12,2,Hangup()

8

This should get you started. This is a very rough example, and I 

[asterisk-users] Horrible problem - calls losing sound

2007-09-20 Thread John Hughes
We're having a horrid problem with our asterisk setup.

Sometimes calls just go dead - we can't hear what the other end is
saying.  (I think they can't hear us either).  The call doesn't hang up
until one of the callers gets bored.

Internaly we use Thomson ST2030 SIP phones.

Externaly we have 3 ISDN BRI lines (6 channels total), connected to an
Eicon Diver server card (4BRI).

We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian
system, with chan-capi 1.0.1.

Any idea what could be going wrong, where to look c?


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[asterisk-users] G.722: ast_channel_make_compatible failure

2007-09-20 Thread Ondrej Valousek
Hi all,

I have an interesting problem with Asterisk 1.4.11 - 3 SIP phones:

[phone1]
allow=g722
allow=alaw


[phone2]
allow=alaw
allow=g722


[phone3]
allow=alaw


Now, when I try to call:
1. phone1 calling phone2, I got through, using G.722 codec
2. phone2 calling phone1, I get through, using Alaw
3. phone3 calling phone1 or phone2, OK using Alaw
But:
4. phone1 calling phone3 fails:

[Sep 20 10:14:32] WARNING[30706]: chan_sip.c:2963 sip_call: No audio
format found to offer. Cancelling call to phone3


Any ideas what could be wrong?
Many thanks for any suggestion

Ondrej

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Re: [asterisk-users] Linux-HA and Asterisk

2007-09-20 Thread Dovid B
Router modell number ? On a linksys or netgear on incoming calls the wrong 
phones start ringing (unless the router is sip aware)
- Original Message - 
From: Jerry Jones [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Wednesday, September 12, 2007 3:18 PM
Subject: Re: [asterisk-users] Linux-HA and Asterisk


 How about 20+ on a Qwest DSL modem hitting our server? Works great.


 On Sep 12, 2007, at 7:23 AM, Dovid B wrote:

 Eric,
 Try 5 polycoms behind the same NAT router. Let me know when you
 grab a drink
 ;)

 - Original Message -
 From: Eric ManxPower Wieling [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Wednesday, September 12, 2007 2:43 PM
 Subject: Re: [asterisk-users] Linux-HA and Asterisk


 Polycoms work just fine behind NAT.

 Mike Clark wrote:
 Chris Mason (Lists) wrote:
 Mike Clark wrote:


 Yes, the Asterisk boxes were on private addresses. The Polycoms
 are
 also
 behind a NAT. Yes, I tried using externip in sip.conf and this
 allowed
 registration, and calls to be placed, but no audio. Unfortunately,
 Polycom does not support STUN.

 Your problem is not Linux-HA, it looks like that is fully
 functional.
 Your problem is the same one many people come across. You can't put
 Polycom phones behind NAT, it won't work.
 If you have to have the phones behind NAT, which I advise
 against, use
 Linksys which probably work better, and use a SIP aware NAT device.
 Better still, put the phones on the same network as the Asterisk
 PBX and
 say goodbye to your problems.


 Thanks Chris. Unfortunately, these solutions aren't an option. I
 guess I
 was hoping someone had found the silver bullet or some undocumented
 Asterisk feature that solved the issue. Back to the drawing board.

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Re: [asterisk-users] Horrible problem - calls losing sound

2007-09-20 Thread Péter Tóth
Hi John!

I have the same problem, the system contains 1 port Billion ISDN BRI
card, and 1 sip trunk. This is a trixbox with Asterisk
1.2.22-BRIstuffed-0.3.0-PRE-1y-i

The ISDN call is forwarded to a ring-group. The 6 sip phones are
welltech lp399 series.

If incoming the call get wrong, we can not hear the other side, but
they hear us. In my case the rtp debug shows there are no incoming rtp
packets from asterisk to SIP phone.

If somebody experienced this problem, please help US!

Thanks!


2007/9/20, John Hughes [EMAIL PROTECTED]:
 We're having a horrid problem with our asterisk setup.

 Sometimes calls just go dead - we can't hear what the other end is
 saying.  (I think they can't hear us either).  The call doesn't hang up
 until one of the callers gets bored.

 Internaly we use Thomson ST2030 SIP phones.

 Externaly we have 3 ISDN BRI lines (6 channels total), connected to an
 Eicon Diver server card (4BRI).

 We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian
 system, with chan-capi 1.0.1.

 Any idea what could be going wrong, where to look c?


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Tel.:  +36703834578
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Re: [asterisk-users] Horrible problem - calls losing sound

2007-09-20 Thread Tzafrir Cohen
On Thu, Sep 20, 2007 at 01:24:52PM +0200, Péter Tóth wrote:
 Hi John!
 
 I have the same problem, the system contains 1 port Billion ISDN BRI
 card, and 1 sip trunk. This is a trixbox with Asterisk
 1.2.22-BRIstuffed-0.3.0-PRE-1y-i
 
 The ISDN call is forwarded to a ring-group. The 6 sip phones are
 welltech lp399 series.
 
 If incoming the call get wrong, we can not hear the other side, but
 they hear us. In my case the rtp debug shows there are no incoming rtp
 packets from asterisk to SIP phone.
 
 If somebody experienced this problem, please help US!

Just as you have rtp debug, you have bri debug . 

bri debug span 1

and hope for a friendly ISDN guru on the list...

-- 
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Re: [asterisk-users] Building an RPM from Asterisk 1.4

2007-09-20 Thread Marcus Franke
On Thu, Sep 20, 2007 at 12:22:43AM +0200, Tzafrir Cohen wrote:
 
  Why is it looking for files that obviously
  don't exist?
 
 That spec uses quite a few discourged methods for rpm packages. There
 are a number of well-maintained RPM packages of Asterisk. Use one of
 them or modify one of them.
 

Do you have any examples for these spec files? 

I found a repository for installing Asterisk on Centos, but it
took a while before I discovered it. Ok, just checked the link
its for RHEL, but as Centos is just recompiled this won't matter.

Same situation for Ubuntu using the debian package format, but
I have not found a repository so far and Ubuntu delivers just 
the old 1.2 release. :)



Marcus

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Re: [asterisk-users] Problems sending more than 2 SMS with asterisk / smsq

2007-09-20 Thread randulo
On 9/19/07, Christoph Adomeit [EMAIL PROTECTED] wrote:
 Especially I do not see how I could add a wait to the dialplan
 as somebody suggested because there seems no dialplan invoked
 when I send sms.

Can you not invoke a shell script and put the sleep in there?

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[asterisk-users] Ghost calls from phones

2007-09-20 Thread Erik Wartusch
Hi all,

I have since days now a strange problem with two Thomson ST2030 phones (FMW 
3.56) on Asterisk 1.4.11. They are both in a queue (only one phone per queue 
to get the MoH played ...)

No i see often times in the CDR these:

33012
2007-09-20 14:16:52+02  s   outgoing-staff  
SIP/test1-082ca1c8  
6

0
NO ANSWER   
3
1190290612.216   


So calls to dst s and always different SIP channels. During that calls I Get 
back on the CLI a 
 -- Got SIP response 486 Busy back from 172.10.4.170
-- SIP/reservation-b5c322c8 is busy

so the phone seems to stuck in calls and is not reachable anymore

Any idea?

Kind Regards,

Erik

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Re: [asterisk-users] Horrible problem - calls losing sound

2007-09-20 Thread Patrick
On Thu, 2007-09-20 at 12:49 +0200, John Hughes wrote:
 We're having a horrid problem with our asterisk setup.
 
 Sometimes calls just go dead - we can't hear what the other end is
 saying.  (I think they can't hear us either).  The call doesn't hang up
 until one of the callers gets bored.
 
 Internaly we use Thomson ST2030 SIP phones.
 
 Externaly we have 3 ISDN BRI lines (6 channels total), connected to an
 Eicon Diver server card (4BRI).
 
 We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian
 system, with chan-capi 1.0.1.
 
 Any idea what could be going wrong, where to look c?

I'm don't know what is causing your problem but afaik you don't need
bristuff with your Eicon Diva Server card. I have several asterisk
1.2.24 boxes with 1x BRI and 4x BRI Eicon Diva Server cards and I only
use asterisk + chan_capi. This setup is very stable and works great.

Perhaps you could try rebuilding asterisk without the bristuff patch
then rebuild and install chan_capi and see if the problem goes away?

Regards,
Patrick


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Re: [asterisk-users] Horrible problem - calls losing sound

2007-09-20 Thread John Hughes
Tzafrir Cohen wrote:
 On Thu, Sep 20, 2007 at 01:24:52PM +0200, Péter Tóth wrote:
   
 Just as you have rtp debug, you have bri debug . 

 bri debug span 1

 and hope for a friendly ISDN guru on the list...
   
Does nothing for me - my isdn is connected via chan-capi.




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Re: [asterisk-users] Building an RPM from Asterisk 1.4

2007-09-20 Thread Tzafrir Cohen
On Thu, Sep 20, 2007 at 02:25:19PM +0200, Marcus Franke wrote:

 Same situation for Ubuntu using the debian package format, but
 I have not found a repository so far and Ubuntu delivers just 
 the old 1.2 release. :)

Ubuntu packages of Asterisk are slightly modified Debian ones.
As for the Debian ones:
http://packages.debian.org/asterisk

As you can see there, the Unstable packages are currently at 1.4.11 .
Not that bad. There are certain petty things that delay that package 
from getting into Testing right now.

Also in that page you an find the sources of the package (tarball, dif
and a small .dsc file).

http://packages.debian.org/source/sid/asterisk will also give you link
to the subversion where the sources (or rather: the diff) are
maintained.

And then there's a build server where new and untested packages are
built. Including backports for various Debian and Ubuntu distributions.
http://buildserver.net/

-- 
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Re: [asterisk-users] Problems sending more than 2 SMS with asterisk / smsq

2007-09-20 Thread Anselm Martin Hoffmeister
Am Mittwoch, den 19.09.2007, 15:25 +0200 schrieb Christoph Adomeit:
 Hi there, 
 
 I experience the same problem here with asterisk 1.2.24 on
 an E1 Line, only 2 of 3 sms are sent, this happens always and 
 is reproducable.
 
 Did someone find out more about the problem ?
 
 Especially I do not see how I could add a wait to the dialplan
 as somebody suggested because there seems no dialplan invoked
 when I send sms.
 
 I use:
 
 smsq -d 017xxx -m TEST1 --motx-channel=Zap/g1/0193010
 (Germen Telekom Message Center)
 
 How could I invoke smsq differently to use an own context
 of the dialplan ?

I think I have been in error there. The wait occurs on incoming calls,
like my Gigaset calling in to Asterisk, between answering the line:

exten = _0193010.,1,Answer()
exten = _0193010.,2,Wait(2)
exten = _0193010.,3,SMS(blahfasel)

Same for _incoming_ messages from the Telco SMSC.

I do not immediately understand where I could have inserted a Wait() for
outgoing SMS, especially as that SMS() seems to open the line itself. I
would have to investigate, but do not have the equipment right here at
the moment, sorry.

BR
Anselm


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Re: [asterisk-users] Horrible problem - calls losing sound

2007-09-20 Thread John Hughes
Tzafrir Cohen wrote:
 On Thu, Sep 20, 2007 at 01:24:52PM +0200, Péter Tóth wrote:
   
 Hi John!

 I have the same problem, the system contains 1 port Billion ISDN BRI
 card, and 1 sip trunk. This is a trixbox with Asterisk
 1.2.22-BRIstuffed-0.3.0-PRE-1y-i

 The ISDN call is forwarded to a ring-group. The 6 sip phones are
 welltech lp399 series.

 If incoming the call get wrong, we can not hear the other side, but
 they hear us. In my case the rtp debug shows there are no incoming rtp
 packets from asterisk to SIP phone.

 If somebody experienced this problem, please help US!
 
 Just as you have rtp debug, you have bri debug . 

 bri debug span 1

 and hope for a friendly ISDN guru on the list...
   
Ah, I can get the same (excessive!) info by

set verbose 8
capi debug


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Re: [asterisk-users] Horrible problem - calls losing sound

2007-09-20 Thread John Hughes
Patrick wrote:
 On Thu, 2007-09-20 at 12:49 +0200, John Hughes wrote:
   
 We're having a horrid problem with our asterisk setup.

 Sometimes calls just go dead - we can't hear what the other end is
 saying.  (I think they can't hear us either).  The call doesn't hang up
 until one of the callers gets bored.

 Internaly we use Thomson ST2030 SIP phones.

 Externaly we have 3 ISDN BRI lines (6 channels total), connected to an
 Eicon Diver server card (4BRI).

 We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian
 system, with chan-capi 1.0.1.

 Any idea what could be going wrong, where to look c?
 

 I'm don't know what is causing your problem but afaik you don't need
 bristuff with your Eicon Diva Server card. I have several asterisk
 1.2.24 boxes with 1x BRI and 4x BRI Eicon Diva Server cards and I only
 use asterisk + chan_capi. This setup is very stable and works great.

 Perhaps you could try rebuilding asterisk without the bristuff patch
 then rebuild and install chan_capi and see if the problem goes away?
   
Yeah, I was being lazy, just using the asterisk from the Debian
repositories.

Guess I'll try with my own build.


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Re: [asterisk-users] Hints / State change on outgoing calls

2007-09-20 Thread James FitzGibbon
On 9/19/07, Alex Epshteyn [EMAIL PROTECTED] wrote:


 Also, Asterisk restart results in all the watchers being lost. Is there a
 way to force the phone to subscribe to notifications after restart (short
 of
 rebooting it) and is it phone specific?


Usually resubscribe-interval for extensions is client controlled, much like
SIP re-register interval.  Just make sure it's in between the min and max
registration times as displayed in the output of 'sip show settings',
otherwise you can run into problems where the phone thinks that the
subscription is valid for longer than Asterisk does.

-- 
j.
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Re: [asterisk-users] Building an RPM from Asterisk 1.4

2007-09-20 Thread James FitzGibbon
On 9/20/07, Marcus Franke [EMAIL PROTECTED] wrote:

 Do you have any examples for these spec files?

 I found a repository for installing Asterisk on Centos, but it
 took a while before I discovered it. Ok, just checked the link
 its for RHEL, but as Centos is just recompiled this won't matter.

 Same situation for Ubuntu using the debian package format, but
 I have not found a repository so far and Ubuntu delivers just
 the old 1.2 release. :)


www.atrpms.net has pretty solid RPMs, and you can grab the SRPMS in order to
get the spec file (either install the SRPM or use rpm2cpio to convert the
package and extract the specfile manually).  This is where I get my libpri
and zaptel RPMs from (though I still build * from source, as the RPM
compilation options they use are not to my liking).

There is a book called Maximum RPM.  The dead tree version is now pretty
out of date with respect to the latest version of RPM (though still a good
introduction if you've never built a package).  I believe there was a
slightly more up-to-date online version, but it still had some gaps the last
time I looked.

The best way to learn seems to be to examine good examples and then build
your own package using their techniques.




-- 
j.
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Re: [asterisk-users] Horrible problem - calls losing sound

2007-09-20 Thread Tzafrir Cohen
On Thu, Sep 20, 2007 at 02:59:49PM +0200, John Hughes wrote:

 Yeah, I was being lazy, just using the asterisk from the Debian
 repositories.

Bristuff includes its own ancient version of chan_capi. Debian removes
it.

-- 
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Re: [asterisk-users] g729 on 1.4.10.1

2007-09-20 Thread Luke Groeneveld
On Thu, 20 Sep 2007 02:15:11 am Scott Moseman wrote:
 I'm getting frustrated simply trying to get this g729 working.

For what it is worth, I had a similar issue to you, and managed to get g729 
working by installing the binary files from http://asterisk.hosting.lv

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Re: [asterisk-users] g729 on 1.4.10.1

2007-09-20 Thread Scott Moseman
I'm trying a simple Echo test and it's failing for g729...

exten = 1267,1,Answer()
exten = 1267,2,Echo()

Test #1 (failure)
gateway33 codecs g729a, g729b

[gateway33]
type=friend
host=gateway33
context=default-inbound
disallow=all
allow=g729

gateway33 INVITE = g729b
Asterisk 200 OK = no media
Asterisk sends (one way) g711u RTP

Test #2 (failure)
gateway33 codecs g729a

[gateway33]
type=friend
host=gateway33
context=default-inbound
disallow=all
allow=g729

gateway33 INVITE = g729a
Asterisk 200 OK = no media
gateway33 ends the session
gateway33 INVITE = g729a
Asterisk 200 OK = no media
gateway33 ends the session
...

Test #3 (success)
gateway33 codec g729a, g729b, g711u

[gateway33]
type=friend
host=gateway33
context=default-inbound
disallow=all
allow=ulaw
allow=g729

gateway33 INVITE = g729b, g711u
Asterisk 200 OK = g711u
Asterisk sends/receives g711u RTP

Does any of this point to a specific problem?  I even have a licensed
g729 channel.

CLI show g729
0/0 encoders/decoders of 1 licensed channels are currently in use

What information can I provide to help troubleshoot?  This is making no sense.

When I setup my desk phone to use G729 and make the test call directly
(bypassing the gateway), the call completes fine and media is sent
using G729 successfully.  I'm not sure why it would work any
differently from a Cisco gateway?

The only difference that I'm aware of is that my phone (Polycom 430)
seemed to ask for G729, while the gateway was either G729a or G729b
specifically.  In the instance of my phone, Asterisk came back with
G729a in the 200 OK message.

Thanks,
Scott

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Re: [asterisk-users] Building an RPM from Asterisk 1.4

2007-09-20 Thread Jared Smith
On Thu, 2007-09-20 at 09:01 -0400, James FitzGibbon wrote:
 www.atrpms.net has pretty solid RPMs, and you can grab the SRPMS in
 order to get the spec file (either install the SRPM or use rpm2cpio to
 convert the package and extract the specfile manually). 

I've looked at a lot of Asterisk spec files (including the ones at
atrpms.net, and the ones at http://www.laimbock.com/asterisk/), but so
far my favorites are the ones done by Jeff C. Ollie at
http://repo.ocjtech.us/misc/fedora/7/SRPMS/.  They're about the most
thorough I've seen.

 There is a book called Maximum RPM.  The dead tree version is now
 pretty out of date with respect to the latest version of RPM (though
 still a good introduction if you've never built a package).

The Red Hat RPM Guide is online at
http://docs.fedoraproject.org/drafts/rpm-guide-en/, and it's fairly
good.  I also highly recommend the guide from Guru Labs, available at
http://www.gurulabs.com/GURULABS-RPM-LAB/GURULABS-RPM-GUIDE-v1.0.PDF


Personally, I agree with the original poster... the spec file that's
currently in the Asterisk source is less than useful, and should
probably be replaced.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Softphone RTP Session Start-up Delay

2007-09-20 Thread Kutman.DK
Thanks for the reply.  I actually found the problem by inserting print 
statements in the code and checking which operations took the most time.  I 
found that one line was taking a long time to run.  The line was the following
 
RTPManager.addtarget(destAddress)
 
After googling this for a while, I found a solution on a JMF site which 
recommended a minor change in the way that the destAddress is obtained.  After 
making this change the delay was removed.
 
Thanks,
 
Denis

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James FitzGibbon
Sent: Wednesday, September 19, 2007 4:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Softphone RTP Session Start-up Delay


On 9/19/07, [EMAIL PROTECTED]  [EMAIL PROTECTED] wrote: 


 here because we are actually specifying the IP Address of the Asterisk server, 
but I am willing to try anything to fix this problem.  The two user pc's are 
setup on workgroups, so I do not believe that there is a domain available that 
can be entered in the hosts file.  Could the DNS still be the issue?  If not, 
would anyone be able to suggest any other possible problems that may be causing 
this delay. 


It wasn't the same magnitude (more like 4 seconds for me), but I had an issue 
where the default eyeBeam (the commercial version of X-Lite) install was 
imposing a delay when a call first came in while it attempted to contact a 
non-existent STUN server.  When I removed the STUN server setting, call setup 
was immediate. 

Might be worth looking at.  Have you done a trace on the PC where the softphone 
is running (without a filter) to see what network packets are flying at the 
time the call setup happens?


-- 
j. 
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[asterisk-users] Outgoing SIP packets out of order?

2007-09-20 Thread Jason Martin
Hello, I've been looking at some SIP packet dumps captured with tcpdump on the 
PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something 
strange, at least to me. All of the SIP packets going out from our Asterisk 
PBX to either of our 2 VoIP providers are consistently 50% out of order. In 
addition, if I use Wireshark's voip call player, the outgoing side of the 
call stutters and is delayed compared to the incoming side of the call. 

Is this normal? Why would the PBX be sending packets already out of order?

Thanks!
-- 
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
Office: 888-865-0065 Ext. 202
Mobile: (585) 721-8679


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Re: [asterisk-users] Problems with Asterisk behind a firewall

2007-09-20 Thread Dovid B
Did you set externip= ?

- Original Message - 
From: Christian [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Wednesday, September 12, 2007 3:23 PM
Subject: [asterisk-users] Problems with Asterisk behind a firewall


 Hi all,
 I have set up Asterisk and I am able to register with my SIP provider and 
 receive calls.
 When I try to register with Asterisk from outside I can place calls but 
 tthe other person can't hear me.
 Have opened port 5060 UDP as well as port 1 to 2 UDP. Any ideas?
 Thanks,
 Christian

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[asterisk-users] The device state is still 'Not in Use' ... check UPGRADE.txt

2007-09-20 Thread John Hughes
Or, in full:

[Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The
device state of this queue member, SIP/612, is still 'Not in Use' when
it probably should not be! Please check UPGRADE.txt for correct
configuration settings.

So, what do I check in UPGRADE.txt?

This is with Asterisk 1.4.11

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Re: [asterisk-users] The device state is still 'Not in Use' ... check UPGRADE.txt

2007-09-20 Thread John Hughes
John Hughes wrote:
 Or, in full:

 [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The
 device state of this queue member, SIP/612, is still 'Not in Use' when
 it probably should not be! Please check UPGRADE.txt for correct
 configuration settings.

 So, what do I check in UPGRADE.txt?

 This is with Asterisk 1.4.11
   
Ah, I'm so dumb - it's this bit:

* Queues depend on the channel driver reporting the proper state
  for each member of the queue. To get proper signalling on
  queue members that use the SIP channel driver, you need to
  enable a call limit (could be set to a high value so it
  is not put into action) and also make sure that both inbound
  and outbound calls are accounted for.

  Example:

   [general]
   limitonpeer = yes

   [peername]
   type=friend
   call-limit=10

Could have sworn I had the call-limit, but in fact it was missing for
the extension I was testing from!


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[asterisk-users] Announcing: Click-to-Call with VIDEO

2007-09-20 Thread C. Savinovich

Dear All:

Just as the name suggests, and evolving from regular Click-to-Call,
Click-to-Call WITH VIDEO provides web sites with the ability to engage
their visitors with a live video agent (plus the phone call).  All with just
a click of a button placed on the customer's web site.  Please visit us at
www.videoreps.net.

The video technology integrates with Asterisk, and it also allows the
agent to video-conference up to 10 callers at the same time, making it an
ideal tool for online seminars, demos, shows, tech support, etc, etc.

Click-to-call WITH VIDEO can be used in:

   1) Customer service mode (web click-me buttons)
   2) Conference mode (meetings of up to 10 participants)
   3) Video-Call mode (enter user destination number and video-connect
to him/her)

It is an ideal add-on for pbx vendors who want to add video services to
their pbxs.

Best of all, it is free for now!! (video only)...  Here is the deal:
Free for one month, no commitments.  Try, test it, call me.  After a month
you can decide if you want to keep any of our plans:  Video-Call starts at
$12.95 per month, and click-to-call with video at $29.95. And yes, there is
20% monthly commission if you include the service in the pbxs you are
selling.

Here is how it works: Register online and you will receive an email with
the confirmation link.  With your new username you will be able to enter the
operator's work area.  Please fill your profile with your web site's ip (so
that the buttons work), your destination phone (so that the phone call
works) and a meeting room username (so that the videocall works).  Go to the
demo buttons page and click on anyone you like to download the button and
kit with a couple lines of php code you need to place the button in your web
page.  Next, assuming you have a camera connected, click on GoOnLine and you
will be online!! (don't eat at your desk!).
 
I appreciate your kind input on this web site.  I am at my desk most of
the time (logged in as a video agent), and I will be glad to be available
when you click on our web site button, or when you enter my number
19176135931 on the meeting room for quick-connect videophone.  However,
hyper as I am, I would prefer to coordinate a demo time with anyone who
wishes a demo session.

Best Regards
Christian Savinovich
VideoReps.net

Note: Being an ActiveX component, please use internet explorer.


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Re: [asterisk-users] AMI extension states

2007-09-20 Thread Anthony Francis
Here is what I use.

sub devstate2str($)
{
  #func name stolen directly from asterisk
  #takes int devstate and returns string val
  my $ids  = shift;
  my $devstatestring   = {};
  $devstatestring-{0} = Unknown; #0 AST_DEVICE_UNKNOWN | Valid, 
but unknown state
  $devstatestring-{1} = Not in use;  #1 AST_DEVICE_NOT_INUSE | Not used
  $devstatestring-{2} = In use;  #2 AST_DEVICE IN USE | In use
  $devstatestring-{3} = Busy;#3 AST_DEVICE_BUSY | Busy
  $devstatestring-{4} = Invalid; #4 AST_DEVICE_INVALID | Invalid 
- not known to Asterisk
  $devstatestring-{5} = Unavailable; #5 AST_DEVICE_UNAVAILABLE | 
Unavailable (not registred)
  $devstatestring-{6} = Ringing; #6 AST_DEVICE_RINGING | Ring, 
ring, ring
  return $devstatestring-{$ids};
}

LIke the comment says, I stole it direct from the source. A code of 16 
may be a typo in which case you have found a bug.

Philipp Kempgen wrote:
 Hi,

 Is there a list of all the extension states as sent by the
 manager interface? (I know I could look them up in the source
 but that involves some backtracing.)

 The ones I know are:

 -1: no hint for the extension
  0: registered  idle
  1: busy
  4: unreachable, not registered
  8: ringing

 I've recently seen 16 (== hold?) but can't find that value
 documented anywhere.

 Regards,
   Philipp Kempgen

   

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] AMI extension states

2007-09-20 Thread Anthony Francis
Oh one other note, when asking questions such as this, it is really wise 
to include which version # you are using.

Philipp Kempgen wrote:
 Hi,

 Is there a list of all the extension states as sent by the
 manager interface? (I know I could look them up in the source
 but that involves some backtracing.)

 The ones I know are:

 -1: no hint for the extension
  0: registered  idle
  1: busy
  4: unreachable, not registered
  8: ringing

 I've recently seen 16 (== hold?) but can't find that value
 documented anywhere.

 Regards,
   Philipp Kempgen

   

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] The device state is still 'Not in Use' ... check UPGRADE.txt

2007-09-20 Thread Mark Michelson
John Hughes wrote:
 Or, in full:

 [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The
 device state of this queue member, SIP/612, is still 'Not in Use' when
 it probably should not be! Please check UPGRADE.txt for correct
 configuration settings.

 So, what do I check in UPGRADE.txt?

 This is with Asterisk 1.4.11

   
Here's what you're looking for:


* Queues depend on the channel driver reporting the proper state
  for each member of the queue. To get proper signalling on
  queue members that use the SIP channel driver, you need to
  enable a call limit (could be set to a high value so it
  is not put into action) and also make sure that both inbound
  and outbound calls are accounted for.

  Example:

   [general]
   limitonpeer = yes

   [peername]
   type=friend
   call-limit=10



Mark Michelson

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Re: [asterisk-users] g729 on 1.4.10.1

2007-09-20 Thread Scott Moseman
On 9/20/07, Luke Groeneveld [EMAIL PROTECTED] wrote:

  I'm getting frustrated simply trying to get this g729 working.

 For what it is worth, I had a similar issue to you, and managed to get
 g729 working by installing the binary files from http://asterisk.hosting.lv


Thanks for the suggestion.  Looks like I'm having the same problem, though.
What's odd is that I can make phone to phone G729 calls through Asterisk,
but G729 calls from my gateway do not work.

Thanks,
Scott

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Re: [asterisk-users] Problem with asterisk-perl-0.08 and Asterisk = 1.2.20

2007-09-20 Thread Benoît Mérouze
Tzafrir Cohen a écrit :
 On Wed, Sep 19, 2007 at 12:25:35PM +0200, Benoît Mérouze wrote:
 Is there any reason this can be fixed in the asterisk-perl-0.10
 (not yet included in Trixbox)? Or is this more an issue from
 Asterisk (since Asterisk 1.2.19 or 1.2.20)?

 Why not give it a shot?

 Install asterisk-perl's modules to a non-standard path and add to
 your script a 'use' directive to use that path first.


I've tried asterisk-perl-0.10 on a development server.  And the result
is the same.

Hence I guess something has changed between Asterisk-1.2.18 and later
revisions that no more works with asterisk-perl when get_variable is
called (calling Asterisk AGI command GET VARIABLE).

I've also discovered the intersting function get_full_variable in
asterisk-perl-0.10.  But the issue is the same with Asterisk  1.2.18.

Regards,
Benoit


-- 
Benoît Mérouze - Telecom Software Developer - IPercom
[EMAIL PROTECTED]
Those who would give up Essential Liberty to purchase a little
  Temporary Safety, deserve neither Liberty nor Safety.
Benjamin Franklin


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[asterisk-users] Queue Question

2007-09-20 Thread Jeremy Mann
I'm curious if anyone has implemented the following:

Need to setup an on-call queue, that activates after 5PM and de-activates at 
8AM, also that activates/deactivates on demand(I'm thinking a feature code 
here).  The agents need to log in via cell phones, and when calls come in 
from outside to the asterisk system, it'll need to call the cell phone agents 
that are active.

I'm thinking that it's a simple SQL query, to update the agents status and 
number, and that asterisk will do a lookup and append that to the ZAP channel 
to dial, but interested in any logic someone might be able to come up with for 
the dialplan.




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any disclosure, copying, printing, or use of this information is strictly 
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Re: [asterisk-users] AMI extension states

2007-09-20 Thread Philipp Kempgen
Anthony Francis wrote:

 Here is what I use.
 
 sub devstate2str($)
 {
   #func name stolen directly from asterisk
   #takes int devstate and returns string val
   my $ids  = shift;
   my $devstatestring   = {};
   $devstatestring-{0} = Unknown; #0 AST_DEVICE_UNKNOWN | Valid, 
 but unknown state
   $devstatestring-{1} = Not in use;  #1 AST_DEVICE_NOT_INUSE | Not used
   $devstatestring-{2} = In use;  #2 AST_DEVICE IN USE | In use
   $devstatestring-{3} = Busy;#3 AST_DEVICE_BUSY | Busy
   $devstatestring-{4} = Invalid; #4 AST_DEVICE_INVALID | Invalid 
 - not known to Asterisk
   $devstatestring-{5} = Unavailable; #5 AST_DEVICE_UNAVAILABLE | 
 Unavailable (not registred)
   $devstatestring-{6} = Ringing; #6 AST_DEVICE_RINGING | Ring, 
 ring, ring
   return $devstatestring-{$ids};
 }
 
 LIke the comment says, I stole it direct from the source. A code of 16 
 may be a typo in which case you have found a bug.
 
 Philipp Kempgen wrote:
 Hi,

 Is there a list of all the extension states as sent by the
 manager interface? (I know I could look them up in the source
 but that involves some backtracing.)

 The ones I know are:

 -1: no hint for the extension
  0: registered  idle
  1: busy
  4: unreachable, not registered
  8: ringing

 I've recently seen 16 (== hold?) but can't find that value
 documented anywhere.

Thanks, but I think device states are different from extension states.
(Please correct me If I'm mistaken.)

Device states from devicestate.h:
---cut---
/*! Device is valid but channel didn't know state */
#define AST_DEVICE_UNKNOWN  0
/*! Device is not used */
#define AST_DEVICE_NOT_INUSE1
/*! Device is in use */
#define AST_DEVICE_INUSE2
/*! Device is busy */
#define AST_DEVICE_BUSY 3
/*! Device is invalid */
#define AST_DEVICE_INVALID  4
/*! Device is unavailable */
#define AST_DEVICE_UNAVAILABLE  5
/*! Device is ringing */
#define AST_DEVICE_RINGING  6
/*! Device is ringing *and* in use */
#define AST_DEVICE_RINGINUSE7
/*! Device is on hold */
#define AST_DEVICE_ONHOLD   8
---cut---

I was looking for the extension states as sent by the manager interface.
Extension states from pbx.h:
---cut---
/*! \brief Extension states */
enum ast_extension_states {
AST_EXTENSION_REMOVED = -2, /*! Extension removed */
AST_EXTENSION_DEACTIVATED = -1, /*! Extension hint removed */
AST_EXTENSION_NOT_INUSE = 0,/*! No device INUSE or BUSY  */
AST_EXTENSION_INUSE = 1  0,   /*! One or more devices INUSE */
AST_EXTENSION_BUSY = 1  1,/*! All devices BUSY */
AST_EXTENSION_UNAVAILABLE = 1  2, /*! All devices 
UNAVAILABLE/UNREGISTERED */
AST_EXTENSION_RINGING = 1  3, /*! All devices RINGING */
AST_EXTENSION_ONHOLD = 1  4,  /*! All devices ONHOLD */
};
---cut---

14 == 16, so probably this is the value I observed.
But there some mangling occurs in ast_extension_state2() in pbx.c like
if (inuse  ring)
return (AST_EXTENSION_INUSE | AST_EXTENSION_RINGING);
which would return
10 | 13 == 1|8 == 9
So the extension state is some kind of a bit mask.

I think I have found what I was looking for. :)

Thanks,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? - http://www.das-asterisk-buch.de
  My pick of the month: rfc 2822 3.6.5

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] AMI extension states

2007-09-20 Thread Philipp Kempgen
Anthony Francis wrote:

 Oh one other note, when asking questions such as this, it is really wise 
 to include which version # you are using.

Right. Sorry.
1.4.11 (for the archives)


Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? - http://www.das-asterisk-buch.de
  My pick of the month: rfc 2822 3.6.5

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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[asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards

2007-09-20 Thread Brian Alexander
I am trying to connect two machines to each other with an T1 crossover
cable. The first machine has two TE120P cards - one connecting to the telco
on an ISDN PRI. The second to a crossover T1 cable to a second machine which
has one TE120P card.

  Telco -cA- Machine1 -cB- Machine2

  Machine1: Two TE120P cards
  Machine2: One TE120P card
  cA: Standard T1 Cable
  cB: Crossover T1 Cable

Configuration files are included at the end of this message.

I have used both 'cat /proc/interupts' and 'lspci -vb' to verify that the
cards do not have IRQ conflicts. Machine2 can be plugged into the telco pri
and works fine. Machine1 works on the telco PRI so long as I have removed
the configuration for the second span (the one on the crossover). I can
leave the card for the second span in - so long as it is not configured.

Machine 1 CLI Notices

If both cards are configured and connected. Zttool reliably reports no
alarms. Asterisk appears to start without errors at verbosity three. Very
quickly after it starts Asterisk starts reporting

  WARNING[28899] chan_zap.c:6668 handle_init_event: Detected alarm on
channel NN: Red Alarm
  WARNING[28899] chan_zap.c:1464 zt_disable_ec: Unable to disable echo
cancellation on channel NN

It will output these for all the channels on the second span (NN ranges from
25 to 47). It then will report

  NOTICE[28898] chan_zap.c:8460 pri_dchannel: PRI got event: Alarm (4) on
Primary D-channel of span 2
  WARNING[28898] chan_zap.c:2393 pri_find_dchan: No D-channels available!
Using primary channel 48 as D-channel anyway!

This is immediately followed by a notice that the alarms were cleared on
each of the channels of span 2, including the data-channel.

  NOTICE [29899]: chan_zap.c:6661 handle_init_event: Alarm cleared on
channel NN
  NOTICE [29898]: chan_zap.c:8460 pri_dchannel: PRI got event: No more alarm
(5) on Primary D-channel of span 2

This is then followed by the first error

  ERROR [29898]: chan_zap.c:8174 zt_pri_error: !! Got S-frame while link
down

I then get constant notices that Primary D-Channel on span 2 up. This will
be periodically broken up by a repeat of the warnings, notices and errors I
describe above. However, now the problems occur for all of the channels of
both spans.

Machine 1 CLI Notices

Machine 1 experiences almost the same behavior on its span. The only
differance I am noticing is that instead of the S-frame error I get the
following notice:

  chan_zap.c:8457 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
D-channel of span 1

Zap Restart Fails

On either machine if I attempt a 'zap restart' I receive

  WARNING[30686]: chan_zap.c:903 zt_open: Unable to specify channel 1:
Device or resource busy
  ERROR[30686]: chan_zap.c:7160 mkintf: Unable to open channel 1: Device or
resource busy
here = 0, tmp-channel = 1, channel = 1
  ERROR[30686]: chan_zap.c:10467 build_channels: Unable to register channel
'1-23'
  WARNING[30686]: chan_zap.c:9764 zap_restart: Reload channels from zap
config failed!




I have not attempted to connect two Asterisk boxes though a T1 crossover
before so I am stumped. I have included my zaptel.conf and zapata.conf files
below. I will certainly appreciate any help you can give.

Thanks,
-Brian




Machine1
=

zaptel.conf
---
defaultzone=us
loadzone=us
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

span=2,0,0,esf,b8zs
bchan=25-47
dchan=48

zapata.conf
---
[trunkgroups]
[channels]
group=1
context=fromtelco
signalling=pri_cpe
switchtype=national
channel=1-23

group=1
context=frommachine2
signalling=pri_net
switchtype=national
channel=25-47

Machine2
=

zaptel.conf
---
defaultzone=us
loadzone=us
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

zapata.conf
---
[trunkgroups]
[channels]
group=1
context=frommachine1
signalling=pri_cpe
switchtype=national
channel=1-23
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Re: [asterisk-users] AMI extension states

2007-09-20 Thread Anthony Francis


Philipp Kempgen wrote:
 Anthony Francis wrote:

   
 Here is what I use.

 sub devstate2str($)
 {
   #func name stolen directly from asterisk
   #takes int devstate and returns string val
   my $ids  = shift;
   my $devstatestring   = {};
   $devstatestring-{0} = Unknown; #0 AST_DEVICE_UNKNOWN | Valid, 
 but unknown state
   $devstatestring-{1} = Not in use;  #1 AST_DEVICE_NOT_INUSE | Not used
   $devstatestring-{2} = In use;  #2 AST_DEVICE IN USE | In use
   $devstatestring-{3} = Busy;#3 AST_DEVICE_BUSY | Busy
   $devstatestring-{4} = Invalid; #4 AST_DEVICE_INVALID | Invalid 
 - not known to Asterisk
   $devstatestring-{5} = Unavailable; #5 AST_DEVICE_UNAVAILABLE | 
 Unavailable (not registred)
   $devstatestring-{6} = Ringing; #6 AST_DEVICE_RINGING | Ring, 
 ring, ring
   return $devstatestring-{$ids};
 }

 LIke the comment says, I stole it direct from the source. A code of 16 
 may be a typo in which case you have found a bug.

 Philipp Kempgen wrote:
 
 Hi,

 Is there a list of all the extension states as sent by the
 manager interface? (I know I could look them up in the source
 but that involves some backtracing.)

 The ones I know are:

 -1: no hint for the extension
  0: registered  idle
  1: busy
  4: unreachable, not registered
  8: ringing

 I've recently seen 16 (== hold?) but can't find that value
 documented anywhere.
   

 Thanks, but I think device states are different from extension states.
 (Please correct me If I'm mistaken.)

 Device states from devicestate.h:
 ---cut---
 /*! Device is valid but channel didn't know state */
 #define AST_DEVICE_UNKNOWN0
 /*! Device is not used */
 #define AST_DEVICE_NOT_INUSE  1
 /*! Device is in use */
 #define AST_DEVICE_INUSE  2
 /*! Device is busy */
 #define AST_DEVICE_BUSY   3
 /*! Device is invalid */
 #define AST_DEVICE_INVALID4
 /*! Device is unavailable */
 #define AST_DEVICE_UNAVAILABLE5
 /*! Device is ringing */
 #define AST_DEVICE_RINGING6
 /*! Device is ringing *and* in use */
 #define AST_DEVICE_RINGINUSE  7
 /*! Device is on hold */
 #define AST_DEVICE_ONHOLD 8
 ---cut---

 I was looking for the extension states as sent by the manager interface.
 Extension states from pbx.h:
 ---cut---
 /*! \brief Extension states */
 enum ast_extension_states {
   AST_EXTENSION_REMOVED = -2, /*! Extension removed */
   AST_EXTENSION_DEACTIVATED = -1, /*! Extension hint removed */
   AST_EXTENSION_NOT_INUSE = 0,/*! No device INUSE or BUSY  */
   AST_EXTENSION_INUSE = 1  0,   /*! One or more devices INUSE */
   AST_EXTENSION_BUSY = 1  1,/*! All devices BUSY */
   AST_EXTENSION_UNAVAILABLE = 1  2, /*! All devices 
 UNAVAILABLE/UNREGISTERED */
   AST_EXTENSION_RINGING = 1  3, /*! All devices RINGING */
   AST_EXTENSION_ONHOLD = 1  4,  /*! All devices ONHOLD */
 };
 ---cut---

 14 == 16, so probably this is the value I observed.
 But there some mangling occurs in ast_extension_state2() in pbx.c like
 if (inuse  ring)
   return (AST_EXTENSION_INUSE | AST_EXTENSION_RINGING);
 which would return
 10 | 13 == 1|8 == 9
 So the extension state is some kind of a bit mask.

 I think I have found what I was looking for. :)

 Thanks,
   Philipp Kempgen

   
Umm, that is the code I use in my manager interface monitoring software, 
and it works. of course I am using 1.2.18 and perhaps you are using a 
different version.

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] AMI extension states

2007-09-20 Thread Anthony Francis

Philipp Kempgen wrote:
 Anthony Francis wrote:

   
 Oh one other note, when asking questions such as this, it is really wise 
 to include which version # you are using.
 

 Right. Sorry.
 1.4.11 (for the archives)


 Regards,
   Philipp Kempgen

   
That is what I thought, makes what I said invalid for you, but valid in 1.2

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] OT: Samsung Sprint CDMAoIP

2007-09-20 Thread Eric Chamberlain
The device is a femtocell device; I would bet that they keep it in a format 
that works with their existing equipment, rather than use SIP.  

The device is also licensed for a specific frequency that is owned by the 
carrier, you wouldn't be able to use this device for any other purpose without 
their permission.

The cell carriers are trying to get people excited about femotcell technology, 
so they can shift the cellular infrastructure costs off onto their customers.  
The traditional model is that the cell providers pay property and tower owners 
rent when they site an antenna.  

The cell providers don't even offer you a cheaper rate when you use the 
infrastructure you paid for.  Notice how the device supports up to three 
simultaneous calls?  You're even paying them to provide a better signal for 
your neighbors.

--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of C F
 Sent: Wednesday, September 19, 2007 6:57 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] OT: Samsung Sprint CDMAoIP
 
 http://gizmodo.com/gadgets/cellphones/sprintsamsung-instant-cell+to+wi+fi-
 box-is-official-named-airave-300451.php
 
 The above is quite interesting, it would be interesting to see if it
 uses sip, which I have no reason to believe otherwise, and if it does,
 can it be hacked to talk to Asteirsk? In which case one could have a
 very good extension to asterisk using any Sprint Cell phone, or maybe
 even any CDMA (Verizon) cell phone as well.
 
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Re: [asterisk-users] Announcing: Click-to-Call with VIDEO ***SPAM***

2007-09-20 Thread Anthony Francis
Can we please block these clowns? It appears they are incapable of learning.

C. Savinovich wrote:
 Dear All:

 Just as the name suggests, and evolving from regular Click-to-Call,
 Click-to-Call WITH VIDEO provides web sites with the ability to engage
 their visitors with a live video agent (plus the phone call).  All with just
 a click of a button placed on the customer's web site.  Please visit us at
 www.videoreps.net.

 The video technology integrates with Asterisk, and it also allows the
 agent to video-conference up to 10 callers at the same time, making it an
 ideal tool for online seminars, demos, shows, tech support, etc, etc.
 
 Click-to-call WITH VIDEO can be used in:

1) Customer service mode (web click-me buttons)
2) Conference mode (meetings of up to 10 participants)
3) Video-Call mode (enter user destination number and video-connect
 to him/her)

 It is an ideal add-on for pbx vendors who want to add video services to
 their pbxs.

 Best of all, it is free for now!! (video only)...  Here is the deal:
 Free for one month, no commitments.  Try, test it, call me.  After a month
 you can decide if you want to keep any of our plans:  Video-Call starts at
 $12.95 per month, and click-to-call with video at $29.95. And yes, there is
 20% monthly commission if you include the service in the pbxs you are
 selling.

 Here is how it works: Register online and you will receive an email with
 the confirmation link.  With your new username you will be able to enter the
 operator's work area.  Please fill your profile with your web site's ip (so
 that the buttons work), your destination phone (so that the phone call
 works) and a meeting room username (so that the videocall works).  Go to the
 demo buttons page and click on anyone you like to download the button and
 kit with a couple lines of php code you need to place the button in your web
 page.  Next, assuming you have a camera connected, click on GoOnLine and you
 will be online!! (don't eat at your desk!).
  
 I appreciate your kind input on this web site.  I am at my desk most of
 the time (logged in as a video agent), and I will be glad to be available
 when you click on our web site button, or when you enter my number
 19176135931 on the meeting room for quick-connect videophone.  However,
 hyper as I am, I would prefer to coordinate a demo time with anyone who
 wishes a demo session.

 Best Regards
 Christian Savinovich
 VideoReps.net

 Note: Being an ActiveX component, please use internet explorer.


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-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] Hfcmulti and B410P Digium Card

2007-09-20 Thread Matthew Fredrickson
voip crazy wrote:
 Hello all,
 
 I am getting the following error in  /var/log/syslog. I have got 2 B410P
 cards in this box.
 
 Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes
 (z1=0153, z2=00d3) TRANS
 Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space
 left (z1=0013, z2=0012) sending 128 of 128 bytes TRANS
 Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes
 (z1=0053, z2=0153) TRANS
 Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space
 left (z1=0093, z2=0092) sending 128 of 128 bytes TRANS
 Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes
 (z1=00d3, z2=0053) TRANS
 Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space
 left (z1=0113, z2=0112) sending 128 of 128 bytes TRANS
 Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes
 (z1=0153, z2=00d3) TRANS
 Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space
 left (z1=0013, z2=0012) sending 128 of 128 bytes TRANS
 Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes
 (z1=0053, z2=0153) TRANS
 Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(2) reading 15 bytes
 (z1=004e, z2=0040) HDLC COMPLETE (f1=3, f2=2) got=15
 Sep 19 17:13:31 localhost kernel: 02 01 06 06 08 01 63 5a 08 02 80 90
 Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space
 left (z1=0093, z2=0092) sending 128 of 128 bytes TRANS
 Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(2) has 383 bytes space
 left (z1=015d, z2=015d) sending 4 of 4 bytes HDLC
 
 I left untouched the /etc/init.d/misdn-init script to load the default
 values.
 
 Is needed the hfcmulti modules with this kind of cards?
 What is the menaing of this errors? Are something missconfigured?

Unless you are having some sort of problem other than this, than I think 
that this is just standard debug output, which you can disable if you 
set the debug option in /etc/misdn-init.conf to 0.

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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[asterisk-users] IAX Java Softphone?

2007-09-20 Thread Matthew Rubenstein
Does anyone know of an IAX softphone in Java, whether applet or
application? Even the most minimum featureset, just voice and dialing,
or even embedded in some other app/let. Preferably GPL. Thanks.
-- 

(C) Matthew Rubenstein


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Re: [asterisk-users] what is softswitch

2007-09-20 Thread Matthew Fredrickson
Alex Balashov wrote:
 On Wed, 19 Sep 2007, Anthony Francis wrote:
 
 IMHO asterisk is a softswitch, it may not be a very high capacity one 
 (right now) but it can be and if you don't mind splitting your physical 
 trunk calls over multiple machines it works very well as a call routing 
 engine, you just need to have carefully designed plans. It is far to 
 easy to create call routing loops, but if you don't know what you are 
 doing with a real telephony switch you can do the same.
 
No SS7/ISUP support (and no TCAP, which is required for LNP and LIDB and 
 traditional CNAM), poor/incomplete IMT support, can't take more than a few 
 T1s per host - if that. No GR.303 support.

Actually, I have been working on an SS7 stack for asterisk called 
libss7.  SS7 support is already in trunk, and should be in the next 
stable release of Asterisk.  Right now it only does ISUP/MTP3/MTP2, but 
with some work an effort, SCCP/TCAP/LNP support could be implemented as 
well.

Asterisk has had GR.303 support for a while, though I don't think it's 
asymmetric (it only supports one particular function of it, or something 
like that IIRC).

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] what is softswitch

2007-09-20 Thread Alex Balashov
On Thu, 20 Sep 2007, Matthew Fredrickson wrote:

 Actually, I have been working on an SS7 stack for asterisk called
 libss7.  SS7 support is already in trunk, and should be in the next
 stable release of Asterisk.  Right now it only does ISUP/MTP3/MTP2

   Look forward to it!  That will certainly confer an entirely new brand
of utility upon Asterisk.

 but with some work an effort, SCCP/TCAP/LNP support could be implemented 
 as well.

   If the work and effort is something Digium decides to invest, it's
certainly going to revolutionise what one can do with Asterisk.


--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO

2007-09-20 Thread Martin Smith
He changed the title of his response. Your post remains intact on the
list in its original form. In the interest of letting others decide, I
think it was a spammy post as well. Others have decided... Hence the
title of the list Asterisk Users Mailing List - Non-Commercial
Discussion. They decided it a long time ago. When you included pricing,
your email became commercial, an advertisement, and spam.

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 C. Savinovich
 Sent: Thursday, September 20, 2007 4:50 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] ***SPAM*** Announcing: 
 Click-to-Call with VIDEO
 
 
 Please don't change the title of my post.  It is 
 disrespectful.  One thing
 is to give your opinion about its content, and another to be 
 self appointed
 editor of this forum.
 
 We posted in this forum because it is a contribution to the asterisk
 community, and because it is free for a month, and maybe even 
 longer if the
 community so demands it.  If you agree or disagree with it 
 fine, but let
 others decide.  They know spam when they see it.  Thank you.
 
 C. Savinovich
 VideoReps.net
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Anselm Martin
 Hoffmeister
 Sent: Thursday, September 20, 2007 10:01 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] ***SPAM*** Announcing: 
 Click-to-Call with
 VIDEO
 
 Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich:
  Dear All:
  
  Just as the name suggests, and evolving from regular 
 Click-to-Call,
  Click-to-Call WITH VIDEO provides web sites with the 
 ability to engage
  their visitors with a live video agent (plus the phone 
 call).  All with
 just
  a click of a button placed on the customer's web site.  
 Please visit us at
  www.videoreps.net.
 
 When I read this, I thougt: Wow, here comes a nice, free, open,
 interesting software.
 
  Best of all, it is free for now!! (video only)...  Here 
 is the deal:
  Free for one month, no commitments.  Try, test it, call me. 
  After a month
  you can decide if you want to keep any of our plans:  
 Video-Call starts at
  $12.95 per month, and click-to-call with video at $29.95. 
 And yes, there
 is
  20% monthly commission if you include the service in the 
 pbxs you are
  selling.
 
 Well, not free at all.
 
  Note: Being an ActiveX component, please use internet explorer.
 
 And not having too much to do with open either, I guess.
 
 I would have called you to personally tell you that your mail was
 misplaced (there is some kind of asterisk-biz list, and I do 
 not read it
 for a purpose), but I do not use the integrated exploder for 
 lack of the
 necessary obfuscation system on my work machine.
 
 Please do not send commercials, ads and product information to this
 list. It might very well be considered SPAM. Just and only because
 _some_ readers might be interested there is no legitimation 
 for sending
 it (else every pen15-en1argm3nt _might_ trigger interest at some
 readers).
 
 Thanks
 Anselm
 
 
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Re: [asterisk-users] ***SPAM*** Announcing: Click-to -Call with VIDEO

2007-09-20 Thread Tilghman Lesher
On Thursday 20 September 2007, C. Savinovich wrote:
 We posted in this forum because it is a contribution to the asterisk
 community, and because it is free for a month, and maybe even longer
 if the community so demands it.  If you agree or disagree with it
 fine, but let others decide.  They know spam when they see it.  Thank
 you.

I am one of the longest running developers of Asterisk and I agree with
Anselm's assessment.  Please post announcements like this in the future
to the asterisk-biz list, as that list is for business discussions.
This list is for users of Asterisk, not for commercial announcements.

-- 
Tilghman

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Re: [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards

2007-09-20 Thread Jared Smith
On Thu, 2007-09-20 at 13:20 -0400, Brian Alexander wrote:
 
 Machine 1 experiences almost the same behavior on its span. The only
 differance I am noticing is that instead of the S-frame error I get
 the following notice:
 
   chan_zap.c:8457 pri_dchannel: PRI got event: HDLC Bad FCS (8) on
 Primary D-channel of span 1 

I'd look at your wiring, as an HDLC error like that is usually an
indication of some type of a problem at the physical layer.  


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] IAX Java Softphone?

2007-09-20 Thread Steven
I use click2call. http://www.geocities.com/babarnazmi/index2.htm

It is an activex control though.

-- 
-- 
Steven

http://www.glimasoutheast.org



Matthew Rubenstein [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Does anyone know of an IAX softphone in Java, whether applet or
 application? Even the most minimum featureset, just voice and dialing,
 or even embedded in some other app/let. Preferably GPL. Thanks.
 -- 

 (C) Matthew Rubenstein


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Re: [asterisk-users] OT: Samsung Sprint CDMAoIP

2007-09-20 Thread C F
On 9/20/07, Eric Chamberlain [EMAIL PROTECTED] wrote:
 The device is a femtocell device; I would bet that they keep it in a format 
 that works with their existing equipment, rather than use SIP.

 The device is also licensed for a specific frequency that is owned by the 
 carrier, you wouldn't be able to use this device for any other purpose 
 without their permission.

 The cell carriers are trying to get people excited about femotcell 
 technology, so they can shift the cellular infrastructure costs off onto 
 their customers.  The traditional model is that the cell providers pay 
 property and tower owners rent when they site an antenna.

AFAIK, the calls are free when you use it thru that device. Sprint
however charges $15 a month per phone or $30 for family plan. While I
agree that sprint should pay me for this, it's not as bad. T-mobile on
the other hand, does the same thing with wifi enabled phones, it
doesn't cost extra, and is completely free.


 The cell providers don't even offer you a cheaper rate when you use the 
 infrastructure you paid for.  Notice how the device supports up to three 
 simultaneous calls?  You're even paying them to provide a better signal for 
 your neighbors.

 --
 Eric Chamberlain, CISSP
 Chief Technical Officer
 Voxilla - http://voxilla.com/

  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of C F
  Sent: Wednesday, September 19, 2007 6:57 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [asterisk-users] OT: Samsung Sprint CDMAoIP
 
  http://gizmodo.com/gadgets/cellphones/sprintsamsung-instant-cell+to+wi+fi-
  box-is-official-named-airave-300451.php
 
  The above is quite interesting, it would be interesting to see if it
  uses sip, which I have no reason to believe otherwise, and if it does,
  can it be hacked to talk to Asteirsk? In which case one could have a
  very good extension to asterisk using any Sprint Cell phone, or maybe
  even any CDMA (Verizon) cell phone as well.
 
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Re: [asterisk-users] IAX Java Softphone?

2007-09-20 Thread Mike Clark
Matthew Rubenstein wrote:
 Does anyone know of an IAX softphone in Java, whether applet or
 application? Even the most minimum featureset, just voice and dialing,
 or even embedded in some other app/let. Preferably GPL. Thanks.
   
Mexuar's Coraletta is nice, but isn't GPL.

http://www.mexuar.com/products_sdk.shtml

Mike Clark

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Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO

2007-09-20 Thread Dean Collins
I was interested in it - commercial or otherwise.but only because I
used to work for the competition.

Commercial or otherwise it looks like a very cool technology and
something I'd be interested in - but only as a one of purchase price
rather than an ASP.



Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Martin Smith
 Sent: Thursday, 20 September 2007 1:59 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call
with VIDEO
 
 He changed the title of his response. Your post remains intact on the
 list in its original form. In the interest of letting others decide, I
 think it was a spammy post as well. Others have decided... Hence the
 title of the list Asterisk Users Mailing List - Non-Commercial
 Discussion. They decided it a long time ago. When you included
pricing,
 your email became commercial, an advertisement, and spam.
 
 Martin Smith, Systems Developer
 [EMAIL PROTECTED]
 Bureau of Economic and Business Research
 University of Florida
 (352) 392-0171 Ext. 221
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  C. Savinovich
  Sent: Thursday, September 20, 2007 4:50 PM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: Re: [asterisk-users] ***SPAM*** Announcing:
  Click-to-Call with VIDEO
 
 
  Please don't change the title of my post.  It is
  disrespectful.  One thing
  is to give your opinion about its content, and another to be
  self appointed
  editor of this forum.
 
  We posted in this forum because it is a contribution to the asterisk
  community, and because it is free for a month, and maybe even
  longer if the
  community so demands it.  If you agree or disagree with it
  fine, but let
  others decide.  They know spam when they see it.  Thank you.
 
  C. Savinovich
  VideoReps.net
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  Anselm Martin
  Hoffmeister
  Sent: Thursday, September 20, 2007 10:01 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] ***SPAM*** Announcing:
  Click-to-Call with
  VIDEO
 
  Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich:
   Dear All:
  
   Just as the name suggests, and evolving from regular
  Click-to-Call,
   Click-to-Call WITH VIDEO provides web sites with the
  ability to engage
   their visitors with a live video agent (plus the phone
  call).  All with
  just
   a click of a button placed on the customer's web site.
  Please visit us at
   www.videoreps.net.
 
  When I read this, I thougt: Wow, here comes a nice, free, open,
  interesting software.
 
   Best of all, it is free for now!! (video only)...  Here
  is the deal:
   Free for one month, no commitments.  Try, test it, call me.
   After a month
   you can decide if you want to keep any of our plans:
  Video-Call starts at
   $12.95 per month, and click-to-call with video at $29.95.
  And yes, there
  is
   20% monthly commission if you include the service in the
  pbxs you are
   selling.
 
  Well, not free at all.
 
   Note: Being an ActiveX component, please use internet explorer.
 
  And not having too much to do with open either, I guess.
 
  I would have called you to personally tell you that your mail was
  misplaced (there is some kind of asterisk-biz list, and I do
  not read it
  for a purpose), but I do not use the integrated exploder for
  lack of the
  necessary obfuscation system on my work machine.
 
  Please do not send commercials, ads and product information to this
  list. It might very well be considered SPAM. Just and only because
  _some_ readers might be interested there is no legitimation
  for sending
  it (else every pen15-en1argm3nt _might_ trigger interest at some
  readers).
 
  Thanks
  Anselm
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
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Re: [asterisk-users] IAX Java Softphone?

2007-09-20 Thread Jean-Denis Girard
Matthew Rubenstein a écrit :
 Does anyone know of an IAX softphone in Java, whether applet or
 application? Even the most minimum featureset, just voice and dialing,
 or even embedded in some other app/let. Preferably GPL. Thanks.

Did you try JIAXClient ?
http://www.hem.za.org/jiaxclient/


Regards,
-- 
Jean-Denis Girard

SysNux  Systèmes Linux en Polynésie française
http://www.sysnux.pf/   Tél: +689 483 527 / GSM: +689 797 527

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Re: [asterisk-users] IAX Java Softphone?

2007-09-20 Thread Dean Collins
Steven, how reliable is that freeware?

I tried it when it first came out but I couldn't get it to work. It
didn't matter at the time as I was working for Mexuar at the time but
now I don't have their service anymore I'd like to use it/something like
it for my other consultancy services.



Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Steven
 Sent: Thursday, 20 September 2007 2:12 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] IAX Java Softphone?
 
 I use click2call. http://www.geocities.com/babarnazmi/index2.htm
 
 It is an activex control though.
 
 --
 --
 Steven
 
 http://www.glimasoutheast.org
 
 
 
 Matthew Rubenstein [EMAIL PROTECTED] wrote in message
 news:[EMAIL PROTECTED]
 Does anyone know of an IAX softphone in Java, whether applet
or
  application? Even the most minimum featureset, just voice and
dialing,
  or even embedded in some other app/let. Preferably GPL. Thanks.
  --
 
  (C) Matthew Rubenstein
 
 
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Re: [asterisk-users] OT: Samsung Sprint CDMAoIP

2007-09-20 Thread Jason Parker
C F wrote:
 AFAIK, the calls are free when you use it thru that device. Sprint
 however charges $15 a month per phone or $30 for family plan. While I
 agree that sprint should pay me for this, it's not as bad. T-mobile on
 the other hand, does the same thing with wifi enabled phones, it
 doesn't cost extra, and is completely free.
 

If you're referring to T-Mobile's [EMAIL PROTECTED] service, it's actually $20
per month, per line on the account (unless it's changed very recently).

As far as how it works on T-Mobile, I recently had some questions answered by
them about that..  They use UMA over wifi, and it does automatic switching
between the wifi and the gsm towers (ie; your call stays up).

Quote from the tech I talked to:
[EMAIL PROTECTED] does not use a VoIP protocol, as the voice data is
transferred from the Internet directly to our UMA Gateway and then
through our regular Mobile Switching Centers.

Pretty interesting stuff.

-- 
Jason Parker
Digium

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Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO

2007-09-20 Thread Troy Ayers
snip
 Please don't change the title of my post.  It is 
 disrespectful.  One thing
 is to give your opinion about its content, and another to be 
 self appointed
 editor of this forum.

 We posted in this forum because it is a contribution to the asterisk
 community, and because it is free for a month, and maybe even 
 longer if the
 community so demands it.  If you agree or disagree with it 
 fine, but let
 others decide.  They know spam when they see it.  Thank you.

 C. Savinovich
 VideoReps.net
snip

As one of the others I say it looks like spam to me too.  I won't be 
trying your product anytime soon, partly because of the way the matter 
was handled.


-Troy


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Re: [asterisk-users] Newcomer Question

2007-09-20 Thread Guenther Sohler
Hallo Martin and Group!

Thank you very much for your perfect introduction into asterisk.
I managed to

* get asterisk server running
* configuring the internal numbers
* registering to 2 sip gateways
* outbound phoning to sipgate works perfect
* outbound phoning to mujtelefon not yet tested


The problem i am having now is, that i cant be reached by inbound phone calls 
from neither sipgate nor mujtelefon

i used my mobile to call this numbers.

sipgate tells me on the phone:Das Endgeraet ist fuer diesen Service nicht
konfiguriert. 
bei mujtelefon kommt nur die mobilbox

In asterisk cli i dont see anything about that. in the sipgate login page
there is neither mentioned.

If this is working, I intend to have nice music played for incoming calls
until the phone call is accepted.

What is very confusing for sipgate is, that my number(734365) is different from 
my user name(1734365). Can anybopdy check, if all settings are ok according to 
that ?

Please find below my sip.conf, only the passwords are scrambled.

If you directly reply to me, also reply to [EMAIL PROTECTED]
because i am afraid missing your answer in the much traffic in that mailing 
list. Thank you very much!

[general]   
   
bindport=5060   
   
bindaddr=0.0.0.0
   
disallow=all
   
allow=ulaw  
   
allow=alaw  
   
allow=ilbc  
   
allow=gsm   
   
musicclass=default  
   
language=de 
   

   
dtmfmode=rfc2833
   
sipdebug=no 
   

   
register = 1734365:[EMAIL PROTECTED]:5060/00437201734365   
 
register = 272048160:[EMAIL PROTECTED]:5060/00420272048160 
 

   
[sipgateat] 
   
host=sipgate.at 
   
secret=NMMTNMKP 
   
username=1734365
   
fromuser=1734365
   
fromdomain=sipgate.at   
   
srvlookup=yes   
   
context=sipgateat-in
   
canreinvite=no  
   
nat=no  
   
type=friend 
   
qualify=yes 

Re: [asterisk-users] IAX Java Softphone?

2007-09-20 Thread Guillermo Salas M.
On Thu, 2007-09-20 at 14:23 -0400, Mike Clark wrote:
 Matthew Rubenstein wrote:
  Does anyone know of an IAX softphone in Java, whether applet or
  application? Even the most minimum featureset, just voice and dialing,
  or even embedded in some other app/let. Preferably GPL. Thanks.

 Mexuar's Coraletta is nice, but isn't GPL.
 
 http://www.mexuar.com/products_sdk.shtml
 


I'm using JIAXClient [1], it is GPL, uses IAX2 and works pretty excelent
with gsm codec.

[1] http://www.hem.za.org/jiaxclient/


Regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net
SIP  : [EMAIL PROTECTED]
FWD  : 558563

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


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Re: [asterisk-users] Announcing: Click-to-Call with VIDEO

2007-09-20 Thread C. Savinovich

  OK gentlemen, thank you very much.

Best Regards
C. Savinovich
VideoReps.net

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: Thursday, September 20, 2007 11:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with
VIDEO

On Thursday 20 September 2007, C. Savinovich wrote:
 We posted in this forum because it is a contribution to the asterisk
 community, and because it is free for a month, and maybe even longer
 if the community so demands it.  If you agree or disagree with it
 fine, but let others decide.  They know spam when they see it.  Thank
 you.

I am one of the longest running developers of Asterisk and I agree with
Anselm's assessment.  Please post announcements like this in the future
to the asterisk-biz list, as that list is for business discussions.
This list is for users of Asterisk, not for commercial announcements.

-- 
Tilghman

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Re: [asterisk-users] IAX Java Softphone?

2007-09-20 Thread Dean Collins
As far as I know Jiaxclient is dead - the developer hasn't touched it in at 
least 18 months.

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jean-Denis Girard
 Sent: Thursday, 20 September 2007 2:46 PM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] IAX Java Softphone?
 
 Matthew Rubenstein a écrit :
  Does anyone know of an IAX softphone in Java, whether applet or
  application? Even the most minimum featureset, just voice and dialing,
  or even embedded in some other app/let. Preferably GPL. Thanks.
 
 Did you try JIAXClient ?
 http://www.hem.za.org/jiaxclient/
 
 
 Regards,
 --
 Jean-Denis Girard
 
 SysNux  Systèmes Linux en Polynésie française
 http://www.sysnux.pf/   Tél: +689 483 527 / GSM: +689 797 527
 
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[asterisk-users] Astricon Ride From Airport to Conf Hotel

2007-09-20 Thread JR Richardson
Hi All,

I'm arriving around noon in Phoenix on Tuesday the 25'Th and wouldn't
mind sharing a cab or car service.  I spoke with the hotel and the
'Super Shuttle' service can take 2-3 hours because the resort is the
last stop on the route.  A cab or car service will only take 30-40
minutes.

If anyone is coming in around the same time and needs a ride, contact
me off-list.

Thanks.

JR
-- 
JR Richardson
Engineering for the Masses

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Re: [asterisk-users] Outgoing SIP packets out of order?

2007-09-20 Thread Alex Balashov
Are you confident it's not a defect in Wireshark's RTP analyser?

On Thu, 20 Sep 2007, Jason Martin wrote:

 Hello, I've been looking at some SIP packet dumps captured with tcpdump on the
 PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something
 strange, at least to me. All of the SIP packets going out from our Asterisk
 PBX to either of our 2 VoIP providers are consistently 50% out of order. In
 addition, if I use Wireshark's voip call player, the outgoing side of the
 call stutters and is delayed compared to the incoming side of the call.

 Is this normal? Why would the PBX be sending packets already out of order?

 Thanks!
 -- 
 Jason Martin
 Metrix Matrix, Inc.
 785 Elmgrove Road, Building 1, Rochester, NY 14624
 Office: 888-865-0065 Ext. 202
 Mobile: (585) 721-8679


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--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] Announcing: Click-to-Call with VIDEO

2007-09-20 Thread C. Savinovich

Dear Dean:

   It is a very cool technology indeed, and please, do not see me as your
competition, but as a friend.  I know you have a click-to-call product, and
if there is any way I can be of help with providing the video technology for
you, I will be glad to set it up for you.  You are most welcomed to use
videoreps for free until you establish how it can benefit you.

C. Savinovich
VideoReps




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Thursday, September 20, 2007 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [PHISH] Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call
with VIDEO

I was interested in it - commercial or otherwise.but only because I
used to work for the competition.

Commercial or otherwise it looks like a very cool technology and
something I'd be interested in - but only as a one of purchase price
rather than an ASP.



Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Martin Smith
 Sent: Thursday, 20 September 2007 1:59 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call
with VIDEO
 
 He changed the title of his response. Your post remains intact on the
 list in its original form. In the interest of letting others decide, I
 think it was a spammy post as well. Others have decided... Hence the
 title of the list Asterisk Users Mailing List - Non-Commercial
 Discussion. They decided it a long time ago. When you included
pricing,
 your email became commercial, an advertisement, and spam.
 
 Martin Smith, Systems Developer
 [EMAIL PROTECTED]
 Bureau of Economic and Business Research
 University of Florida
 (352) 392-0171 Ext. 221
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  C. Savinovich
  Sent: Thursday, September 20, 2007 4:50 PM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: Re: [asterisk-users] ***SPAM*** Announcing:
  Click-to-Call with VIDEO
 
 
  Please don't change the title of my post.  It is
  disrespectful.  One thing
  is to give your opinion about its content, and another to be
  self appointed
  editor of this forum.
 
  We posted in this forum because it is a contribution to the asterisk
  community, and because it is free for a month, and maybe even
  longer if the
  community so demands it.  If you agree or disagree with it
  fine, but let
  others decide.  They know spam when they see it.  Thank you.
 
  C. Savinovich
  VideoReps.net
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  Anselm Martin
  Hoffmeister
  Sent: Thursday, September 20, 2007 10:01 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] ***SPAM*** Announcing:
  Click-to-Call with
  VIDEO
 
  Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich:
   Dear All:
  
   Just as the name suggests, and evolving from regular
  Click-to-Call,
   Click-to-Call WITH VIDEO provides web sites with the
  ability to engage
   their visitors with a live video agent (plus the phone
  call).  All with
  just
   a click of a button placed on the customer's web site.
  Please visit us at
   www.videoreps.net.
 
  When I read this, I thougt: Wow, here comes a nice, free, open,
  interesting software.
 
   Best of all, it is free for now!! (video only)...  Here
  is the deal:
   Free for one month, no commitments.  Try, test it, call me.
   After a month
   you can decide if you want to keep any of our plans:
  Video-Call starts at
   $12.95 per month, and click-to-call with video at $29.95.
  And yes, there
  is
   20% monthly commission if you include the service in the
  pbxs you are
   selling.
 
  Well, not free at all.
 
   Note: Being an ActiveX component, please use internet explorer.
 
  And not having too much to do with open either, I guess.
 
  I would have called you to personally tell you that your mail was
  misplaced (there is some kind of asterisk-biz list, and I do
  not read it
  for a purpose), but I do not use the integrated exploder for
  lack of the
  necessary obfuscation system on my work machine.
 
  Please do not send commercials, ads and product information to this
  list. It might very well be considered SPAM. Just and only because
  _some_ readers might be interested there is no legitimation
  for sending
  it (else every pen15-en1argm3nt _might_ trigger interest at some
  readers).
 
  Thanks
  Anselm
 
 
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[asterisk-users] Polycom 330 + Asterisk, phone locks up. * key will do it

2007-09-20 Thread telmnstr
Hello all,

   We have an Asterisk server that has worked without issue for a while. 
Before, only Sipura and Polycom 500 series phones were used.

   Recently, we've added a few POE switches and 20 or so Polycom 330's.

   The 330's seem to lock up often. One easy way to do this is by hitting 
the * key. I looked thru the XML files for dialplan and what not, nothing 
stood out.

   I've tried the latest everyone-can-download (bootrom 3.2.3.0021 and 
firmware 2.1.2.0048).

   Thoughts? The 500 series phones (501) have no issues.


   Also, the phones log called ID information and make it easy for users to 
hit a button and call back missed calls. Except, it doesn't log a 1 in 
front of the number. I guess I could adjust the dialplan to assume 9 digit 
numbers need a 1 in front, but has anyone found a better way?



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Re: [asterisk-users] Announcing: Click-to-Call with VIDEO

2007-09-20 Thread Dean Collins
I haven't been involved with Mexuar for about 4 months.

In the middle of moving at the moment but will be in touch in about 2
weeks once things get back to normal.

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of C. Savinovich
 Sent: Thursday, 20 September 2007 6:26 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Announcing: Click-to-Call with VIDEO
 
 
 Dear Dean:
 
It is a very cool technology indeed, and please, do not see me as
your
 competition, but as a friend.  I know you have a click-to-call
product, and
 if there is any way I can be of help with providing the video
technology for
 you, I will be glad to set it up for you.  You are most welcomed to
use
 videoreps for free until you establish how it can benefit you.
 
 C. Savinovich
 VideoReps
 
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
 Sent: Thursday, September 20, 2007 11:24 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [PHISH] Re: [asterisk-users] ***SPAM*** Announcing:
Click-to-Call
 with VIDEO
 
 I was interested in it - commercial or otherwise.but only because
I
 used to work for the competition.
 
 Commercial or otherwise it looks like a very cool technology and
 something I'd be interested in - but only as a one of purchase price
 rather than an ASP.
 
 
 
 Regards,
 
 Dean Collins
 Cognation Pty Ltd
 [EMAIL PROTECTED]
 +1-212-203-4357 Ph
 +61-2-9016-5642 (Sydney in-dial).
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
[mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Martin Smith
  Sent: Thursday, 20 September 2007 1:59 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call
 with VIDEO
 
  He changed the title of his response. Your post remains intact on
the
  list in its original form. In the interest of letting others decide,
I
  think it was a spammy post as well. Others have decided... Hence the
  title of the list Asterisk Users Mailing List - Non-Commercial
  Discussion. They decided it a long time ago. When you included
 pricing,
  your email became commercial, an advertisement, and spam.
 
  Martin Smith, Systems Developer
  [EMAIL PROTECTED]
  Bureau of Economic and Business Research
  University of Florida
  (352) 392-0171 Ext. 221
 
 
 
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On Behalf Of
   C. Savinovich
   Sent: Thursday, September 20, 2007 4:50 PM
   To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   Subject: Re: [asterisk-users] ***SPAM*** Announcing:
   Click-to-Call with VIDEO
  
  
   Please don't change the title of my post.  It is
   disrespectful.  One thing
   is to give your opinion about its content, and another to be
   self appointed
   editor of this forum.
  
   We posted in this forum because it is a contribution to the
asterisk
   community, and because it is free for a month, and maybe even
   longer if the
   community so demands it.  If you agree or disagree with it
   fine, but let
   others decide.  They know spam when they see it.  Thank you.
  
   C. Savinovich
   VideoReps.net
  
  
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On Behalf Of
   Anselm Martin
   Hoffmeister
   Sent: Thursday, September 20, 2007 10:01 AM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] ***SPAM*** Announcing:
   Click-to-Call with
   VIDEO
  
   Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich:
Dear All:
   
Just as the name suggests, and evolving from regular
   Click-to-Call,
Click-to-Call WITH VIDEO provides web sites with the
   ability to engage
their visitors with a live video agent (plus the phone
   call).  All with
   just
a click of a button placed on the customer's web site.
   Please visit us at
www.videoreps.net.
  
   When I read this, I thougt: Wow, here comes a nice, free, open,
   interesting software.
  
Best of all, it is free for now!! (video only)...  Here
   is the deal:
Free for one month, no commitments.  Try, test it, call me.
After a month
you can decide if you want to keep any of our plans:
   Video-Call starts at
$12.95 per month, and click-to-call with video at $29.95.
   And yes, there
   is
20% monthly commission if you include the service in the
   pbxs you are
selling.
  
   Well, not free at all.
  
Note: Being an ActiveX component, please use internet explorer.
  
   And not having too much to do with open either, I guess.
  
   I would have called you to personally tell you that your mail was
   misplaced (there is some kind of asterisk-biz 

[asterisk-users] Newcomer Question

2007-09-20 Thread Guenther Sohler
Hallo Martin and Group!

Thank you very much for your perfect introduction into asterisk.
I managed to

* get asterisk server running
* configuring the internal numbers
* registering to 2 sip gateways
* outbound phoning to sipgate works perfect
* outbound phoning to mujtelefon not yet tested


The problem i am having now is, that i cant be reached by inbound phone calls 
from neither sipgate nor mujtelefon

i used my mobile to call this numbers.

sipgate tells me on the phone:Das Endgeraet ist fuer diesen Service nicht
konfiguriert. 
bei mujtelefon kommt nur die mobilbox

In asterisk cli i dont see anything about that. in the sipgate login page
there is neither mentioned.

If this is working, I intend to have nice music played for incoming calls
until the phone call is accepted.

What is very confusing for sipgate is, that my number(734365) is different from 
my user name(1734365). Can anybopdy check, if all settings are ok according to 
that ?

Please find below my sip.conf, only the passwords are scrambled.

If you directly reply to me, also reply to [EMAIL PROTECTED]
because i am afraid missing your answer in the much traffic in that mailing 
list. Thank you very much!

[general]   
   
bindport=5060   
   
bindaddr=0.0.0.0
   
disallow=all
   
allow=ulaw  
   
allow=alaw  
   
allow=ilbc  
   
allow=gsm   
   
musicclass=default  
   
language=de 
   

   
dtmfmode=rfc2833
   
sipdebug=no 
   

   
register = 1734365:[EMAIL PROTECTED]:5060/00437201734365   
 
register = 272048160:[EMAIL PROTECTED]:5060/00420272048160 
 

   
[sipgateat] 
   
host=sipgate.at 
   
secret=NMMTNMKP 
   
username=1734365
   
fromuser=1734365
   
fromdomain=sipgate.at   
   
srvlookup=yes   
   
context=sipgateat-in
   
canreinvite=no  
   
nat=no  
   
type=friend 
   
qualify=yes 

Re: [asterisk-users] IAX Java Softphone?

2007-09-20 Thread Jean-Denis Girard
Dean Collins a écrit :
 As far as I know Jiaxclient is dead - the developer hasn't touched it in at 
 least 18 months.

Correct, but this is free software, anybody with the skills can revive it :)

Regards,
-- 
Jean-Denis Girard

SysNux  Systèmes Linux en Polynésie française
http://www.sysnux.pf/   Tél: +689 483 527 / GSM: +689 797 527

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Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO

2007-09-20 Thread Steve Totaro
Speaking for the Asterisk community as a whole, we demand that it be 
free forever.  Please honor your statement below, We posted in this 
forum because it is a contribution to the asterisk
  community, and because it is free for a month, and maybe even longer 
if the
  community so demands it.



C. Savinovich wrote:
 Please don't change the title of my post.  It is disrespectful.  One thing
 is to give your opinion about its content, and another to be self appointed
 editor of this forum.
 
   If you agree or disagree with it fine, but let
 others decide.  They know spam when they see it.  Thank you.
 
 C. Savinovich
 VideoReps.net
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin
 Hoffmeister
 Sent: Thursday, September 20, 2007 10:01 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with
 VIDEO
 
 Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich:
 Dear All:

 Just as the name suggests, and evolving from regular Click-to-Call,
 Click-to-Call WITH VIDEO provides web sites with the ability to engage
 their visitors with a live video agent (plus the phone call).  All with
 just
 a click of a button placed on the customer's web site.  Please visit us at
 www.videoreps.net.
 
 When I read this, I thougt: Wow, here comes a nice, free, open,
 interesting software.
 
 Best of all, it is free for now!! (video only)...  Here is the deal:
 Free for one month, no commitments.  Try, test it, call me.  After a month
 you can decide if you want to keep any of our plans:  Video-Call starts at
 $12.95 per month, and click-to-call with video at $29.95. And yes, there
 is
 20% monthly commission if you include the service in the pbxs you are
 selling.
 
 Well, not free at all.
 
 Note: Being an ActiveX component, please use internet explorer.
 
 And not having too much to do with open either, I guess.
 
 I would have called you to personally tell you that your mail was
 misplaced (there is some kind of asterisk-biz list, and I do not read it
 for a purpose), but I do not use the integrated exploder for lack of the
 necessary obfuscation system on my work machine.
 
 Please do not send commercials, ads and product information to this
 list. It might very well be considered SPAM. Just and only because
 _some_ readers might be interested there is no legitimation for sending
 it (else every pen15-en1argm3nt _might_ trigger interest at some
 readers).
 
 Thanks
 Anselm
 


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Re: [asterisk-users] [PHISH] Re: ***SPAM*** Announcing: Click-to-Call with VIDEO

2007-09-20 Thread C. Savinovich
Please honor your statement below, We posted in this forum because it is
a contribution to the asterisk community, and because it is free for a
month, and maybe even longer 
if the community so demands it.

   I take your request as an official demand from the community to provide
the service for free for a longer time.  We are happy to work with you guys.
Herewith, for anyone who signs up until Sunday September 23, will get 3
months free.   I will personally go through the database and extend everyone
who has signed with the committed time.

   We sincerely think we were making our 2 cents contribution to the
asterisk community by announcing an innovative concept on this forum.  We
apologize if we have inconvenienced anyone but are nevertheless glad to be
of any help.  Don't want to abuse our welcome here anymore, so we will be
happy to respond to personal inquiries.

Best Regards
C. Savinovich




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Thursday, September 20, 2007 12:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [PHISH] Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call
with VIDEO

Speaking for the Asterisk community as a whole, we demand that it be 
free forever.  Please honor your statement below, We posted in this 
forum because it is a contribution to the asterisk
  community, and because it is free for a month, and maybe even longer 
if the
  community so demands it.



C. Savinovich wrote:
 Please don't change the title of my post.  It is disrespectful.  One thing
 is to give your opinion about its content, and another to be self
appointed
 editor of this forum.
 
   If you agree or disagree with it fine, but let
 others decide.  They know spam when they see it.  Thank you.
 
 C. Savinovich
 VideoReps.net
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin
 Hoffmeister
 Sent: Thursday, September 20, 2007 10:01 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with
 VIDEO
 
 Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich:
 Dear All:

 Just as the name suggests, and evolving from regular Click-to-Call,
 Click-to-Call WITH VIDEO provides web sites with the ability to engage
 their visitors with a live video agent (plus the phone call).  All with
 just
 a click of a button placed on the customer's web site.  Please visit us
at
 www.videoreps.net.
 
 When I read this, I thougt: Wow, here comes a nice, free, open,
 interesting software.
 
 Best of all, it is free for now!! (video only)...  Here is the deal:
 Free for one month, no commitments.  Try, test it, call me.  After a
month
 you can decide if you want to keep any of our plans:  Video-Call starts
at
 $12.95 per month, and click-to-call with video at $29.95. And yes, there
 is
 20% monthly commission if you include the service in the pbxs you are
 selling.
 
 Well, not free at all.
 
 Note: Being an ActiveX component, please use internet explorer.
 
 And not having too much to do with open either, I guess.
 
 I would have called you to personally tell you that your mail was
 misplaced (there is some kind of asterisk-biz list, and I do not read it
 for a purpose), but I do not use the integrated exploder for lack of the
 necessary obfuscation system on my work machine.
 
 Please do not send commercials, ads and product information to this
 list. It might very well be considered SPAM. Just and only because
 _some_ readers might be interested there is no legitimation for sending
 it (else every pen15-en1argm3nt _might_ trigger interest at some
 readers).
 
 Thanks
 Anselm
 


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[asterisk-users] Paging MEETME_RECORDINGFILE Variable

2007-09-20 Thread Forrest Beck
I am having a weird issue with setting the recording file for the  
Page app.  Here is some quick background info


I have a macro that pages all my phones:

[macro-pageall]
; Context for paging all devices.
;   This will search the sip table in the realtime database
;   for all phones that start with a number.  That number is
;   passed to this macro as ${ARG1}.
;
;   ARG1 = The first digit of the phones to be paged (6=US  
Campus, 4=MS, 2=LS)

;   ARG2 = Device for the PA system.  If the user selected to
;   page the PA system.  That will be included.
;
exten = s,1,MYSQL(Connect connid ${realdb_host} ${realdb_user} $ 
{realdb_pass} ${realdb_db})
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ sip\  
WHERE\ name\ LIKE\ '${ARG1}%')

exten = s,3,MYSQL(Fetch fetchid ${resultid} number)
exten = s,4,GoToIf($[${fetchid} = 1]?5:7)
exten = s,5,Set(pagedevice=${pagedevice}SIP/${number})
exten = s,6,GoToIf($[${fetchid} = 1]?3:7)
exten = s,7,Set(pagedevice=${pagedevice:1})
exten = s,8,MYSQL(Clear ${resultid})
exten = s,9,MYSQL(Disconnect ${connid})
exten = s,10,GoToIf($[${ARG2} != ]?11:12)
exten = s,11,Set(pagedevice=${pagedevice}${ARG2})
;Add Call Info for GrandStream Phone on the PA system
exten = s,12,SIPAddHeader(Call-Info:answer-after=0)
;Add Alert-Info for all Polycom Phones
exten = s,13,SIPAddHeader(Alert-Info: Ring Answer)
exten = s,14,Set(MEETME_RECORDINGFILE=custom/paging/campuslastpage_$ 
{RAND(1|100)})

exten = s,15,NoOp(${MEETME_RECORDINGFILE})
exten = s,16,Set(CALLERID(all)=System Page 1010)
exten = s,17,Page(${pagedevice},r)
exten = h,1,System(/var/lib/asterisk/scripts/mail_lastpage ${ARG1} $ 
{MEETME_RECORDINGFILE})

exten = h,2,Hangup()

I call the macro with:
;Page All Phones including the PA system.
exten = 1010,1,Authenticate(12345)
exten = 1010,2,Macro(pageall,2,SIP/ls-pa)

Basically the macro goes through my sip realtime database and finds  
all the phones that begin with the number 2 (my lower school  
campus).  The generates a variable named pagedevice that looks like  
this:

SIP/2101SIP/2102SIP/2103

This part works great.

The issue I am having is setting the MEETME_RECORDINGFILE.  It should  
be set to an audio file in the custom sounds directory with a random  
number at the end.  I then use a hangup (h) extension to execute a  
script (at bottom of email) to email the audio file to a conference  
area in our email system (FirstClass).


What is weird is after I restart the asterisk process, this works  
fine for about a week.  It does exactly as it is supposed to, creates  
the audio file with a random number, then the email script delivers  
it.  After a week or so Asterisk will stop setting the variable  
MEETME_RECORDINGFILE and start placing the recordings in the sounds  
directory named meetme-conf-rec.##.wav.  Which is the default is  
MEETME_RECORDINGFILE is not set.


Anyone seen this issue before?

Thanks!


Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz


#!/bin/bash
#Set some variables
USFACULTY=[EMAIL PROTECTED]
LSFACULTY=[EMAIL PROTECTED]
USFACULTY=[EMAIL PROTECTED]
MONTH=`date +%B`
DAY=`date +%d`
YEAR=`date +%Y`
HOUR=`date +%I`
MINUTE=`date +%M`
ZONE=`date +%Z`
AMPM=`date +%P`
PGSOUNDDIR=/var/lib/asterisk/sounds/
LOGFILE=/var/log/mail_lastpage.log

#Write Log
echo `date` Running script for campus $1 with file $2  $LOGFILE

#Let give asterisk time to finish creating the recordng file.  Just  
in Case.

sleep 10

#
#Create a temp file with our message body
#
echo Repeat Last Page  /tmp/repeatpage_$1
echo   /tmp/repeatpage_$1
echo The attached WAV file is a copy of the last broadcast over the  
phone system.   /tmp/repeatpage_$1

echo   /tmp/repeatpage_$1
echo The page was broadcasted $MONTH $DAY, $YEAR at $HOUR:$MINUTE  
$AMPM. You may play this file back if you missed the page.  /tmp/ 
repeatpage_$1

echo   /tmp/repeatpage_$1
echo   /tmp/repeatpage_$1
echo If you wish to mark this email as read (Remove Red Flag)  
without opening the email, you may right-click (or control-click for  
Mac) and left-click Mark

as Read before opening the email.  /tmp/repeatpage_$1
#
#Send the email with the recorded Page attached
#

# Was it Upper School?
if [ $1 -eq 6 ]
then
cat /tmp/repeatpage_$1 | mutt -a $PGSOUNDDIR$2.wav - 
s Recording of Last Page for Upper School $USFACULTY

fi

# Was it Middle School?
if [ $1 -eq 4 ]
then
cat /tmp/repeatpage_$1 | mutt -a $PGSOUNDDIR$2.wav - 
s Recording of Last Page for Middle School $MSFACULTY

fi

# How about Lower?
if [ $1 -eq 2 ]
then
cat /tmp/repeatpage_$1 | mutt -a $PGSOUNDDIR$2.wav - 
s Recording of Last Page for Lower School $LSFACULTY

fi

rm -rf /tmp/repeatpage_$1
rm -f $PGSOUNDDIR$2.wav
exit

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To 

Re: [asterisk-users] Polycom 330 + Asterisk, phone locks up. * key will do it

2007-09-20 Thread Darren Nickerson
[EMAIL PROTECTED] wrote:

 Hello all,

   We have an Asterisk server that has worked without issue for a while.
 Before, only Sipura and Polycom 500 series phones were used.

   Recently, we've added a few POE switches and 20 or so Polycom 330's.

   The 330's seem to lock up often. One easy way to do this is by hitting
 the * key. I looked thru the XML files for dialplan and what not, nothing
 stood out.

   I've tried the latest everyone-can-download (bootrom 3.2.3.0021 and
 firmware 2.1.2.0048).

   Thoughts? The 500 series phones (501) have no issues.


Without a doubt the most common problem with deploying newer Polycom 
handsets is the presence of old configurations instructions in the XML 
provisioning files. In most cases, people plug in a 650, provision it just 
like they did with all their 501s, and the reliability is just, well, .. 
terrible! The degree of pain depends on how (and how much) you have tweaked 
the configs over the years.

The fix can be a little painful, but is worth doing. Upgrade to the latest 
firmware and XML config templates from Polycom (assuming you purchased from 
an official polycom reseller), and port your local changes as overrides from 
the new default file.

Config files are backward compatible, but not forward compatible. What 
worked with your 501 will not make your 430, 650, 330 or 320 happy.

-Darren

-- 
Darren Nickerson
Senior Sales  Support Engineer
Telephony Depot
www.telephonydepot.com
+1.215.825.8710 ext 8106 (office)
+1.215.243.8335 (fax) 


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[asterisk-users] GROUP() issues for me

2007-09-20 Thread Nicholas Blasgen
I've got a macro that tries to find the first available SIP trunk to send
outgoing calls on.  It tracks the usage of the lines (since each trunk has a
call-limit of 2) by using GROUP().  My problem is that once a call switched
to ANSWER state, ``group show channels`` stops listing it and then my Macro
starts screwing up because it's sending calls to a line that sometimes is
full even though GROUP() shows it as being less than 2.  I'm tempted to send
this to the Asterisk Dev team just because I believe it's an issue of the
GROUP information being released when Asterisk consolidates the channels
(removes all the MASQ channels) once the call is answered.  But maybe it's
something else so I'll ask here first.

The dialplan setup:

exten = 555,1,Dial(Local/1234567890)
exten = _NXXNXX,1,Macro(which-line,${EXTEN})

[macro-which-line]
exten = s,1,set(GROUP()=${DIALSTRING})
exten = s,n,Dial(${DIALSTRING}/1${ARG1})
Things are a bit more complex, but it's all just logic.  The extensions
above should give a decent representation of what's going on.  I think each
time you switch extensions, Asterisk creates a MASQ channel and that's
what's causing the issue since the GROUP() is set only at the end, inside
the macro.  Are there any EVENTS for unlocking of GROUPs?  Anything I can do
to better show where this is happening?

I'd love some help if anyone has a guess.
-- 
/Nick
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[asterisk-users] Asterisk 1.2.24 simultaneous call limits.

2007-09-20 Thread Wai Wu
 
Hi everyone,

I am running into wall today with simultaneous call limits. I have two
Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a
lot of sip calls from one machine to the other by issuing AMI Originate
commands to one machine. The machine that makes calls plays a message
(demo-intruct) upon the other machine answer. The machine receives the
calls just waits for 40 seconds then hangs up. Throught the manager
connection, I was creating 10 calls per-second. I also have sip phone
registered with the calling machine. At around 150 to 200 calls. When I
call the machine that's making all the calls, most of the calls couldn't
go through. For the ones that went through, most of them will drop off
within seconds of the call. But here is catch. When I run 'top', the cpu
is idling 97%. My question is. Is there a limit on the number of
simultaneous calls Asterisk can handle? I know I have very fast systems.
Shouldn't they be able to handle that many calls? What is your take?

Thnx

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Re: [asterisk-users] GROUP() issues for me

2007-09-20 Thread Julian Lyndon-Smith
try

Nicholas Blasgen wrote:
 I've got a macro that tries to find the first available SIP trunk to send
 outgoing calls on.  It tracks the usage of the lines (since each trunk has a
 call-limit of 2) by using GROUP().  My problem is that once a call switched
 to ANSWER state, ``group show channels`` stops listing it and then my Macro
 starts screwing up because it's sending calls to a line that sometimes is
 full even though GROUP() shows it as being less than 2.  I'm tempted to send
 this to the Asterisk Dev team just because I believe it's an issue of the
 GROUP information being released when Asterisk consolidates the channels
 (removes all the MASQ channels) once the call is answered.  But maybe it's
 something else so I'll ask here first.
 
 The dialplan setup:
 
 exten = 555,1,Dial(Local/1234567890)

exten = 555,1,Dial(Local/1234567890/n)

note the /n

 exten = _NXXNXX,1,Macro(which-line,${EXTEN})
 
 [macro-which-line]
 exten = s,1,set(GROUP()=${DIALSTRING})
 exten = s,n,Dial(${DIALSTRING}/1${ARG1})
 Things are a bit more complex, but it's all just logic.  The extensions
 above should give a decent representation of what's going on.  I think each
 time you switch extensions, Asterisk creates a MASQ channel and that's
 what's causing the issue since the GROUP() is set only at the end, inside
 the macro.  Are there any EVENTS for unlocking of GROUPs?  Anything I can do
 to better show where this is happening?
 
 I'd love some help if anyone has a guess.

that's my guess

Julian

 
 
 
 
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Re: [asterisk-users] GROUP() issues for me

2007-09-20 Thread Nicholas Blasgen

 exten = 555,1,Dial(Local/1234567890/n)

 note the /n


I'm going to try this in a bit (can't hurt anything, might as well), but I'd
like to understand you're reasoning.  You're dialing an extra extension?

I'm also going to be trying this with Asterisk 1.6 TRUNK to see if it's even
a current issues in the development branch but I wont have a chance untill
tomorrow sometime.
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Re: [asterisk-users] Asterisk 1.2.24 simultaneous call limits.

2007-09-20 Thread Nicholas Blasgen
Just thinking about it quickly, it's always possible it has nothing to do
with Asterisk.  There are many instances where I run into issues with a
poorly configured servers when they have even a little bump in HTTP
traffic.  This was years ago though, and it was an issue to do with a web
server and not Asterisk, but look into your kernel's configuration.
Sometimes the kernel's settings are setup for a normal USER and not designed
to handle the memory allocation a server demands.  The fix for me back then
was something to do with the MAXIMUM PAGE REQUESTS or SIZE maybe.  Basicly
the kernel couldn't keep track of all the HTTP processes.

Now that I'm reading this over I doubt it's your problem because Asterisk
doesn't fork.  But while we're at it, tell me a bit more about your system.
What operating system (and version)?  The problem could also be with your
method of load generation, but I wouldn't know that since I've never tried
load testing a system.

Lastly, I know FreeBSD started incorporating a basic DDoS protection a few
years back and maybe that's also in some of these newer Linux distros.  They
would detect a flood and start to limit the bandwidth.  These are just
ideas, I don't really like any of them.

Sometimes the kernel will report issues to SYSLOGD.  Might want to check
your error and message logs.

cat /proc/meminfo

On a Linux box will give you memory limits and how close you are to them.
They're not exactly what I was looking for, but maybe that will help.  All
TCP connections require the Kernel to page the information but I can't seem
to find out how to access that limit if any.


On 9/20/07, Wai Wu [EMAIL PROTECTED] wrote:


 Hi everyone,

 I am running into wall today with simultaneous call limits. I have two
 Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a
 lot of sip calls from one machine to the other by issuing AMI Originate
 commands to one machine. The machine that makes calls plays a message
 (demo-intruct) upon the other machine answer. The machine receives the
 calls just waits for 40 seconds then hangs up. Throught the manager
 connection, I was creating 10 calls per-second. I also have sip phone
 registered with the calling machine. At around 150 to 200 calls. When I
 call the machine that's making all the calls, most of the calls couldn't
 go through. For the ones that went through, most of them will drop off
 within seconds of the call. But here is catch. When I run 'top', the cpu
 is idling 97%. My question is. Is there a limit on the number of
 simultaneous calls Asterisk can handle? I know I have very fast systems.
 Shouldn't they be able to handle that many calls? What is your take?

 Thnx

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Re: [asterisk-users] Queue Question

2007-09-20 Thread Kevin Smith
Hi Jeremy,
A few thoughts that come to mind. We have a queue that is open between 
certain hours. I have a few checks in place before a caller enters, 
first it checks to see if there it is within the time window, then 
checks to see if there are any agents log into queue, if any fail they 
get our closed message. Sounds like you are trying to do something similar.
Not sure what you have for extension numbers numbers, but you will get 
the idea.

Your first friend:
GotoIfTime(time range|days of week|days of 
month|months?[[context|]extension|]pri)
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime

I don't know how your dial plan is structured. My guess is the after 
hours operation is in a separate part of the code from the other. Since 
we are just looking at after hours, I would use the reverse on your 
time. Because the command jumps when the statement is true. I do not 
know what will happen if you say go from 17:00-8:00, but you can try it.

Example:
exten = 800,1,GotoIfTime(8:00-17:00|mon-fri|*|*?NormalOp,900,1) ; Since 
this will fail if it is 9pm, it moves on to the next priority in this 
exten.

[NormalOp]
exten = 900,1,blah

Next, is your other test. Use the queue agent count function 
QUEUEAGENTCOUT(queuename)
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+queueagentcount

If the number is greater then 0, then you move them into the queue, if 
not, whatever you want.

Finally, in terms of your other questions about logging the agents in. 
You could do the database way. You also could create a log in extension 
where you can take their cell number ( caller id) and use the 
application AddQueueMember(queuename[|interface][|penalty])

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AddQueueMember

So you should be able to do something like
AddQueueMember(queueName|ZAP/${CALLID(num)})

Anyway hope that helps.

Kevin




Jeremy Mann wrote:

 I’m curious if anyone has implemented the following:

 Need to setup an on-call queue, that activates after 5PM and 
 de-activates at 8AM, also that activates/deactivates on demand(I’m 
 thinking a feature code here). The “agents” need to log in via cell 
 phones, and when calls come in from outside to the asterisk system, 
 it’ll need to call the cell phone agents that are active.

 I’m thinking that it’s a simple SQL query, to update the agents status 
 and number, and that asterisk will do a lookup and append that to the 
 ZAP channel to dial, but interested in any logic someone might be able 
 to come up with for the dialplan.


 
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Re: [asterisk-users] Asterisk 1.2.24 simultaneous call limits.

2007-09-20 Thread Wai Wu
Interesting. I am using PCLinuxOS(Mandrak) in console mode. Here is my
memory info and you can see that I still have a lot of memory while
asterisk is running
 
[EMAIL PROTECTED] ~]# cat /proc/meminfo
MemTotal:  2076000 kB
MemFree:   1855636 kB
Buffers: 17224 kB
Cached: 115916 kB
SwapCached:  0 kB
Active: 124100 kB
Inactive:73468 kB
HighTotal: 1179264 kB
HighFree:   992808 kB
LowTotal:   896736 kB
LowFree:862828 kB
SwapTotal: 1365484 kB
SwapFree:  1365484 kB
Dirty: 216 kB
Writeback:   0 kB
AnonPages:   64524 kB
Mapped:  43912 kB
Slab:13168 kB
PageTables:   1344 kB
NFS_Unstable:0 kB
Bounce:  0 kB
CommitLimit:   2403484 kB
Committed_AS:   142512 kB
VmallocTotal:   114680 kB
VmallocUsed:  8484 kB
VmallocChunk:   104492 kB
[EMAIL PROTECTED] ~]#

You mentioned DDoS projection. How can I find out if my distro has it
built in?



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nicholas
Blasgen
Sent: Thursday, September 20, 2007 8:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.2.24 simultaneous call limits.


Just thinking about it quickly, it's always possible it has nothing to
do with Asterisk.  There are many instances where I run into issues with
a poorly configured servers when they have even a little bump in HTTP
traffic.  This was years ago though, and it was an issue to do with a
web server and not Asterisk, but look into your kernel's configuration.
Sometimes the kernel's settings are setup for a normal USER and not
designed to handle the memory allocation a server demands.  The fix for
me back then was something to do with the MAXIMUM PAGE REQUESTS or SIZE
maybe.  Basicly the kernel couldn't keep track of all the HTTP
processes. 
 
Now that I'm reading this over I doubt it's your problem because
Asterisk doesn't fork.  But while we're at it, tell me a bit more about
your system.  What operating system (and version)?  The problem could
also be with your method of load generation, but I wouldn't know that
since I've never tried load testing a system. 
 
Lastly, I know FreeBSD started incorporating a basic DDoS protection a
few years back and maybe that's also in some of these newer Linux
distros.  They would detect a flood and start to limit the bandwidth.
These are just ideas, I don't really like any of them. 
 
Sometimes the kernel will report issues to SYSLOGD.  Might want to check
your error and message logs.
 
cat /proc/meminfo
 
On a Linux box will give you memory limits and how close you are to
them.  They're not exactly what I was looking for, but maybe that will
help.  All TCP connections require the Kernel to page the information
but I can't seem to find out how to access that limit if any. 

 
On 9/20/07, Wai Wu [EMAIL PROTECTED] wrote: 


Hi everyone,

I am running into wall today with simultaneous call limits. I
have two
Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to
create a 
lot of sip calls from one machine to the other by issuing AMI
Originate
commands to one machine. The machine that makes calls plays a
message
(demo-intruct) upon the other machine answer. The machine
receives the 
calls just waits for 40 seconds then hangs up. Throught the
manager
connection, I was creating 10 calls per-second. I also have sip
phone
registered with the calling machine. At around 150 to 200 calls.
When I
call the machine that's making all the calls, most of the calls
couldn't
go through. For the ones that went through, most of them will
drop off
within seconds of the call. But here is catch. When I run 'top',
the cpu 
is idling 97%. My question is. Is there a limit on the number of
simultaneous calls Asterisk can handle? I know I have very fast
systems.
Shouldn't they be able to handle that many calls? What is your
take?

Thnx

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Re: [asterisk-users] GROUP() issues for me

2007-09-20 Thread Eric ManxPower Wieling
Nicholas Blasgen wrote:
 exten = 555,1,Dial(Local/1234567890/n)

 note the /n
 
 
 I'm going to try this in a bit (can't hurt anything, might as well), but I'd
 like to understand you're reasoning.  You're dialing an extra extension?
 
 I'm also going to be trying this with Asterisk 1.6 TRUNK to see if it's even
 a current issues in the development branch but I wont have a chance untill
 tomorrow sometime.

read localchannel.txt or whatever obvious file in your version of 
Asterisk.  /path/to/src/asterisk/docs/

Also, you need an @context, for example:
   exten = 555,1,Dial(Local/[EMAIL PROTECTED]/n)

Where corporate is whatever context the extension 1234567890 is 
located in.  Maybe not needing a context is new in 1.4, but the doc file 
would tell you.

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Re: [asterisk-users] OT: Samsung Sprint CDMAoIP

2007-09-20 Thread C F
On 9/20/07, Jason Parker [EMAIL PROTECTED] wrote:
 C F wrote:
  AFAIK, the calls are free when you use it thru that device. Sprint
  however charges $15 a month per phone or $30 for family plan. While I
  agree that sprint should pay me for this, it's not as bad. T-mobile on
  the other hand, does the same thing with wifi enabled phones, it
  doesn't cost extra, and is completely free.
 

 If you're referring to T-Mobile's [EMAIL PROTECTED] service, it's actually 
 $20
 per month, per line on the account (unless it's changed very recently).


I don't know about that, could be you are right.

 As far as how it works on T-Mobile, I recently had some questions answered by
 them about that..  They use UMA over wifi, and it does automatic switching
 between the wifi and the gsm towers (ie; your call stays up).

The same goes for Sprint.


 Quote from the tech I talked to:
 [EMAIL PROTECTED] does not use a VoIP protocol, as the voice data is
 transferred from the Internet directly to our UMA Gateway and then
 through our regular Mobile Switching Centers.

I know it's a quote from the tech, but isn't it voice packets that
travels over the Internet (a packet switched network) instead of over
GSM (TDM switched network) which makes that statement incorrect? It
doesn't matter what the higher level protocol is, it's still VoIP.


 Pretty interesting stuff.

 --
 Jason Parker
 Digium

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[asterisk-users] Dialing an external number and then passing it to an extension...

2007-09-20 Thread Carlos Chavez
 I am in need of some guidance regarding the following problem:

I need to dial an external number from a list(PSTN)
I need to check if the number is busy, no answer or fail
If any of the above are met then I try another number from a list
If none of the above happen then I first need to determine if the line
answering is a fax machine or an answering machine
If fax or answering machine then hangup and try next number
If human then connect to an internal extension

 An outbound callcenter suite is overkill since we only need two or three
calls at a time.  Can something like this be done using the Originate command
on AMI?  The main problem I have is that if I dial an external call and it
fails for some reason how do I know?  Is there something like ${DIALSTATUS}
that can give me the result of that part of the call?

 We plan to have a web interface that will fire the call when you click a
button.  That will fire an event that connects to the manager interface and
uses originate to dial the external call and then dial the internal extension
if all conditions are met.  The numbers will be in a database.

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001


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Re: [asterisk-users] OT: Samsung Sprint CDMAoIP

2007-09-20 Thread Alexander Lopez


Snip headers
 On 9/20/07, Jason Parker [EMAIL PROTECTED] wrote:
  C F wrote:
   AFAIK, the calls are free when you use it thru that device. Sprint
   however charges $15 a month per phone or $30 for family plan.
While I
   agree that sprint should pay me for this, it's not as bad.
T-mobile on
   the other hand, does the same thing with wifi enabled phones, it
   doesn't cost extra, and is completely free.
  
 
  If you're referring to T-Mobile's [EMAIL PROTECTED] service, it's
actually
 $20
  per month, per line on the account (unless it's changed very
recently).
 
 
 I don't know about that, could be you are right.
 
  As far as how it works on T-Mobile, I recently had some questions
 answered by
  them about that..  They use UMA over wifi, and it does automatic
 switching
  between the wifi and the gsm towers (ie; your call stays up).
 
 The same goes for Sprint.
 
 
  Quote from the tech I talked to:
  [EMAIL PROTECTED] does not use a VoIP protocol, as the voice data is
  transferred from the Internet directly to our UMA Gateway and then
  through our regular Mobile Switching Centers.
 
 I know it's a quote from the tech, but isn't it voice packets that
 travels over the Internet (a packet switched network) instead of over
 GSM (TDM switched network) which makes that statement incorrect? It
 doesn't matter what the higher level protocol is, it's still VoIP.
 
Your right it is STILL VoIP by definition but its not...

From: http://www.newstep.com/our%20market/technologies.asp

 Gateway-based Solutions
By placing special gateways at the edge of a GSM network, Unlicensed
Mobile Access (UMA) allows users with dual-mode handsets to access
mobile phone services via both cellular and Wi-Fi links. In cellular
mode, voice traffic travels over standard GSM radio waves. In Wi-Fi
mode, an IP tunnel carries GSM traffic across the enterprise network
and/or the Internet to a UMA gateway. The gateway looks like a base
station controller (BSC) to the cellular network, so when a handset
moves between cellular and Wi-Fi coverage, the network handles it as an
ordinary BSC-to-BSC handoff. MSC emulation-also known as IP VLR-is
similar to UMA, except that the gateway mimics a mobile switching center
(MSC) and a visitor location register (VLR) instead of a BSC.

Intimately tied to cellular technology and dual-mode handsets,
gateway-based solutions provide access only to mobile network services
and can be deployed only by facilities-based mobile network operators.
Moreover, gateway-based solutions cannot leverage the full capabilities
of IP and VoIP because all voice traffic remains in TDM format. Service
providers, therefore, view gateway-based solutions as an inadequate
response to the FMC opportunity. They are turning instead to
server-based technology, a more generalized approach that spans all
types of networks: fixed and mobile, IP and TDM, business and
residential.

 
  Pretty interesting stuff.
 
  --
  Jason Parker
  Digium
 
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