date:20060906


Hi Marco,
in attachment you can find my misdn.conf. Consider that I'm still fixing 
some warning because I recently upgraded from install-misdn to 
install-misdn-mqueue but the driver installation manual has not changed. 
Some comments are due to the fact I'm still making tests to solve the 
incoming calls problem I mentioned.



Thank you.


Giorgio Incantalupo




Marco Mouta wrote:
Please post your *misdn*-*init*.*conf as well as misdn.conf so i can 
try to help u*


On 9/5/06, *Giorgio Incantalupo* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Hi,
I hava an Asterisk box with a monoBRI + install-misdn-mqueue
0.3.1-rc23
package installed.
I can make outbound calls but cannot receive any. I get no Asterisk
messages on the console except for these:

P[ 1] GOT IGNORE SETUP
P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE]
P[ 1] release_chan: Ch not found!

Is there anybody who can help me, please?


TIA


Giorgio Incantalupo
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Com os melhores cumprimentos,

Marco Mouta


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[general]
debug = 5
tracefile = /var/log/asterisk/misdn.trace
trace_calls = false
trace_dir = /var/log/asterisk/misdn
bridging = yes
stop_tone_after_first_digit = yes
append_digits2exten = yes
l1_info_ok = yes
clear_l3 = no
method = standard

;;; CRYPTION STUFF

dynamic_crypt = no
crypt_prefix = **
crypt_keys = test,muh

; users sections:

[default]
context = misdn
language = it
nationalprefix = 0
internationalprefix = 00
rxgain = 0
txgain = 0
te_choose_channel = no
dialplan = 0
use_callingpres = yes
echocancelwhenbridged = no
echotraining = yes

;
; inbound group
;
[inbound]
; Giorgio test:
pmp_l1_check=no
;msns=*
; end test
ports = 1
context = outbound_isdn


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Re: [asterisk-users] macros in Realtime

2006-09-06 Thread Simon Woodhead

Hi Ben,Yes it is but you need to remember to still include[macro-stdpbx1exten]switch = Realtime/in your extensions.confSimonOn 9/6/06, 
Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello all,Another question related to Realtime.Is it possible to call macros using Realtime arch?I have a macro definition in table extensions_conf in my MySql db as: 30 | macro-stdpbx1exten | s |
1 | SET| fwdedNum=${DB(CFWD/${ARG1})}And calling the Macro using another entry in table extensions_conf inMysql db: 40 | pbx1 | _[345]. |
1 | Macro| stdpbx1exten|${EXTEN}I get errors like :Sep6 11:18:53 WARNING[14493]: app_macro.c:154 macro_exec: No suchcontext 'macro-stdpbx1exten' for macro 'stdpbx1exten'
Are there issues with the same??TiA-Ben.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
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[asterisk-users] How to test TE405P T1

2006-09-06 Thread Andy Chung (Power-All)


Hi all,

I have connected a T1IDA-P to the Digium TE405P. Checked with the Telco, 
and confirmed the T1 is up and connected. However, I have no idea how to 
test the T1 is really work, because the Asterisk server not yet be 
configure. Anyone has the method on how to test the calls through the T1?


Thanks!
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[asterisk-users] Budgetones - multiple phones losing IP address during day

2006-09-06 Thread Garth van Sittert


Hi All

I have a site with 50 Budgetone 102's and about 5 snom phones.

At random intervals during the day about 20 or 30 of the Budgetones lose 
their connection to the network all at the same time.  It happens about 
once a day.  The Snom phones are fine and never get disconnected.  I 
can't ping the Budgetones IP's and the way to fix them is to simply 
unplug and reconnect.  Nothing interesting shows up in the logs even in 
debug mode except for the 'Peer XXX is now unreachable' repeated for 
each extension.


I haven't had much experience with the Budgetones.  Does anyone have any 
idea what could be causing this?


Thanks
Garth

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Re: [asterisk-users] Budgetones - multiple phones losing IP address during day

2006-09-06 Thread Brandon Galbraith

Garth,Are they all on the same switch? Possibly could be a network-level issue, and not something wrong with the phones.-brandonOn 9/6/06, 
Garth van Sittert [EMAIL PROTECTED] wrote:
Hi AllI have a site with 50 Budgetone 102's and about 5 snom phones.At random intervals during the day about 20 or 30 of the Budgetones losetheir connection to the network all at the same time.It happens about
once a day.The Snom phones are fine and never get disconnected.Ican't ping the Budgetones IP's and the way to fix them is to simplyunplug and reconnect.Nothing interesting shows up in the logs even in
debug mode except for the 'Peer XXX is now unreachable' repeated foreach extension.I haven't had much experience with the Budgetones.Does anyone have anyidea what could be causing this?Thanks
Garth___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: 
[EMAIL PROTECTED]AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost
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[asterisk-users] core dumps

2006-09-06 Thread Anthony Musaluke


Hell all,

I have asterisk dying and restarting. After it dies, it creates a core file. 
Running gdb on this file produces the details below. This is very random and 
I am not sure what could be happening. Any ideas where I need to look?

Thanks 
Anthony
--
(gdb) bt full
#0  0x00881402 in __kernel_vsyscall ()
No symbol table info available.
#1  0x001d38f8 in raise () from /lib/libc.so.6
No symbol table info available.
#2  0x001d5068 in abort () from /lib/libc.so.6
No symbol table info available.
#3  0x00208a0a in __libc_message () from /lib/libc.so.6
No symbol table info available.
#4  0x0020f8ca in _int_malloc () from /lib/libc.so.6
No symbol table info available.
#5  0x002109b9 in calloc () from /lib/libc.so.6
No symbol table info available.
#6  0x00727051 in sip_alloc (callid=0x0, sin=0x0, useglobal_nat=0, 
intended_method=3) at chan_sip.c:3076
p = (struct sip_pvt *) 0x20e4ad
__PRETTY_FUNCTION__ = sip_alloc
#7  0x0073100e in sip_poke_peer (peer=0x8b19a60) at chan_sip.c:11629
__PRETTY_FUNCTION__ = sip_poke_peer
#8  0x0073dd87 in sip_poke_peer_s (data=0x0) at chan_sip.c:5761
No locals.
#9  0x08056778 in ast_sched_runq (con=0x8b251f8) at sched.c:373
x = 0
res = 0
#10 0x00747ddf in do_monitor (data=0x0) at chan_sip.c:11523
res = 0
sip = Variable sip is not available.
--
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Re: [asterisk-users] Merlin Legend - Working Now!

Rule #1.  Never read the manual. :-D


Sterling Moses wrote:
I was able to track this problem down and solved it with a little 
manual reading.


I used the remote access features documented on page 3-189 in the 
Revision 7 manual with a few security tweaks.


I will soon update the wiki with my procedures.

Thanks guys.

Sterling.

On Sep 5, 2006, at 6:00 PM, Steve Totaro wrote:

I took a different approach with a Definity system and put asterisk 
between the Definity and the telco.  For my setup I had to create a 
trunk group and a route then tell the Definity to pattern match and 
use that new route, it is called ARS Analysis or something obscure.
In your setup, you would need to look for trunk to trunk transfer 
options or something similar.  I have only worked with a Merlin 
Legend a couple of times and didn't get that involved with it.
Thanks for reminding me to document my efforts with the Definity, 
there is some stuff on Google but nothing close to a step by step 
how-to.


Thanks,
Steve

Sterling Moses wrote:

Hello List,

I was able to track down my issues with the PRI from a few weeks 
back. Turns out the legend switch had a Terminal Equipment ID of 2 
and was supposed to be a 0. Go figure. This was keeping the 
D-Channel from coming up although the rest of the B-Channels were 
up.  It was very easy to configure the legend and the asterisk 
system to talk both ways once we figured out the PRI hang up.


 I am in the process of writing up a wiki page for the user 
community on how to accomplish a legend to asterisk integration from 
start to finish and have one more issue to solve, so this is where 
all you legend gurus come in to play.


The system is laid out with Telco PRI connected to Legend which 
connects via PRI to Asterisk. Calls are working both directions 
across the legend-asterisk pri.


How can the Legend switch be programmed to provide an outside trunk 
(via the telco pri) to the asterisk server when an asterisk based 
employee attempts an outside call?


We want an employee to pick up an asterisk line and dial outside the 
company.


--Sterling.






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Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

please post also your extensions.conf !;; inbound group;[inbound]; Giorgio test:pmp_l1_check=no;msns=*This Line must be uncomment for sure, of course you may match only specific msns, but for now keep it msns=*
msns=*; end testports = 1context = outbound_isdnOn 9/6/06, Giorgio Incantalupo 
[EMAIL PROTECTED] wrote:Hi Marco,in attachment you can find my 
misdn.conf. Consider that I'm still fixingsome warning because I recently upgraded from install-misdn toinstall-misdn-mqueue but the driver installation manual has not changed.Some comments are due to the fact I'm still making tests to solve the
incoming calls problem I mentioned.Thank you.Giorgio IncantalupoMarco Mouta wrote: Please post your *misdn*-*init*.*conf as well as misdn.conf so i can try to help u*
 On 9/5/06, *Giorgio Incantalupo* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 wrote: Hi, I hava an Asterisk box with a monoBRI + install-misdn-mqueue 0.3.1-rc23 package installed. I can make outbound calls but cannot receive any. I get no Asterisk
 messages on the console except for these: P[ 1] GOT IGNORE SETUP P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE] P[ 1] release_chan: Ch not found! Is there anybody who can help me, please?
 TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by 
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http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta 
 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users[general]debug = 5tracefile = /var/log/asterisk/misdn.trace
trace_calls = falsetrace_dir = /var/log/asterisk/misdnbridging = yesstop_tone_after_first_digit = yesappend_digits2exten = yesl1_info_ok = yesclear_l3 = nomethod = standard;;; CRYPTION STUFF
dynamic_crypt = nocrypt_prefix = **crypt_keys = test,muh; users sections:[default]context = misdnlanguage = itnationalprefix = 0internationalprefix = 00rxgain = 0txgain = 0
te_choose_channel = nodialplan = 0use_callingpres = yesechocancelwhenbridged = noechotraining = yes;; inbound group;[inbound]; Giorgio test:pmp_l1_check=no;msns=*; end test
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[asterisk-users] sangoma A104d echo canceller and fax

2006-09-06 Thread Klaus Darilion


Hi!

I have a sangoma A104d and have some questions about the echo canceller. 
Maybe someone can answer by questions.


1. EC can be activated with wancfg and in /etc/asterisk/zapata.conf. 
What is the difference? I have some suspicions - can someone confirm or 
correct this?


  wancfg: hw echo cancel = yes
  zapata.conf: hw echo cancel = yes
chan_zap activates an echo canceller. As there is an hw echo canceller, 
this is used instead of the SW canceller


  wancfg: hw echo cancel = no
  zapata.conf: hw echo cancel = yes
chan_zap activates an echo canceller. As the hw echo canceller is 
disabled the software EC is used


  wancfg: hw echo cancel = yes/no
  zapata.conf: hw echo cancel = no
chan_zap does not activates any echo canceller.

Is this correct?

2. During Fax transmissions the EC should be disabled. IIRC there is a 
deubg message similar to fax tone detected, disabling echo canceller 
 I made some tests with Fax and activated EC. In log messages I see 
that the EC is activated as soon as the call is connected, but there is 
no log message that a fax is detected and the EC is disabled.


How can I disable the EC during fax transmissions?

Thanks
Klaus
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[asterisk-users] flag 'g' in Dail() is'nt working with agentcallbacklogin()

2006-09-06 Thread umar tarar

here i've come accross another different frustrating prob,  that is; when ever i use flag 'g' in dial() to dial an agent through 
*'s ACD queue system (the agent for which has already been logged in through agentcallbacklogin() ) , it simply does'nt execute any apps below 
dial() but when ever i dial any normal sip-user with 'g' flag in dial(), it always works fine my dial plan for agentcallbacklogin is: 

-extensions.conf--...
[agents] ... exten = 100 , 1 , Wait() exten = 100 , 2 , AgentCallbackLogin(100, s ,[EMAIL PROTECTED]) exten = 500 , 1 , Answer() exten = 500 , 2 , Playback(connecting)
 exten = 500 , 3 , Dial(SIP/100,,g) exten = 500 , 4 , Playback(vm-goodbye) exten = 500 , 5 , HangUp() 
the control never goes to priority 4  5 when some one dials the queue which the agent-100 is a memeber of,... when the 
dial() terminates, the execution terminates simply but in the following smple dial plan, it works just fine; ... exten = 123, 1 , Answer() exten = 123, 2 , Dial(SIP/555,,g) 
exten = 123, 3, layback(vm-goodbye) exten = 123, 4 , HangUp() where 555 is a normal sip-user that is just registered with * anyone plz help me or suggest me something abt why 'g' is not workin in the above case, or suggest any other way to get down to 
Dial() app and execute the rest
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Re: [asterisk-users] core dumps

post your tail /var/log/asterisk/fullOn 9/6/06, Anthony Musaluke [EMAIL PROTECTED] wrote:
Hell all,I have asterisk dying and restarting. After it dies, it creates a core file.Running gdb on this file produces the details below. This is very random andI am not sure what could be happening. Any ideas where I need to look?
ThanksAnthony--(gdb) bt full#00x00881402 in __kernel_vsyscall ()No symbol table info available.#10x001d38f8 in raise () from /lib/libc.so.6
No symbol table info available.#20x001d5068 in abort () from /lib/libc.so.6No symbol table info available.#30x00208a0a in __libc_message () from /lib/libc.so.6No symbol table info available.#40x0020f8ca in _int_malloc () from /lib/libc.so.6
No symbol table info available.#50x002109b9 in calloc () from /lib/libc.so.6No symbol table info available.#60x00727051 in sip_alloc (callid=0x0, sin=0x0, useglobal_nat=0,intended_method=3) at chan_sip.c:3076
p = (struct sip_pvt *) 0x20e4ad__PRETTY_FUNCTION__ = sip_alloc#70x0073100e in sip_poke_peer (peer=0x8b19a60) at chan_sip.c:11629__PRETTY_FUNCTION__ = sip_poke_peer
#80x0073dd87 in sip_poke_peer_s (data="" at chan_sip.c:5761No locals.#90x08056778 in ast_sched_runq (con=0x8b251f8) at sched.c:373x = 0res = 0#10 0x00747ddf in do_monitor (data="" at chan_sip.c:11523
res = 0sip = Variable sip is not available.--___
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[asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...

2006-09-06 Thread Tony Mountifield

In article [EMAIL PROTECTED],
Jason Parker [EMAIL PROTECTED] wrote:
 To expand on what Eric said..  People commonly use _X. for what you're 
 wanting.  It's just
 as effective, but doesn't match the special extensions.

The only catch with _X. is that it doesn't match a single-digit extension.
To do so, you need to duplicate the entries with _X

It's a pity that . means one-or-more-chars rather than zero-or-more-chars.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] IAX and rsa

2006-09-06 Thread picciuX

try adding a username setting in your friends: auth is username/key, not only key.Keep in mind that username in peers and friends is valid only for outbound, 
i.e. it is used for outbound authentication, no inbound. When you receive a call, the incoming username is matched with the [.] part of the peer/friend definition.In other words, username is for peers, not for users. So when used in a friend, username is used only when that friend is acting as a peer, 
i.e. when you're calling it.Be sure also to put private and public keys in correct directory (/var/lib/asterisk/keys, as I remember).In your conf:iax.conf on box 1[asterisk2]type=friend
username=asterisk1 ; === add this: your outbound usernamecontext=mainauth=rsainkey=asterisk2.mydomain.comoutkey=
asterisk1.mydomain.comhost=asterisk2.mydomain.comextensions.conf looks like this:exten = _XX.,1,Dial(IAX2/asterisk2/${EXTEN})iax on box 2
[asterisk1]type=friendcontext=mainusername=asterisk2  ; === add this: your outbound username
auth=rsainkey=asterisk1.mydomain.comoutkey=asterisk2.mydomain.comhost=asterisk1.mydomain.com
extensions.conf looks like thisexten = _XX.,1,Dial(IAX2/asterisk1/${EXTEN})Hope this helps05 Sep 2006 20:37:40 +0200, andrutto 
[EMAIL PROTECTED]:HiI am tyring to connect two * boxes over IAX with rsa, but I am having a slight problem. It just doesn't work. My configuration looks like this:
iax.conf on box 1[asterisk2]type=friendcontext=mainauth=rsainkey=asterisk2.mydomain.comoutkey=asterisk1.mydomain.com
host=asterisk2.mydomain.comextensions.conf looks like this:exten = _XX.,1,Dial(IAX2/asterisk2/${EXTEN})iax on box 2[asterisk1]
type=friendcontext=mainauth=rsainkey=asterisk1.mydomain.comoutkey=asterisk2.mydomain.comhost=
asterisk1.mydomain.comextensions.conf looks like thisexten = _XX.,1,Dial(IAX2/asterisk1/${EXTEN})I generated the key with astgenkey -n asterisk1.mydoamin.com
 on box 1 and astgenkey -n asterisk2.mydomain.com on box 2. I have also exchanged the .pub files between the servers.When I try to call, I can see on a console that the call is not authenticated.
I know I did something wrong (but what?). Is it possible to have rsa authentication with type=friend? Any help would be appreciated.CheersAndrutto--
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Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages


Hi Marco,
I have not a normal extensions.conf:
[outbound_isdn]

include = parkedcalls

exten = _X.,1,DeadAGI(exten2.py)
exten = _X.,2,Hangup

exten = s,1,DeadAGI(exten2.py)
exten = s,2,Hangup

The problem is I do not see the usual coloured output on Asterisk 
console (-- Executing DeadAGI(SIP/8-1d1a, exten2.py) in new stack) 
even with verbose 100. It seems like extensions.conf is not considered.



I looked for msns parameter on internet but it is not very clear what it 
means...is it for inbound or outbound calls?



Thank You


Giorgio Incantalupo



Marco Mouta wrote:

please post also your extensions.conf !
;
; inbound group
;
[inbound]
; Giorgio test:
pmp_l1_check=no
;msns=*

This Line must be uncomment for sure, of course you may match only 
specific msns, but for now keep it msns=*


msns=*

; end test
ports = 1
context = outbound_isdn


On 9/6/06, *Giorgio Incantalupo*  [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Hi Marco,
in attachment you can find my misdn.conf. Consider that I'm still
fixing
some warning because I recently upgraded from install-misdn to
install-misdn-mqueue but the driver installation manual has not
changed.
Some comments are due to the fact I'm still making tests to solve the
incoming calls problem I mentioned.


Thank you.


Giorgio Incantalupo




Marco Mouta wrote:
 Please post your *misdn*-*init*.*conf as well as misdn.conf so i can
 try to help u*

 On 9/5/06, *Giorgio Incantalupo* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

 Hi,
 I hava an Asterisk box with a monoBRI + install-misdn-mqueue
 0.3.1-rc23
 package installed.
 I can make outbound calls but cannot receive any. I get no
Asterisk
 messages on the console except for these:

 P[ 1] GOT IGNORE SETUP
 P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE]
 P[ 1] release_chan: Ch not found!

 Is there anybody who can help me, please?


 TIA


 Giorgio Incantalupo
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 --
 Com os melhores cumprimentos,

 Marco Mouta




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[general]
debug = 5
tracefile = /var/log/asterisk/misdn.trace
trace_calls = false
trace_dir = /var/log/asterisk/misdn
bridging = yes
stop_tone_after_first_digit = yes
append_digits2exten = yes
l1_info_ok = yes
clear_l3 = no
method = standard

;;; CRYPTION STUFF

dynamic_crypt = no
crypt_prefix = **
crypt_keys = test,muh

; users sections:

[default]
context = misdn
language = it
nationalprefix = 0
internationalprefix = 00
rxgain = 0
txgain = 0
te_choose_channel = no
dialplan = 0
use_callingpres = yes
echocancelwhenbridged = no
echotraining = yes

;
; inbound group
;
[inbound]
; Giorgio test:
pmp_l1_check=no
;msns=*
; end test
ports = 1
context = outbound_isdn




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Marco Mouta


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[asterisk-users] mobile refusing call

2006-09-06 Thread René Enskat [Teamware GmbH]




Hi
list,

I have a
problem.
I have an asterisk
-- Cisco Pots gateway.
The problem is when
i call via sip over the asterisk over the pots GW to a mobile phone and refuse th ecall on this mobile the sip phone is
still ringing.
it seems the cisco gw se on th eone site
that the call ist busy/refused but on the gw-sip side the cal is still
active!

somebody has a
solution or hint for me?

Thx!
regards
rene

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Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

msns is as far as i know, similar to DIDs but it includes the complete Dialed number (the number your customer has dialed to call you).put msns=*also test this:[outbound_isdn]exten= _X.,1,Answer()
exten = _X.,n,Playback(vm-goodbye)exten = _X.,n,Hangupexten= s,1,Answer()
exten = s,n,Playback(vm-goodbye)
exten = s,n,HangupEnable an higher debug level for misdn messages in misdn.conf (I think is this the file).Pls post your results Asterisk CLI.On 9/6/06, 
Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi Marco,I have not a normal extensions.conf:[outbound_isdn]include = parkedcallsexten = _X.,1,DeadAGI(exten2.py)exten = _X.,2,Hangupexten = s,1,DeadAGI(exten2.py)exten = s,2,Hangup
The problem is I do not see the usual coloured output on Asteriskconsole (-- Executing DeadAGI(SIP/8-1d1a, exten2.py) in new stack)even with verbose 100. It seems like extensions.conf
 is not considered.I looked for msns parameter on internet but it is not very clear what itmeans...is it for inbound or outbound calls?Thank YouGiorgio IncantalupoMarco Mouta wrote:
 please post also your extensions.conf ! ; ; inbound group ; [inbound] ; Giorgio test: pmp_l1_check=no ;msns=* This Line must be uncomment for sure, of course you may match only
 specific msns, but for now keep it msns=* msns=* ; end test ports = 1 context = outbound_isdn On 9/6/06, *Giorgio Incantalupo*  
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marco, in attachment you can find my misdn.conf
. Consider that I'm still fixing some warning because I recently upgraded from install-misdn to install-misdn-mqueue but the driver installation manual has not changed. Some comments are due to the fact I'm still making tests to solve the
 incoming calls problem I mentioned. Thank you. Giorgio Incantalupo Marco Mouta wrote:  Please post your *misdn*-*init*.*conf as well as 
misdn.conf so i can  try to help u*   On 9/5/06, *Giorgio Incantalupo* [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED]  mailto:[EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote:   Hi,  I hava an Asterisk box with a monoBRI + install-misdn-mqueue  0.3.1-rc23  package installed.
  I can make outbound calls but cannot receive any. I get no Asterisk  messages on the console except for these:   P[ 1] GOT IGNORE SETUP
  P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE]  P[ 1] release_chan: Ch not found!   Is there anybody who can help me, please?  
  TIAGiorgio Incantalupo  ___  --Bandwidth and Colocation provided by 
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http://Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit: 
http://lists.digium.com/mailman/listinfo/asterisk-users  [general] debug = 5 tracefile = /var/log/asterisk/misdn.trace trace_calls = false
 trace_dir = /var/log/asterisk/misdn bridging = yes stop_tone_after_first_digit = yes append_digits2exten = yes l1_info_ok = yes clear_l3 = no method = standard
 ;;; CRYPTION STUFF dynamic_crypt = no crypt_prefix = ** crypt_keys = test,muh ; users sections: [default] context = misdn
 language = it nationalprefix = 0 internationalprefix = 00 rxgain = 0 txgain = 0 te_choose_channel = no dialplan = 0 use_callingpres = yes
 echocancelwhenbridged = no echotraining = yes ; ; inbound group ; [inbound] ; Giorgio test: pmp_l1_check=no ;msns=*
 ; end test ports = 1 context = outbound_isdn ___ --Bandwidth and Colocation provided by 
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Re: [asterisk-users] Budgetones - multiple phones losing IP address during day

2006-09-06 Thread Rob Lith

GarthSounds like a DHCP lease issue that the BT102's are not playing nicely with?RegardsRobOn 06/09/06, Brandon Galbraith 
[EMAIL PROTECTED] wrote:Garth,Are they all on the same switch? Possibly could be a network-level issue, and not something wrong with the phones.
-brandonOn 9/6/06, 
Garth van Sittert [EMAIL PROTECTED] wrote:

Hi AllI have a site with 50 Budgetone 102's and about 5 snom phones.At random intervals during the day about 20 or 30 of the Budgetones losetheir connection to the network all at the same time.It happens about
once a day.The Snom phones are fine and never get disconnected.Ican't ping the Budgetones IP's and the way to fix them is to simplyunplug and reconnect.Nothing interesting shows up in the logs even in

debug mode except for the 'Peer XXX is now unreachable' repeated foreach extension.I haven't had much experience with the Budgetones.Does anyone have anyidea what could be causing this?Thanks

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[EMAIL PROTECTED]AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost

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Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages


Hi Marco,
it seems that msns=* is necessary  to make Asterisk work correctly...I 
do not why...
We have another PBX with a monoBRI but have not this problem, maybe is 
the different ISDN telco or the old misdn driver does not 
complainthe important is that now it works!!
I have only to fix some warning (I hope for an update of the beronet 
manual) and it seems all right.



Thank you again for help!!





Marco Mouta wrote:
msns is as far as i know, similar to DIDs but it includes the complete 
Dialed number (the number your customer has dialed to call you).


put msns=*

also test this:
[outbound_isdn]
exten= _X.,1,Answer()
exten = _X.,n,Playback(vm-goodbye)
exten = _X.,n,Hangup

exten= s,1,Answer()
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup


Enable an higher debug level for misdn messages in misdn.conf (I think 
is this the file).


Pls post your results Asterisk CLI.

On 9/6/06, * Giorgio Incantalupo* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Hi Marco,
I have not a normal extensions.conf:
[outbound_isdn]

include = parkedcalls

exten = _X.,1,DeadAGI(exten2.py)
exten = _X.,2,Hangup

exten = s,1,DeadAGI(exten2.py)
exten = s,2,Hangup

The problem is I do not see the usual coloured output on Asterisk
console (-- Executing DeadAGI(SIP/8-1d1a, exten2.py) in new stack)
even with verbose 100. It seems like extensions.conf is not
considered.


I looked for msns parameter on internet but it is not very clear
what it
means...is it for inbound or outbound calls?


Thank You


Giorgio Incantalupo



Marco Mouta wrote:
 please post also your extensions.conf !
 ;
 ; inbound group
 ;
 [inbound]
 ; Giorgio test:
 pmp_l1_check=no
 ;msns=*

 This Line must be uncomment for sure, of course you may match only
 specific msns, but for now keep it msns=*

 msns=*

 ; end test
 ports = 1
 context = outbound_isdn


 On 9/6/06, *Giorgio Incantalupo*  [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

 Hi Marco,
 in attachment you can find my misdn.conf . Consider that I'm
still
 fixing
 some warning because I recently upgraded from install-misdn to
 install-misdn-mqueue but the driver installation manual has not
 changed.
 Some comments are due to the fact I'm still making tests to
solve the
 incoming calls problem I mentioned.


 Thank you.


 Giorgio Incantalupo




 Marco Mouta wrote:
  Please post your *misdn*-*init*.*conf as well as
misdn.conf so i can
  try to help u*
 
  On 9/5/06, *Giorgio Incantalupo*
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 mailto: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
 mailto: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
 
  Hi,
  I hava an Asterisk box with a monoBRI +
install-misdn-mqueue
  0.3.1-rc23
  package installed.
  I can make outbound calls but cannot receive any. I get no
 Asterisk
  messages on the console except for these:
 
  P[ 1] GOT IGNORE SETUP
  P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE]
  P[ 1] release_chan: Ch not found!
 
  Is there anybody who can help me, please?
 
 
  TIA
 
 
  Giorgio Incantalupo
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  Com os melhores cumprimentos,
 
  Marco Mouta
 





 
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 [general]
 debug = 5
 tracefile = /var/log/asterisk/misdn.trace
 trace_calls = false
 trace_dir = /var/log/asterisk/misdn
 bridging = yes
 stop_tone_after_first_digit

[asterisk-users] how to setup poxy sip server

2006-09-06 Thread Ranjeet Kumar









Hi,



I want to setup proxy sip server. In case of asterisk I dont
know how to do that. 

Please help me doing this.



Thanks,

Ranjeet










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[asterisk-users] Answer Machine detection

2006-09-06 Thread Mark Ackroyd

I use Asterisk mainly in the IVR world and sometimes do a few outbound based
campaigns. The horrible subject of Answer Machine Detection as lifted its
head again. To my knowledge there are 3 ways to deal with it.

1) When the call is answered, you please a message something like There is 
   an important message for you please press 1 and you detect the 
   Live/Machine state from that.

2) You listen for the Hello? using backgrounddetect then a pause. If there

   is little or no pause assume it's an answer machine else it's a live

3) You listen for the tone or beep given off by the answer machine at the 
   end of the announcement.


At the moment prefer 1, it's easy and quick to implement.  Using method 2 is
tricky you need to play with various settings and I only usually get a 60%
success rate with it.  Option 3, I have never seen an asterisk based
solution for this one, or know that they exist. 

Does anyone have any snippets or even a commercial solution to get option 2
success rate to the 90% mark? or anyone know of some that has an asterisk
solution for option 3?.  Failing that is there an option 4 or 5?

Thanks,
Mark













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Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

Hi,Multiple Subscriber Number. This is a telephone number
  associated with an ETS 300 BRI line. Providers
  of ETS 300 often give you three MSNs with a BRI, although additional MSNs can
  be purchased. An ISDN terminal will ring (provide an alerting
  signal) only when calls are made to the MSN (or MSNs) entered in that
  terminal. If a terminal has no MSNs entered it will ring whenever
  there is a call to any of the MSN's on that BRI.You can have specific ports of your Beronet card handling specific MSNs, and then route it to diferent contexts into Asterisk, like handling two companies in one asterisk server.
Got it?Best regardsOn 9/6/06, Giorgio Incantalupo 
[EMAIL PROTECTED] wrote:Hi Marco,it seems that msns=* is necessaryto make Asterisk work correctly...I
do not why...We have another PBX with a monoBRI but have not this problem, maybe isthe different ISDN telco or the old misdn driver does notcomplainthe important is that now it works!!I have only to fix some warning (I hope for an update of the beronet
manual) and it seems all right.Thank you again for help!!Marco Mouta wrote: msns is as far as i know, similar to DIDs but it includes the complete Dialed number (the number your customer has dialed to call you).
 put msns=* also test this: [outbound_isdn] exten= _X.,1,Answer() exten = _X.,n,Playback(vm-goodbye) exten = _X.,n,Hangup exten= s,1,Answer()
 exten = s,n,Playback(vm-goodbye) exten = s,n,Hangup Enable an higher debug level for misdn messages in misdn.conf (I think is this the file). Pls post your results Asterisk CLI.
 On 9/6/06, * Giorgio Incantalupo* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 wrote: Hi Marco, I have not a normal extensions.conf: [outbound_isdn] include = parkedcalls exten = _X.,1,DeadAGI(exten2.py
) exten = _X.,2,Hangup exten = s,1,DeadAGI(exten2.py) exten = s,2,Hangup The problem is I do not see the usual coloured output on Asterisk console (-- Executing DeadAGI(SIP/8-1d1a, 
exten2.py) in new stack) even with verbose 100. It seems like extensions.conf is not considered. I looked for msns parameter on internet but it is not very clear
 what it means...is it for inbound or outbound calls? Thank You Giorgio Incantalupo Marco Mouta wrote:  please post also your 
extensions.conf !  ;  ; inbound group  ;  [inbound]  ; Giorgio test:  pmp_l1_check=no  ;msns=* 
  This Line must be uncomment for sure, of course you may match only  specific msns, but for now keep it msns=*   msns=*   ; end test
  ports = 1  context = outbound_isdnOn 9/6/06, *Giorgio Incantalupo*  [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]  mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:   Hi Marco,  in attachment you can find my misdn.conf
 . Consider that I'm still  fixing  some warning because I recently upgraded from install-misdn to  install-misdn-mqueue but the driver installation manual has not
  changed.  Some comments are due to the fact I'm still making tests to solve the  incoming calls problem I mentioned.  
  Thank you.Giorgio Incantalupo  Marco Mouta wrote:   Please post your *misdn*-*init*.*conf as well as
 misdn.conf so i can   try to help u* On 9/5/06, *Giorgio Incantalupo* 
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  mailto: [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]   mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]  mailto: [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote: Hi,   I hava an Asterisk box with a monoBRI +
 install-misdn-mqueue   0.3.1-rc23   package installed.   I can make outbound calls but cannot receive any. I get no  Asterisk
   messages on the console except for these: P[ 1] GOT IGNORE SETUP   P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE]
   P[ 1] release_chan: Ch not found! Is there anybody who can help me, please?   TIA
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http://lists.digium.com/mailman/listinfo/asterisk-users   --   Com os melhores cumprimentos,
 Marco Mouta 
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   [general]  debug = 5  tracefile = /var/log/asterisk/misdn.trace  trace_calls = false
  trace_dir = /var/log/asterisk/misdn  bridging = yes  stop_tone_after_first_digit = yes  append_digits2exten = yes  l1_info_ok = yes
  clear_l3 = no  method = standard   ;;; CRYPTION STUFF   dynamic_crypt = no  crypt_prefix = **
  crypt_keys = test,muh   ; users sections:   [default]  context = misdn  language = it

Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

Also your problem could be related with the Answer() you weren't answering the calls on your previous extensions.confPls test both configs with and without answer and reply your results.
On 9/6/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,Multiple Subscriber Number. This is a telephone number
  associated with an ETS 300 BRI line. Providers
  of ETS 300 often give you three MSNs with a BRI, although additional MSNs can
  be purchased. An ISDN terminal will ring (provide an alerting
  signal) only when calls are made to the MSN (or MSNs) entered in that
  terminal. If a terminal has no MSNs entered it will ring whenever
  there is a call to any of the MSN's on that BRI.You can have specific ports of your Beronet card handling specific MSNs, and then route it to diferent contexts into Asterisk, like handling two companies in one asterisk server.
Got it?Best regardsOn 9/6/06, Giorgio Incantalupo
 
[EMAIL PROTECTED] wrote:Hi Marco,it seems that msns=* is necessaryto make Asterisk work correctly...I
do not why...We have another PBX with a monoBRI but have not this problem, maybe isthe different ISDN telco or the old misdn driver does notcomplainthe important is that now it works!!I have only to fix some warning (I hope for an update of the beronet
manual) and it seems all right.Thank you again for help!!Marco Mouta wrote: msns is as far as i know, similar to DIDs but it includes the complete Dialed number (the number your customer has dialed to call you).
 put msns=* also test this: [outbound_isdn] exten= _X.,1,Answer() exten = _X.,n,Playback(vm-goodbye) exten = _X.,n,Hangup exten= s,1,Answer()
 exten = s,n,Playback(vm-goodbye) exten = s,n,Hangup Enable an higher debug level for misdn messages in misdn.conf (I think is this the file). Pls post your results Asterisk CLI.
 On 9/6/06, * Giorgio Incantalupo* [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED]
 wrote: Hi Marco, I have not a normal extensions.conf: [outbound_isdn] include = parkedcalls exten = _X.,1,DeadAGI(exten2.py

) exten = _X.,2,Hangup exten = s,1,DeadAGI(exten2.py) exten = s,2,Hangup The problem is I do not see the usual coloured output on Asterisk console (-- Executing DeadAGI(SIP/8-1d1a, 
exten2.py) in new stack) even with verbose 100. It seems like extensions.conf is not considered. I looked for msns parameter on internet but it is not very clear

 what it means...is it for inbound or outbound calls? Thank You Giorgio Incantalupo Marco Mouta wrote:  please post also your 
extensions.conf !  ;  ; inbound group  ;  [inbound]  ; Giorgio test:  pmp_l1_check=no  ;msns=* 
  This Line must be uncomment for sure, of course you may match only  specific msns, but for now keep it msns=*   msns=*   ; end test
  ports = 1  context = outbound_isdnOn 9/6/06, *Giorgio Incantalupo*  
[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]  mailto:
[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:   Hi Marco,
  in attachment you can find my misdn.conf
 . Consider that I'm still  fixing  some warning because I recently upgraded from install-misdn to  install-misdn-mqueue but the driver installation manual has not
  changed.  Some comments are due to the fact I'm still making tests to solve the  incoming calls problem I mentioned.  
  Thank you.Giorgio Incantalupo  Marco Mouta wrote:   Please post your *misdn*-*init*.*conf as well as
 misdn.conf so i can   try to help u* On 9/5/06, *Giorgio Incantalupo* 

[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  mailto: 
[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]   mailto:
[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]  mailto: 
[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:  
   Hi,   I hava an Asterisk box with a monoBRI +
 install-misdn-mqueue   0.3.1-rc23   package installed.   I can make outbound calls but cannot receive any. I get no  Asterisk
   messages on the console except for these: P[ 1] GOT IGNORE SETUP   P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE]

   P[ 1] release_chan: Ch not found! Is there anybody who can help me, please?   TIA
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http://lists.digium.com/mailman/listinfo/asterisk-users   --   Com os melhores cumprimentos,
 Marco Mouta 

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   [general]  debug = 5  tracefile = /var/log/asterisk/misdn.trace  trace_calls = false
  trace_dir = /var/log/asterisk/misdn  bridging = yes

RE: [asterisk-users] how to setup poxy sip server

2006-09-06 Thread brian

Hi Ranjeet,If you just want a SIP proxy that can route SIP messages without taking care of media flows, you should consider using SER or OpenSer. Only use Asterisk is a PBX, so use it when you need PBX's functionalities like voicemail, IVR ... You can find documentation about Asterisk's configuration on this site www.voip-info.orgRegards,Brianwww.neotiq.com

Hi,

I want to setup proxy sip server. In case of asterisk I don’t
know how to do that.

Please help me doing this.

Thanks,

Ranjeet

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Re: [asterisk-users] Answer Machine detection

2006-09-06 Thread Matt Florell


We use app_amd in the VICIDIAL project and it works fairly well(better
than 60% if you tune it right), although I still don't recommend that
anyone use Answering Machine detection.
http://bugs.digium.com/view.php?id=5959

When we usually do automated notification campaigns we will most often
wait 2 seconds from Answer and then play the message twice(depending
on it's length). This gets the message on close to 100% of the
answering machine/voicemail even if it starts in the middle of the
first play it is played again and we have had rather good response
from this.

MATT---

On 9/6/06, Mark Ackroyd [EMAIL PROTECTED] wrote:

I use Asterisk mainly in the IVR world and sometimes do a few outbound based
campaigns. The horrible subject of Answer Machine Detection as lifted its
head again. To my knowledge there are 3 ways to deal with it.

1) When the call is answered, you please a message something like There is
   an important message for you please press 1 and you detect the
   Live/Machine state from that.

2) You listen for the Hello? using backgrounddetect then a pause. If there

   is little or no pause assume it's an answer machine else it's a live

3) You listen for the tone or beep given off by the answer machine at the
   end of the announcement.


At the moment prefer 1, it's easy and quick to implement.  Using method 2 is
tricky you need to play with various settings and I only usually get a 60%
success rate with it.  Option 3, I have never seen an asterisk based
solution for this one, or know that they exist.

Does anyone have any snippets or even a commercial solution to get option 2
success rate to the 90% mark? or anyone know of some that has an asterisk
solution for option 3?.  Failing that is there an option 4 or 5?

Thanks,
Mark













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Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages


Hi Marco,
unfortunately I cannot make any further test on that machine because it 
has already been delivered to the customer but I can tell you that its 
extensions.conf is the same of our production PBX which has not that 
problem. The only differences are the telco and the ISDN driver.

If you want I can search for old driver version if you still use it.
Let me know, maybe can be useful if you deploy PBX in Italy.



Giorgio Incantalupo





Marco Mouta wrote:
Also your problem could be related with the Answer() you weren't 
answering the calls on your previous extensions.conf


Pls test both configs with and without answer and reply your results.



On 9/6/06, *Marco Mouta* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Hi,

Multiple Subscriber Number. This is a telephone number associated
with an *ETS 300
http://www.telos-systems.com/techtalk/gldefs.htm#ETS%20300* BRI
line. Providers of ETS 300 often give you three MSNs with a BRI,
although additional MSNs can be purchased. An ISDN terminal will
ring (provide an alerting signal) only when calls are made to
the MSN (or MSNs) entered in that terminal. If a terminal has no
MSNs entered it will ring whenever there is a call to any of the
MSN's on that BRI.

You can have specific ports of your Beronet card handling specific
MSNs, and then route it to diferent contexts into Asterisk, like
handling two companies in one asterisk server.

Got it?


Best regards





On 9/6/06, *Giorgio Incantalupo*  [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

Hi Marco,
it seems that msns=* is necessary  to make Asterisk work
correctly...I
do not why...
We have another PBX with a monoBRI but have not this problem,
maybe is
the different ISDN telco or the old misdn driver does not
complainthe important is that now it works!!
I have only to fix some warning (I hope for an update of the
beronet
manual) and it seems all right.


Thank you again for help!!





Marco Mouta wrote:
 msns is as far as i know, similar to DIDs but it includes the
complete
 Dialed number (the number your customer has dialed to call you).

 put msns=*

 also test this:
 [outbound_isdn]
 exten= _X.,1,Answer()
 exten = _X.,n,Playback(vm-goodbye)
 exten = _X.,n,Hangup

 exten= s,1,Answer()
 exten = s,n,Playback(vm-goodbye)
 exten = s,n,Hangup


 Enable an higher debug level for misdn messages in misdn.conf
(I think
 is this the file).

 Pls post your results Asterisk CLI.

 On 9/6/06, * Giorgio Incantalupo*
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
 mailto: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

 Hi Marco,
 I have not a normal extensions.conf:
 [outbound_isdn]

 include = parkedcalls

 exten = _X.,1,DeadAGI(exten2.py )
 exten = _X.,2,Hangup

 exten = s,1,DeadAGI(exten2.py)
 exten = s,2,Hangup

 The problem is I do not see the usual coloured output on
Asterisk
 console (-- Executing DeadAGI(SIP/8-1d1a,  exten2.py)
in new stack)
 even with verbose 100. It seems like extensions.conf is not
 considered.


 I looked for msns parameter on internet but it is not
very clear
 what it
 means...is it for inbound or outbound calls?


 Thank You


 Giorgio Incantalupo



 Marco Mouta wrote:
  please post also your extensions.conf !
  ;
  ; inbound group
  ;
  [inbound]
  ; Giorgio test:
  pmp_l1_check=no
  ;msns=*
 
  This Line must be uncomment for sure, of course you may
match only
  specific msns, but for now keep it msns=*
 
  msns=*
 
  ; end test
  ports = 1
  context = outbound_isdn
 
 
  On 9/6/06, *Giorgio Incantalupo* 
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
  mailto: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
 
  Hi Marco,
  in attachment you can find my misdn.conf . Consider
that I'm
 still
  fixing
  some warning because I

[asterisk-users] cmd SET time value

2006-09-06 Thread Benjamin Jacob


Hello ppl,

Ive a couple of macros defined to call fwd based on time to a 
number/voicemail.

Very elementary.

=
11. [macro-dialexten]
12. exten = s,1,Dial(SIP/${ARG1})   ;

1. [macro-stdpbx1exten]
2. exten = s,1,Set(fwdedNum=${DB(CFWD/${ARG1})})

3. exten = s,n,GotoIf(${fwdedNum}?getFwdTime:dialExten)

4. exten = s,n(getFwdTime),Set(fwdTime=${DB(CFWDTime/${ARG1})}) ;

5. exten = s,n,GotoIf(${fwdedNum} != 
VoiceMail?s-dialFwdTime,1:s-vmFwdTime,1) ; goto VoiceMail or dial 
Fwded num


6. exten = s,n(dialExten),Macro(dialexten,${ARG1}) ; dial Called exten

7. exten = 
s-vmFwdTime,1,GotoIfTime(${fwdTime}?s-vmFwdTime,vmFwd:s,dialExten) ;if 
fwdTime not set or time matches,
   
;send to VM, else dialExten

8. exten = s-vmFwdTime,n(vmFwd),VoiceMail(${ARG1})

9. exten = 
s-dialFwdTime,1,GotoIfTime(${fwdTime}?s-dialFwdTime,dialFwd:s,dialExten) 
;if fwdTime not set or time matches,
 
; call fwdedNum, else dialExten

10. exten = s-dialFwdTime,n(dialFwd),Macro(dialexten,${fwdedNum})

===

I save the fwdedNum in DB, and also the fwding time.
Now, when i retrieve the time value from db and set it, using cmd SET, 
it takes only the initial part of the time value string.

e.g. if time to be checked is *|mon-tue|*|*, the time set is * ONLY!!

The cmd Set's syntax uses the | (pipe) notation to separate variables. 
Thats why this behaviour.

Any work around this guys??

Thanks in advance

Ben.

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Re: [asterisk-users] macros in Realtime

2006-09-06 Thread Benjamin Jacob


Thanks Simon,
will try and get back  on this one .

Ben.

Simon Woodhead wrote:


Hi Ben,

Yes it is but you need to remember to still include

[macro-stdpbx1exten]
switch = Realtime/

in your extensions.conf

Simon

On 9/6/06, * Benjamin Jacob* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Hello all,
Another question related to Realtime.
Is it possible to call macros using Realtime arch?

I have a macro definition in table extensions_conf in my MySql db as:
30 | macro-stdpbx1exten | s |
1 | SET| fwdedNum=${DB(CFWD/${ARG1})}

And calling the Macro using another entry in table extensions_conf in
Mysql db:

40 | pbx1   | _[345]. |
1 | Macro  | stdpbx1exten|${EXTEN}

I get errors like :

Sep  6 11:18:53 WARNING[14493]: app_macro.c:154 macro_exec: No such
context 'macro-stdpbx1exten' for macro 'stdpbx1exten'

Are there issues with the same??

TiA

-Ben.
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Re: [asterisk-users] includes in realtime ??

2006-09-06 Thread Benjamin Jacob


lol RR.
will def do some RnD on this one, and wil get back. have put this on the 
back burner for now.


thanks again.

cheerz
Ben

RR wrote:


I use rtcachefriends=yes and any changes I make in my database become
effective immediately along with also getting the MWI functionality.
Even though what you say makes sense. Go figure!

Ben, yeah if it shows it's loaded then it's there for sure. Sorry I
asked for it as in your module listing there wasn't any of these
modules. I'm at the end of the rope on troubleshooting your issue.
Maybe more detail is needed. Esp when you're saying that your sip.conf
general section has just two entries. Where's the rest of it, not that
a lot needs to necessarily be there if you're not doing anything too
tricky. But I would go with removing the rtcache command from the
sip.conf file and try and get realtime working in realtime, if that
doesn't sound too whacked, just in case it's working off of some
cached data, which is why your old codec selection seems to still work
even after you change it.

Have you looked in your asterisk log file (full) to see if its
complaining about errors when you do a realtime load command?  The
only time my realtime load comes back empty is when it's got a
permission problem of some sort on the DB side and one time it
happened because of some delay that was introduced coz of some heavy
logging or something, don't quite remember it.
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Re: [asterisk-users] Asterisk + Samsung OffServ 500

2006-09-06 Thread Garth van Sittert


Eugeniy Khvastunov wrote:
What signaling method i should use for connecting Asterisk(Gentoo, 
Tormenta 2) + Samsung OffServ 500 by PRI flow?

What parametrs in zaptel.conf, zapata.conf?
---
Какой метод сигнализации нужно использовать при подключении 
Asterisk(Gentoo, Tormenta 2) + Samsung OffServ 500 по PRI потоку?

Интересуют параметры zaptel.conf, zapata.conf.
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Hi Eugeniy

You should set the Asterisk configuration as if you are connecting to 
your local Telco provider and set the Samsung to fit Asterisk.


Garth

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Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages


Hi Marco,
found drivers versions   :)

Our PBX working without msns parameter has:
   - ISDN line provided by fastweb italy
   - ISDN driver: install-misdn version 0.2.1-rc13

Other PBX NOT working without msns parameter has:
   - ISDN line provided by telecom italy
   - ISDN driver: install-misdn-mqueue version 0.3.1-rc23

Hope may be useful!!


Giorgio Incantalupo




Marco Mouta wrote:
Also your problem could be related with the Answer() you weren't 
answering the calls on your previous extensions.conf


Pls test both configs with and without answer and reply your results.



On 9/6/06, *Marco Mouta* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Hi,

Multiple Subscriber Number. This is a telephone number associated
with an *ETS 300
http://www.telos-systems.com/techtalk/gldefs.htm#ETS%20300* BRI
line. Providers of ETS 300 often give you three MSNs with a BRI,
although additional MSNs can be purchased. An ISDN terminal will
ring (provide an alerting signal) only when calls are made to
the MSN (or MSNs) entered in that terminal. If a terminal has no
MSNs entered it will ring whenever there is a call to any of the
MSN's on that BRI.

You can have specific ports of your Beronet card handling specific
MSNs, and then route it to diferent contexts into Asterisk, like
handling two companies in one asterisk server.

Got it?


Best regards





On 9/6/06, *Giorgio Incantalupo*  [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

Hi Marco,
it seems that msns=* is necessary  to make Asterisk work
correctly...I
do not why...
We have another PBX with a monoBRI but have not this problem,
maybe is
the different ISDN telco or the old misdn driver does not
complainthe important is that now it works!!
I have only to fix some warning (I hope for an update of the
beronet
manual) and it seems all right.


Thank you again for help!!





Marco Mouta wrote:
 msns is as far as i know, similar to DIDs but it includes the
complete
 Dialed number (the number your customer has dialed to call you).

 put msns=*

 also test this:
 [outbound_isdn]
 exten= _X.,1,Answer()
 exten = _X.,n,Playback(vm-goodbye)
 exten = _X.,n,Hangup

 exten= s,1,Answer()
 exten = s,n,Playback(vm-goodbye)
 exten = s,n,Hangup


 Enable an higher debug level for misdn messages in misdn.conf
(I think
 is this the file).

 Pls post your results Asterisk CLI.

 On 9/6/06, * Giorgio Incantalupo*
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
 mailto: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

 Hi Marco,
 I have not a normal extensions.conf:
 [outbound_isdn]

 include = parkedcalls

 exten = _X.,1,DeadAGI(exten2.py )
 exten = _X.,2,Hangup

 exten = s,1,DeadAGI(exten2.py)
 exten = s,2,Hangup

 The problem is I do not see the usual coloured output on
Asterisk
 console (-- Executing DeadAGI(SIP/8-1d1a,  exten2.py)
in new stack)
 even with verbose 100. It seems like extensions.conf is not
 considered.


 I looked for msns parameter on internet but it is not
very clear
 what it
 means...is it for inbound or outbound calls?


 Thank You


 Giorgio Incantalupo



 Marco Mouta wrote:
  please post also your extensions.conf !
  ;
  ; inbound group
  ;
  [inbound]
  ; Giorgio test:
  pmp_l1_check=no
  ;msns=*
 
  This Line must be uncomment for sure, of course you may
match only
  specific msns, but for now keep it msns=*
 
  msns=*
 
  ; end test
  ports = 1
  context = outbound_isdn
 
 
  On 9/6/06, *Giorgio Incantalupo* 
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
  mailto: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
 
  Hi Marco,
  in attachment you can find my misdn.conf . Consider
that I'm
 still
  fixing
  some warning because I recently upgraded from
install-misdn to

Re: [asterisk-users] How to test TE405P T1

2006-09-06 Thread Garth van Sittert


Andy Chung (Power-All) wrote:

Hi all,

I have connected a T1IDA-P to the Digium TE405P. Checked with the 
Telco, and confirmed the T1 is up and connected. However, I have no 
idea how to test the T1 is really work, because the Asterisk server 
not yet be configure. Anyone has the method on how to test the calls 
through the T1?


Thanks!
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Hi Andy

Asterisk does part of the processing for the Digium cards (that's why 
they are so cheap) so you cannot test the PRI without getting asterisk 
up and running.  The zaptel drivers will get Layer 1 and I think Layer 2 
up, but Asterisk is needed for more than this.


Kind Regards
Garth

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Re: [asterisk-users] Unable to make calls from CallManager to Asterisk

2006-09-06 Thread Gary Richardson

What version of call manager?On 9/5/06, Anantha Padmanabha.M.L [EMAIL PROTECTED] wrote:
HI,I have successfully integrated CallManager and Asterisk and was able to make call from one of Asterisk phone to CallManager 
Phone.But Could not able to make call from CallManager to asterisk.I have also tried the below link :-  
http://www.voip-info.org/wiki/index.php?page=Asterisk+Cisco+CallManager+Integration  But still not able to place calls from CallManager to AsteriskCan anybody send me sample of Configuration that i have to make to make calls from CallManager to Asterisk.
  This is really Urgent for me!!.Thanks in Advance,Anantha 
		Do you Yahoo!? 
Get on board. You're invited to try the new Yahoo! Mail.
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Re: [asterisk-users] Deadlock

2006-09-06 Thread BJ Weschke


On 9/5/06, Michael Welter [EMAIL PROTECTED] wrote:

(I'm getting 404 Not Found from the search engines)

I have a system that gets a deadlock every week or so.  On the logs I
have many channel.c:787 channel_find_locked avoided deadlock for
0x837730 messages.

The system has an Eschelon T1 with 6 voice (with dchan) arriving on a
TE110P.  Asterisk 1.2.7.1, Linux FC5.  The system also has a TDM30B
card.  The phones are IP501.

During the deadlock period, outbound calls are ok.  However, an inbound
call (on channel two) is rejected because Asterisk thinks the channel is
in use.

There are no call queues on this system.  I see a deadlock bug, but it
has to do with queues.

Can someone shed some light on this situation?



Does show channels still show the original channel in its list? If
so, what extension and priority and what app is it at?

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [asterisk-users] SIP client with video???

2006-09-06 Thread Joao Pereira


The problems with X-Lite 3 are:

- just accepts one SIP registration
- doesnt send video to other X-Lite or eyeBeam versions
- sometimes loses the SIP informations when you reboot the PC
.
Regards
Joao Pereira

Blake Krone wrote:

What's wrong with X-Lite 3.0? I haven't had any issues with it and 
find it to be one of the best SIP video software choices, and it's free.


On 7/27/06, *Joao Pereira * [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Hello to all
can someone recommend me a nice SIP client with video for windows??

I tried X-Lite 3.0 but it's a lousy piece of software.

Does someone knows about a better software?
Regards
Joao Pereira

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[asterisk-users] Asterisk video support

2006-09-06 Thread Joao Pereira


Hello to all
I used SER for SIP calls with video, but now Im trying the same in 
Asterisk and It doesnt work.

I m using X-Lite 3.0 (the same that worked with SER).
Do Asterisk needs any special configuration to allow SIP calls with 
video between its clients?


Regards
Joao Pereira

Asterisk's support for video over SIP is very rudimentary. Only to 
video codecs H.261, H.263, and H.263+ are supported, and even then, 
not very well. There is no support for dynamic negotiation of frame 
rates, etc. Queries to the -dev list, as to progress on these features 
were recently met with silence. We will be looking to jump into the 
project to support our own initiatives in the area of video in a few 
weeks.
 
Until things change, your best bet for connecting SIP video phones is 
SER.




 



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Re: [asterisk-users] asterisk t.38 fax failed

2006-09-06 Thread Kokfoo Soo

Ricardo,Thanks, could you please share some of your t.38 passthrough configuration in sip.conf and also udptl.conf?Thanks,Ricardo Carvalho [EMAIL PROTECTED] wrote: No, T.38 doesn't work with Asterisk. Only works with Asterisk t38passthrough patch that you can find at URL: http://bugs.digium.com/file_download.php?file_id=9335type=bugFor me it only worked well with patch for version 1.2.4 of Asterisk.Regards,Ricardo.Kokfoo Soo wrote: Is T.38 fax work through Asterisk? I have the config below in my  sip.conf, but the fax doesn't work and give me the CLI lines below. My  current version is 1.2.10. Please help. [Inboundtopbx] type=friend context=pbx host=10.18.188.84 insecure=port
 dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 allow=ulaw t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no [OutboundfromPBX] type=peer host=10.18.161.222   canreinvite=no dtmfmode=rfc2833 disallow=all allow=g729 qualify=yes t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no -- SIP read from 10.18.188.84:50096: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP  10.18.188.84:5060 From: ;tag=19D429E8-2084 To: ;tag=as3c87a22e Date: Tue, 05 Sep 2006 19:42:28 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK --- (9 headers 0 lines)--- Sep  5 15:30:31 NOTICE[25233]: rtp.c:564
 ast_rtp_read: Unknown RTP  codec 100 received Sep  5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received Sep  5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received Sep  5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received Sep  5 15:30:34 WARNING[6839]: chan_sip.c:3475 process_sdp: Unknown  SDP media type in offer: image 16406 udptl t38  Yahoo! Messenger with Voice. Make PC-to-Phone Callsto the US (and 30+ countries) for 2¢/min or less.  ___
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[asterisk-users] app_rxfax Only Receives One Page

I am trying to setup a fax server and all I get is the first page of a 
multipage fax.  The first page is perfect quality.


I have googled and found people with the same problem but no good 
answers.  The one answer given was to use a multipage tiff viewer which 
I did and confirmed that only one page was received.


I am not sure how to debug this.  I have an HP DL320 with a quad Sangoma 
T1 board.


Any help would be appreciated.

Thanks,
Steve Totaro
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Re: [asterisk-users] Native Chinese speaker needed

2006-09-06 Thread Steve Underwood


Hi,

John Williams wrote:


No, I'm not looking for a voice talent.

I have been deploying an IVR in my company's China office, and our people
there complained about the way asterisk spoke the date in Chinese.

After discussing it with them, I have submitted a patch, which can be
found on the Digium Issue Tracker at
http://bugs.digium.com/view.php?id=7827

I would greatly appreciate it if another native Chinese speaker could
review the proposed change, and comment on whether it is culturally and/or
grammatically correct.  Please add a note to the Issue at the above URL
with your comments.

Here is my description from the Issue:

My contacts in China/Taiwan tell me that the way the day-of-month is done
by asterisk in chinese is wrong. The proper way is to say an cardinal
number followed by the chinese word for day.

The current implementation speaks an ordinal number for the day-of-month,
and does so in a way which is grammatically incorrect.

The current implementation speaks the day-of-month as an ordinal number in
an odd way. For example, 25th is done with the recordings digits/h-20h,
digits/h-5; 17th is digits/h-10h, digits/h-7 instead of using
digits/h-17!

Chinese uses a prefix for ordinal numbers, so a grammatically correct
expression would be h-20, 5. Using h-5 as the second recording
places the prefix in the middle of the number, which is wrong. What
h-20h should contain is not explained any where, and the only readily
available collection of sound files in chinese for asterisk (at
iaxtalk.com) does not contain any of the h-%d files.

The attached patch changes the day-of-month code to do (for example)
digits/20, digits/5, digits/day for the 25th, or digits/10,
digits/7, digits/day for the 17th.
 

Chinese uses a prefix for ordinal numebrs. However ordinal number are 
not used in dates. In Chinese today (its the 6th September here) is 
expressed as

   zero six year nine month six day
You can skip the zero six year part, and just say the month and day. 
You can also use the word for number (not the ordinal prefix) in place 
of the word for day, if you like. A complication is that the second of 
the month uses a word that we might translate as a couple (sounds odd 
to say a couple as a day, but words don't quite trannslate cleanly). So 
the second of September was

  zero six year nine month couple day
but the twelfth will be
  zero six year nine month ten two day
I hope that's not too confusing :-\ Do you want to know about times too? 
To express those in a general colloquial way, ten past two would be

   two time two
Because the minutes part is expressed like the face of the clock, 10 
minutes is expressed as 2. You can say

   two time ten minute
but it sounds a bit stuffy. Notice I didn't use the word for couple in 
that. If I wanted to say 2 minutes after two, I would say

   two time couple minute
Here I did use the couple word, and I expressed a time with greater 
precision than the clock face numbers permit.


Those expressions might sound odd as English, because none of the words 
have a plural ending. That is how Chinese is expressed - no plurals.


Regards,
Steve

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Re: [asterisk-users] asterisk t.38 fax failed

2006-09-06 Thread Ricardo Carvalho

In sip.conf add to [general] context and to every peer context that you 
want to register in Asterisk to use T.38 the following lines:

t38pt_udptl=yes
t38pt_rtp=no
t38pt_tcp=no

In udptl.conf file I have the following configurations:
[general]
udptlstart=4000
udptlend=4999
T38FaxUdpEC = t38UDPRedundancy
T38FaxMaxDatagram = 400
udptlfecentries = 3
udptlfecspan = 3


Good luck,

Ricardo.








Kokfoo Soo wrote:

Ricardo,
Thanks, could you please share some of your t.38 passthrough 
configuration in sip.conf and also udptl.conf?


Thanks,

*/Ricardo Carvalho [EMAIL PROTECTED]/* wrote:

No, T.38 doesn't work with Asterisk. Only works with Asterisk
t38passthrough patch that you can find at URL:
http://bugs.digium.com/file_download.php?file_id=9335type=bug
For me it only worked well with patch for version 1.2.4 of Asterisk.

Regards,

Ricardo.






Kokfoo Soo wrote:
 Is T.38 fax work through Asterisk? I have the config below in my
 sip.conf, but the fax doesn't work and give me the CLI lines
below. My
 current version is 1.2.10. Please help.

 [Inboundtopbx]
 type=friend
 context=pbx
 host=10.18.188.84
 insecure=port
 dtmfmode=rfc2833
 canreinvite=no
 disallow=all
 allow=g729
 allow=ulaw
 t38pt_udptl=yes
 t38pt_rtp=no
 t38pt_tcp=no

 [OutboundfromPBX]
 type=peer
 host=10.18.161.222
 canreinvite=no
 dtmfmode=rfc2833
 disallow=all
 allow=g729
 qualify=yes
 t38pt_udptl=yes
 t38pt_rtp=no
 t38pt_tcp=no

 -- SIP read from 10.18.188.84:50096:
 ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
 Via: SIP/2.0/UDP 10.18.188.84:5060
 From: ;tag=19D429E8-2084
 To: ;tag=as3c87a22e
 Date: Tue, 05 Sep 2006 19:42:28 GMT
 Call-ID: [EMAIL PROTECTED]
 Max-Forwards: 6
 Content-Length: 0
 CSeq: 101 ACK


 --- (9 headers 0 lines)---
 Sep 5 15:30:31 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP
 codec 100 received
 Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP
 codec 100 received
 Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP
 codec 100 received
 Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP
 codec 100 received
 Sep 5 15:30:34 WARNING[6839]: chan_sip.c:3475 process_sdp: Unknown
 SDP media type in offer: image 16406 udptl t38



 Yahoo! Messenger with Voice. Make PC-to-Phone Calls

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How low will we go? Check out Yahoo! Messenger’s low PC-to-Phone call 
rates. 
http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com 



 
http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com


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Re: [asterisk-users] How to notify an ACD agent before he/she picks up

2006-09-06 Thread MF

Anyone knows if I can get to know the operator ACD choosed to send the 
call by using  Realtime Queue,   or maybe via the manager?




Hi,

I need to send a message to an agent when the ACD starts to ring on 
he/she.
I have and application already built that sends such a message (just 
like a cti),  just don't know how to get from asterisk which agent was 
selected prior to ringing him   (or during ringing),  so that I can 
get information about the call and send it over.



any one done this?
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Re: [asterisk-users] Has anyone tried to install both digital card and analog card in one machine

2006-09-06 Thread MF

We are currently working with two TDM2400P,  with 32 FXS ports and one 
TE205P all on the same machine,   and it works fine,  (haven't done any 
stress testing yet though,  maybe if someone could share his/her 
experience with high load)


Xue Liangliang escribió:
Hi, all, I am not sure whether we can install both TDM400P and TE110P 
in the same machine, sometimes our customers have this kind 
requirement. And further more, is it possible to install both TE110P 
and TE410P in the  same machine?

Regards,
Liangliang
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Re: [asterisk-users] app_rxfax Only Receives One Page


Steve Totaro wrote:
I am trying to setup a fax server and all I get is the first page of a 
multipage fax.  The first page is perfect quality.


I am not sure how to debug this.  I have an HP DL320 with a quad 
Sangoma T1 board.


You'll be much happier moving it over to HylaFAX and iaxmodem.  Very 
easy to setup, low impact on the server and has error correction.


Doug

--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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[Asterisk-Users] Which SIP hardphone with embedded VPNClient ?

2006-09-06 Thread Olivier

Hi,Snom offers sip hardphones with embedded VPN client (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/162680/match=hardphone+vpn+client
).Beside that, I've read Zultys used to provide one (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/162680/match=hardphone+vpn+client
).Is anyone aware of others ?If positive, which kind of VPN does it support ?Cheers
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RE: [asterisk-users] Cisco PIX firewall and nat=yes

2006-09-06 Thread Bill Gibbs

Title: Message








As a follow up those commands helped with
the outbound calls but inbound still had issues. Asterisk would still show the
peer UNREACHABLE. Turning off qualify has fixed the problem!



Bill











From: Bill D'Anjou
[mailto:[EMAIL PROTECTED] 
Sent: Wednesday, August 23, 2006
12:47 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Cc: Bill Gibbs
Subject: RE: [asterisk-users]
Cisco PIX firewall and nat=yes







You might need:











fixup protocol sip 5060







fixup protocol sip udp 5060











in the PIX if these commands aren't
supported you might need newer code.











Bill







-Original Message-
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Bill Gibbs
Sent: Wednesday, August 23, 2006
8:53 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Cisco
PIX firewall and nat=yes

I have a Polycom 501 that works great from behind simple
firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the
register messages for the extensions on the Asterisk CLI but when I do a sip
show peers I see:



702/702
x.x.x.x D N 54297
UNREACHABLE

701/701
x.x.x.x D N
54297 UNREACHABLE

700/700
x.x.x.x D N
54297 UNREACHABLE



But I see stuff like

n Registered
SIP '702' at x.x.x.x port 54297 expires 60



I have a single phone with multiple extensions in the
example above. As a test I changed that phone to a single extension
(700), I see the Registered line but it still says UNREACHABLE.



I know the Asterisk config is good because every device
(soft, hard phone) works and I know the NAT works because Ive tested
that out.



SoIm thinking it has something to do with the
PIX. Any ideas? 



Bill








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Re: [asterisk-users] Budgetones - multiple phones losing IP address during day

2006-09-06 Thread Jessee J Holmes

Garth,This may be a silly question, but are you running the latest firmware on the phones from Grandstream? If not, try upgrading one phone to see if it helps solve the problem.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 6, 2006, at 3:21 AM, Garth van Sittert wrote:Hi AllI have a site with 50 Budgetone 102's and about 5 snom phones.At random intervals during the day about 20 or 30 of the Budgetones lose their connection to the network all at the same time.  It happens about once a day.  The Snom phones are fine and never get disconnected.  I can't ping the Budgetones IP's and the way to fix them is to simply unplug and reconnect.  Nothing interesting shows up in the logs even in debug mode except for the 'Peer XXX is now unreachable' repeated for each extension.I haven't had much experience with the Budgetones.  Does anyone have any idea what could be causing this?ThanksGarth___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [asterisk-users] SIP client with video???

2006-09-06 Thread Blake Krone

I've never experienced any of those problems. I can send video to other eyeBeam versions, 3.0 is the only version that supports video on the lite side. I've never lost any SIP information and only one registration isn't a big deal to me. If you need more than one buy the full eyebeam version.
On 9/6/06, Joao Pereira [EMAIL PROTECTED] wrote:
The problems with X-Lite 3 are:- just accepts one SIP registration- doesnt send video to other X-Lite or eyeBeam versions- sometimes loses the SIP informations when you reboot the PC.Regards
Joao PereiraBlake Krone wrote: What's wrong with X-Lite 3.0? I haven't had any issues with it and find it to be one of the best SIP video software choices, and it's free. On 7/27/06, *Joao Pereira *
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello to all can someone recommend me a nice SIP client with video for windows??
I tried X-Lite 3.0 but it's a lousy piece of software. Does someone knows about a better software? Regards Joao Pereira ___
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RE: [asterisk-users] Budgetones - multiple phones losing IP addressduring day

2006-09-06 Thread Harden, Bob









Its running now



tethereal -f host 12.20.121.2 and
icmp

Capturing on eth0













From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jessee J Holmes
Sent: Wednesday, September 06,
2006 11:47 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] Budgetones
- multiple phones losing IP addressduring day





Garth,









This may be a silly question, but are you running the latest firmware
on the phones from Grandstream? If not, try upgrading one phone to see if it
helps solve the problem.













Jessee
Holmes

Atacomm
/ Ataractic Corporation

www.atacomm.com

V:
1-877-700-VOIP

[EMAIL PROTECTED]



Looking
for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/















On Sep 6, 2006, at 3:21 AM, Garth van Sittert wrote:









Hi All











I have a site with 50 Budgetone 102's and about 5 snom phones.











At random intervals during the day about 20 or 30 of the Budgetones
lose their connection to the network all at the same time. It happens about once a day. The Snom phones are fine and never
get disconnected. I can't ping
the Budgetones IP's and the way to fix them is to simply unplug and reconnect. Nothing interesting shows up in the
logs even in debug mode except for the 'Peer XXX is now unreachable' repeated
for each extension.











I haven't had much experience with the Budgetones. Does anyone have any idea what could
be causing this?











Thanks





Garth











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Re: [asterisk-users] Cisco PIX firewall and nat=yes

2006-09-06 Thread Peder @ NetworkOblivion

There is a Timeout SIP in the config.  What is it set to?  If it is 
less than the the qualify interval, which I believe is 60 seconds, then 
the PIX will close the inbound hole for qualify traffic.  We've got lots 
of phones at several remote sites all running behind PIX's and all being 
NAT'd to the same IP (per location) and everything works perfect if 
qualify is on.  If we disable qualify, then the SIP inbound hole gets 
closed per the Timeout SIP and calls don't go through until the phone 
re-registers and the hole opens again (they can still call out).


Bill Gibbs wrote:
As a follow up those commands helped with the outbound calls but inbound 
still had issues.  Asterisk would still show the peer UNREACHABLE.  
Turning off qualify has fixed the problem!


 


Bill

 




*From:* Bill D'Anjou [mailto:[EMAIL PROTECTED]
*Sent:* Wednesday, August 23, 2006 12:47 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Cc:* Bill Gibbs
*Subject:* RE: [asterisk-users] Cisco PIX firewall and nat=yes

 


You might need:

 


fixup protocol sip 5060

fixup protocol sip udp 5060

 


in the PIX if these commands aren't supported you might need newer code.

 


Bill

-Original Message-
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Bill
Gibbs
*Sent:* Wednesday, August 23, 2006 8:53 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Cisco PIX firewall and nat=yes

I have a Polycom 501 that works great from behind simple firewalls,
like Dlink, etc however behind a Cisco PIX Firewall I see the
register messages for the extensions on the Asterisk CLI but when I
do a sip show peers I see:

 


702/702x.x.x.x D   N  54297UNREACHABLE

701/701x.x.x.x D   N  54297UNREACHABLE

700/700x.x.x.x D   N  54297UNREACHABLE

 


But I see stuff like

n   Registered SIP '702' at x.x.x.x port 54297 expires 60

 


I have a single phone with multiple extensions in the example above.
 As a test I changed that phone to a single extension (700), I see
the Registered line but it still says UNREACHABLE.

 


I know the Asterisk config is good because every device (soft, hard
phone) works and I know the NAT works because I’ve tested that out.

 


So…I’m thinking it has something to do with the PIX.  Any ideas?

 


Bill




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--

Network stuff you didn't know
http://www.networkoblivion.com

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RE: [asterisk-users] Cisco PIX firewall and nat=yes

2006-09-06 Thread Bill Gibbs

Thanks I will check into this.  I don't actually have access to the PIX
(I have to talk to like 3 people to get to the person who actually
manages this for the client) ...but that makes sense too

I currently have it registering at 60 secs

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peder @
NetworkOblivion
Sent: Wednesday, September 06, 2006 12:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco PIX firewall and nat=yes

There is a Timeout SIP in the config.  What is it set to?  If it is 
less than the the qualify interval, which I believe is 60 seconds, then 
the PIX will close the inbound hole for qualify traffic.  We've got lots

of phones at several remote sites all running behind PIX's and all being

NAT'd to the same IP (per location) and everything works perfect if 
qualify is on.  If we disable qualify, then the SIP inbound hole gets 
closed per the Timeout SIP and calls don't go through until the phone 
re-registers and the hole opens again (they can still call out).

Bill Gibbs wrote:
 As a follow up those commands helped with the outbound calls but
inbound 
 still had issues.  Asterisk would still show the peer UNREACHABLE.  
 Turning off qualify has fixed the problem!
 
  
 
 Bill
 
  
 


 
 *From:* Bill D'Anjou [mailto:[EMAIL PROTECTED]
 *Sent:* Wednesday, August 23, 2006 12:47 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Cc:* Bill Gibbs
 *Subject:* RE: [asterisk-users] Cisco PIX firewall and nat=yes
 
  
 
 You might need:
 
  
 
 fixup protocol sip 5060
 
 fixup protocol sip udp 5060
 
  
 
 in the PIX if these commands aren't supported you might need newer
code.
 
  
 
 Bill
 
 -Original Message-
 *From:* [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] *On Behalf Of
*Bill
 Gibbs
 *Sent:* Wednesday, August 23, 2006 8:53 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Cisco PIX firewall and nat=yes
 
 I have a Polycom 501 that works great from behind simple
firewalls,
 like Dlink, etc however behind a Cisco PIX Firewall I see the
 register messages for the extensions on the Asterisk CLI but when
I
 do a sip show peers I see:
 
  
 
 702/702x.x.x.x D   N  54297
UNREACHABLE
 
 701/701x.x.x.x D   N  54297
UNREACHABLE
 
 700/700x.x.x.x D   N  54297
UNREACHABLE
 
  
 
 But I see stuff like
 
 n   Registered SIP '702' at x.x.x.x port 54297 expires 60
 
  
 
 I have a single phone with multiple extensions in the example
above.
  As a test I changed that phone to a single extension (700), I see
 the Registered line but it still says UNREACHABLE.
 
  
 
 I know the Asterisk config is good because every device (soft,
hard
 phone) works and I know the NAT works because I've tested that
out.
 
  
 
 So...I'm thinking it has something to do with the PIX.  Any ideas?
 
  
 
 Bill
 
 


 
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Re: [asterisk-users] asterisk t.38 fax failed

2006-09-06 Thread Kokfoo Soo

Ricardo,I found compilation error below, any thought?chan_sip.c:3895: error: `UDPTL_ERROR_CORRECTION_REDUNDANCY' undeclared (first use in this function)chan_sip.c:3898: error: `UDPTL_ERROR_CORRECTION_FEC' undeclared (first use in this function)chan_sip.c:3901: error: `UDPTL_ERROR_CORRECTION_NONE' undeclared (first use in this function)chan_sip.c: In function `add_t38_sdp':chan_sip.c:4728: warning: implicit declaration of function `ast_udptl_get_us'chan_sip.c:4772: warning: implicit declaration of function `ast_udptl_get_local_max_datagram'chan_sip.c: In function `transmit_response_with_t38_sdp':chan_sip.c:5044: warning: implicit declaration of function `ast_udptl_offered_from_local'chan_sip.c: In function `handle_response':chan_sip.c:10516: warning: implicit declaration of function `ast_udptl_stop'chan_sip.c: In function `sip_set_udptl_peer':chan_sip.c:13488: warning: implicit declaration of function
 `ast_udptl_get_peer'chan_sip.c: At top level:chan_sip.c:13821: error: variable `sip_udptl' has initializer but incomplete typechan_sip.c:13822: error: unknown field `type' specified in initializerchan_sip.c:13822: warning: excess elements in struct initializerchan_sip.c:13822: warning: (near initialization for `sip_udptl')chan_sip.c:13823: error: unknown field `get_udptl_info' specified in initializerchan_sip.c:13823: warning: excess elements in struct initializerchan_sip.c:13823: warning: (near initialization for `sip_udptl')chan_sip.c:13824: error: unknown field `set_udptl_peer' specified in initializerchan_sip.c:13824: warning: excess elements in struct initializerchan_sip.c:13824: warning: (near initialization for `sip_udptl')chan_sip.c: In function `load_module':chan_sip.c:13986: warning: implicit declaration of function `ast_udptl_proto_register'chan_sip.c: In function `unload_module':chan_sip.c:14038:
 warning: implicit declaration of function `ast_udptl_proto_unregister'chan_sip.c: At top level:chan_sip.c:13821: error: storage size of `sip_udptl' isn't knownmake[1]: *** [chan_sip.o] Error 1make[1]: Leaving directory `/usr/src/asterisk-1.2.4/channels'make: *** [subdirs] Error 1Ricardo Carvalho [EMAIL PROTECTED] wrote: In sip.conf add to [general] context and to every peer context that you want to register in Asterisk to use T.38 the following lines:t38pt_udptl=yest38pt_rtp=not38pt_tcp=noIn udptl.conf file I have the following configurations:[general]udptlstart=4000udptlend=4999T38FaxUdpEC = t38UDPRedundancyT38FaxMaxDatagram = 400udptlfecentries = 3udptlfecspan = 3Good luck,Ricardo.Kokfoo Soo
 wrote: Ricardo, Thanks, could you please share some of your t.38 passthrough  configuration in sip.conf and also udptl.conf? Thanks, */Ricardo Carvalho <[EMAIL PROTECTED]>/* wrote: No, T.38 doesn't work with Asterisk. Only works with Asterisk t38passthrough patch that you can find at URL: http://bugs.digium.com/file_download.php?file_id=9335type=bug For me it only worked well with patch for version 1.2.4 of Asterisk. Regards, Ricardo. Kokfoo Soo wrote:  Is T.38 fax work through Asterisk? I have the config below in my  sip.conf, but the fax doesn't work and give me the CLI lines below. My  current version is 1.2.10. Please help.   [Inboundtopbx] 
 type=friend  context=pbx  host=10.18.188.84  insecure=port  dtmfmode=rfc2833  canreinvite=no  disallow=all  allow=g729  allow=ulaw  t38pt_udptl=yes  t38pt_rtp=no  t38pt_tcp=no   [OutboundfromPBX]  type=peer  host=10.18.161.222  canreinvite=no  dtmfmode=rfc2833  disallow=all  allow=g729  qualify=yes  t38pt_udptl=yes  t38pt_rtp=no  t38pt_tcp=no   -- SIP read from 10.18.188.84:50096:  ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0  Via: SIP/2.0/UDP 10.18.188.84:5060  From: ;tag=19D429E8-2084  To:
 ;tag=as3c87a22e  Date: Tue, 05 Sep 2006 19:42:28 GMT  Call-ID: [EMAIL PROTECTED]  Max-Forwards: 6  Content-Length: 0  CSeq: 101 ACK--- (9 headers 0 lines)---  Sep 5 15:30:31 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received  Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received  Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received  Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received  Sep 5 15:30:34 WARNING[6839]: chan_sip.c:3475 process_sdp: Unknown  SDP media type in offer: image 16406 udptl t38   
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[asterisk-users] Cisco MWI

2006-09-06 Thread Steve Kennedy

I have Asterisk 1.2.11 running and a Cisco 7960 (SIP v7.3). I cant seem
to get the message waiting indicator working.

I did try changing the MIME type as suggest, but then the phone kept
continuously ringing.

Any pointers?


Steve

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Re: [asterisk-users] Budgetones - multiple phones losing IP addressduring day

2006-09-06 Thread Jessee J Holmes

So the phones aren't loosing the IP address any longer?Just confirming what you meant by "It's running now". Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 6, 2006, at 11:02 AM, Harden, Bob wrote: Its running now tethereal -f "host 12.20.121.2 and icmp"Capturing on eth0  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, September 06, 2006 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Budgetones - multiple phones losing IP addressduring day  Garth,    This may be a silly question, but are you running the latest firmware on the phones from Grandstream? If not, try upgrading one phone to see if it helps solve the problem.      Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/   On Sep 6, 2006, at 3:21 AM, Garth van Sittert wrote:Hi All     I have a site with 50 Budgetone 102's and about 5 snom phones.     At random intervals during the day about 20 or 30 of the Budgetones lose their connection to the network all at the same time.  It happens about once a day.  The Snom phones are fine and never get disconnected.  I can't ping the Budgetones IP's and the way to fix them is to simply unplug and reconnect.  Nothing interesting shows up in the logs even in debug mode except for the 'Peer XXX is now unreachable' repeated for each extension.     I haven't had much experience with the Budgetones.  Does anyone have any idea what could be causing this?     Thanks  Garth     ___  --Bandwidth and Colocation provided by Easynews.com --     asterisk-users mailing list  To UNSUBSCRIBE or update options visit:    http://lists.digium.com/mailman/listinfo/asterisk-users     ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [asterisk-users] Submenus

Try breaking up the contexts. Contexts are what you call 'submenus'. 
For example:


[MetarMain]
exten = 1,1,answer
exten = 1,n,Background(Met_welcome)
exten = 1,n,Background(Met_Instructions)

exten = 3575,1,set(airport=ekrk) ; non-ambiguous
exten = 3575,n,goto(Metar_Process,s,1)

exten = 3524,1,goto(Metar_ekch_ekah,s,1) ; ambiguous, so submenu

[Metar_ekch_ekah]
exten = s,1,Background(Metar_ekch_or_ekah)
exten = s,n,WaitExten(3000)
exten = s,n,goto(1)

exten = 1,1,set(airport=ekch)
exten = 1,n,goto(Metar_Process,s,1)

exten = 2,1,set(airport=ekah)
exten = 2,n,goto(Metar_Process,s,1)

[Metar_Process]
exten = s,1,NoOP(Airport code chosen: ${airport})
exten = s,n,...

Made it up as it went along, so it could be strewn with bugs... but it 
should lead you in the right direction.


Moj

Mir wrote:

Hello
I'm doing an IVR-service, where pilot can check metar (airport weather
information), they enter the 4 letter airport code on their phone, and
get the metar read back by text-to-speech.


[Metar]

exten = 1,1,answer

exten = 1,2,Background(Met_welcome)

exten = 1,3,set(airport=)2,Background(Met_welcome)

exten = 1,3,set(airport=)3,set(airport=)

exten = 1,n,Background(Met_Instructions)

(When they press an airportcode, I set a variable)

exten = 3575,1,set(airport=ekrk)

exten = 3575,n,goto(Metar,s,1)

exten = 3598,1,set(airport=ekyt)

and so on

exten = 3575,n,goto(Metar,s,1)

exten = 3598,1,set(airport=ekyt)

and so on

exten = 3598,1,set(airport=ekyt)

and so on
In the S extension, I do all of the processing .
My problem is that some of the airports has the same code, for
instance EKCH wich is entered by pressing 3524, but EKAH has the same
digits, so I need to make a sub-menu, where I can ask the caller to
press 1 for EKCH or 2 for EKAH.
Right now, I do like this:


exten = 3524,1,background(Met_ch_bi_ah)background(Met_ch_bi_ah)

exten = 4,1,set(airport=ekch)

exten = 4,n,goto(Metar,s,1)

exten = 6,1,set(airport=ekbi)

exten = 6,n,goto(Metar,s,1)1,set(airport=ekch)

exten = 4,n,goto(Metar,s,1)

exten = 6,1,set(airport=ekbi)

exten = 6,n,goto(Metar,s,1)6,1,set(airport=ekbi)

exten = 6,n,goto(Metar,s,1)6,n,goto(Metar,s,1)

This is not a very good solution, if a user by mistake press 4 in the
main loop, it goes right to EKCH metar, instead of informing the user
that it is an invalid code.

So what I need is a way to make a submenu, that is only visible when
needed, and can set the airport variable.

How do I do this?



Michael
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!DSPAM:500,44fc4a7d289876491211187!



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(907) 747- x112
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Re: [asterisk-users] app_rxfax Only Receives One Page


Doug Lytle wrote:

Steve Totaro wrote:
I am trying to setup a fax server and all I get is the first page of 
a multipage fax.  The first page is perfect quality.


I am not sure how to debug this.  I have an HP DL320 with a quad 
Sangoma T1 board.


You'll be much happier moving it over to HylaFAX and iaxmodem.  Very 
easy to setup, low impact on the server and has error correction.


Doug

I used the asterfax install script which installs iaxmodem.  I will look 
into HylaFAX but I would love to find the solution to the one page 
problem while I pursue HylaFAX.

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Re: [asterisk-users] app_rxfax Only Receives One Page

Try to increase your rxgain, and check you have echocancel disabledpls post your resultsOn 9/6/06, Steve Totaro 
[EMAIL PROTECTED] wrote:Doug Lytle wrote: Steve Totaro wrote:
 I am trying to setup a fax server and all I get is the first page of a multipage fax.The first page is perfect quality. I am not sure how to debug this.I have an HP DL320 with a quad
 Sangoma T1 board. You'll be much happier moving it over to HylaFAX and iaxmodem.Very easy to setup, low impact on the server and has error correction. Doug
I used the asterfax install script which installs iaxmodem.I will lookinto HylaFAX but I would love to find the solution to the one pageproblem while I pursue HylaFAX.___
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Re: [asterisk-users] app_rxfax Only Receives One Page


Steve Totaro wrote:
I used the asterfax install script which installs iaxmodem.  I will 
look into HylaFAX but I would love to find the solution to the one 
page problem while I pursue HylaFAX.


Any errors being displayed on the Console?  Have you run in in debug 
mode while a fax in coming in?  Can you forward me one of the TIFs off list?


Doug


--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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Re: [asterisk-users] Cisco MWI


Steve Kennedy wrote:

I have Asterisk 1.2.11 running and a Cisco 7960 (SIP v7.3). I cant seem
to get the message waiting indicator working.

I did try changing the MIME type as suggest, but then the phone kept
continuously ringing.

Any pointers?


Steve

  

Lets see your sip.conf entry for the phones setup.

Doug

--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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[asterisk-users] Conditional IF based on IP address?

2006-09-06 Thread Steve Hsieh

Greetings,

Is it possible to create a conditional IF inside extensions.conf based on the source IP address of a SIP phone (as opposed to extension)? What I would like to do is the following:


1. IfSIP phone IP belongs to 192.168.0.0/24 subnet, set CALLERID=
2. Else, set CALLERID=

Thanks in advance for any examples or help.

Steve
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Re: [asterisk-users] app_rxfax Only Receives One Page


Doug Lytle wrote:

Steve Totaro wrote:
I used the asterfax install script which installs iaxmodem.  I will 
look into HylaFAX but I would love to find the solution to the one 
page problem while I pursue HylaFAX.


Any errors being displayed on the Console?  Have you run in in debug 
mode while a fax in coming in?  Can you forward me one of the TIFs off 
list?


Doug


Turns out it is just from a particular fax on demand service I found 
through Google, I have tried two other real faxes and they come across 
just fine (although they are in landscape and not portrait)  Still not 
sure if they were only sending one page but I didn't get any errors.


I was using http://www.vchannel.com/fodindex.htm

Thanks for the suggestions, I am looking into HylaFAX for a more 
permanent solution.

Steve
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Re: [asterisk-users] app_rxfax Only Receives One Page

2006-09-06 Thread Steve Underwood


Marco Mouta wrote:


Try to increase your rxgain, and check you have echocancel disabled


Better still, try leaving the gains alone. If the gain controls were 
removed completely from Asterisk, support issues would decrease 
dramatically.


Steve


pls post your results

On 9/6/06, *Steve Totaro*  [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Doug Lytle wrote:
 Steve Totaro wrote:
 I am trying to setup a fax server and all I get is the first
page of
 a multipage fax.  The first page is perfect quality.

 I am not sure how to debug this.  I have an HP DL320 with a quad
 Sangoma T1 board.

 You'll be much happier moving it over to HylaFAX and iaxmodem.  Very
 easy to setup, low impact on the server and has error correction.

 Doug

I used the asterfax install script which installs iaxmodem.  I
will look
into HylaFAX but I would love to find the solution to the one page
problem while I pursue HylaFAX.



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[asterisk-users] Is asterisk's mgcp support(NAS) Network access server package

2006-09-06 Thread Ibrar Ahmed

hi

Is asterisk's mgcp support(NAS) Network access server package


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Re: [asterisk-users] Is asterisk's mgcp support(NAS) Network access server package

2006-09-06 Thread Davor Grgicevic

i am out of office until  11.09.2006.
If urgent please contact me on my mobile phone 
00385 91 1988 815

davor
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Re: [asterisk-users] Asterisk 1.2.11 and # key

2006-09-06 Thread Matt


Exactly... why WOULD you use two?   We use 1 # for blind transfer.
If I need to enter # to an external call I enter it twice.   (## will
send a single # to the called party).

On 9/6/06, Michael Strelnikov [EMAIL PROTECTED] wrote:

So, I don't understand the reason to use ## then. I think you are wrong.


On 9/6/06, David Gagnon [EMAIL PROTECTED]  wrote:





 No, when you press the first time, Asterisk is in standby waiting for the
other one.



 David



 


 De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
de Michael Strelnikov
 Envoyé : 5 septembre 2006 02:37

 À : Asterisk Users Mailing List - Non-Commercial Discussion
 Objet : Re: [asterisk-users] Asterisk 1.2.11 and # key







 But the behaviour should like: if pressed once the # should be
transmitted. If pressed ## (fast) the it should be blind transfer. Isn't it?


 On 9/5/06, David Gagnon  [EMAIL PROTECTED] wrote:




 That why, when you dial one # then Asterisk wait to see if you dial two of
them. You should consider changing the blindxfer function or play with the
timer in the features.conf. In think its look like featuresdigittimeut.



 For the moment, if you dial #( wait 1 sec) then press it again, the second
will work.



 David

 


 De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
de Michael Strelnikov
 Envoyé : 4 septembre 2006 22:47
 À : Asterisk Users Mailing List - Non-Commercial Discussion
 Objet : Re: [asterisk-users] Asterisk 1.2.11 and # key




 I have blindxfer = ## line in my features.conf


 On 9/5/06, David Gagnon [EMAIL PROTECTED] wrote:




 Are you sure this is not because of the dynamic features in features.conf
?

 By default, # is defined for the transfer feature.



 David



 


 De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
de Michael Strelnikov
 Envoyé : 4 septembre 2006 09:53
 À : asterisk-users@lists.digium.com
 Objet : [asterisk-users] Asterisk 1.2.11 and # key




 Hello,

Does anybody have problems with recognition of the hash (#) key with *
1.2.11? It seams that after pressing # the call is in a progress but no data
is sent.

 Thanks in advance,
 Michael


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Re: [asterisk-users] Submenus


In fact, change the n in
exten = 1,n,Background(Met_Instructions) to 3 like so:
exten = 1,3,Background(Met_Instructions)
and then add immediately after that line:
exten = 1,4,WaitExten(3000)
exten = 1,5,Goto(3)

So if they don't answer immediately they are read the instructions again.

Mojo with Horan  Company, LLC wrote:
Try breaking up the contexts. Contexts are what you call 'submenus'. 
For example:


[MetarMain]
exten = 1,1,answer
exten = 1,n,Background(Met_welcome)
exten = 1,n,Background(Met_Instructions)

exten = 3575,1,set(airport=ekrk) ; non-ambiguous
exten = 3575,n,goto(Metar_Process,s,1)

exten = 3524,1,goto(Metar_ekch_ekah,s,1) ; ambiguous, so submenu

[Metar_ekch_ekah]
exten = s,1,Background(Metar_ekch_or_ekah)
exten = s,n,WaitExten(3000)
exten = s,n,goto(1)

exten = 1,1,set(airport=ekch)
exten = 1,n,goto(Metar_Process,s,1)

exten = 2,1,set(airport=ekah)
exten = 2,n,goto(Metar_Process,s,1)

[Metar_Process]
exten = s,1,NoOP(Airport code chosen: ${airport})
exten = s,n,...

Made it up as it went along, so it could be strewn with bugs... but it 
should lead you in the right direction.


Moj

Mir wrote:

Hello
I'm doing an IVR-service, where pilot can check metar (airport weather
information), they enter the 4 letter airport code on their phone, and
get the metar read back by text-to-speech.


[Metar]

exten = 1,1,answer

exten = 1,2,Background(Met_welcome)

exten = 1,3,set(airport=)2,Background(Met_welcome)

exten = 1,3,set(airport=)3,set(airport=)

exten = 1,n,Background(Met_Instructions)

(When they press an airportcode, I set a variable)

exten = 3575,1,set(airport=ekrk)

exten = 3575,n,goto(Metar,s,1)

exten = 3598,1,set(airport=ekyt)

and so on

exten = 3575,n,goto(Metar,s,1)

exten = 3598,1,set(airport=ekyt)

and so on

exten = 3598,1,set(airport=ekyt)

and so on
In the S extension, I do all of the processing .
My problem is that some of the airports has the same code, for
instance EKCH wich is entered by pressing 3524, but EKAH has the same
digits, so I need to make a sub-menu, where I can ask the caller to
press 1 for EKCH or 2 for EKAH.
Right now, I do like this:


exten = 3524,1,background(Met_ch_bi_ah)background(Met_ch_bi_ah)

exten = 4,1,set(airport=ekch)

exten = 4,n,goto(Metar,s,1)

exten = 6,1,set(airport=ekbi)

exten = 6,n,goto(Metar,s,1)1,set(airport=ekch)

exten = 4,n,goto(Metar,s,1)

exten = 6,1,set(airport=ekbi)

exten = 6,n,goto(Metar,s,1)6,1,set(airport=ekbi)

exten = 6,n,goto(Metar,s,1)6,n,goto(Metar,s,1)

This is not a very good solution, if a user by mistake press 4 in the
main loop, it goes right to EKCH metar, instead of informing the user
that it is an invalid code.

So what I need is a way to make a submenu, that is only visible when
needed, and can set the airport variable.

How do I do this?



Michael
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Re: [asterisk-users] Dell Poweredge SC430 and Digium cards compatability enquiry

2006-09-06 Thread Matt Birmingham

I'm using a 400SC with two TDM400P cards and a TrixBox install. This is an older version of the SC430 and it's been working like a champ. I also have an SC430, but I'm not using it for Asterisk, so I can't give you insight on that one. Sorry.
On 9/5/06, Matthew Thompson [EMAIL PROTECTED] wrote:
We're looking at using a number of Dell Poweredge SC430 servers as Asterisk hosts in our smaller overseas offices with Digium cards in to provide local breakout over the pre-existing analogue or digital phone lines (One office uses ISDN2 the others analogue)
I note that the SC420 is listed as incompatible but the SC430 appears to be a slightly different beast in terms of chipset, the 430 has the newer E7230 as opposed to the E7221 - does this make a difference to compatibility?
 
--Matthew Thompson[EMAIL PROTECTED]
http://www.voipnews.org.uk/ 
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Re: [asterisk-users] Native Chinese speaker needed

2006-09-06 Thread John Williams

 Chinese uses a prefix for ordinal numebrs. However ordinal number are
 not used in dates. In Chinese today (its the 6th September here) is
 expressed as
 zero six year nine month six day

It is currently programmed to say two thousand six year for the year
part.

 You can skip the zero six year part, and just say the month and day.
 You can also use the word for number (not the ordinal prefix) in place
 of the word for day, if you like. A complication is that the second of
 the month uses a word that we might translate as a couple (sounds odd
 to say a couple as a day, but words don't quite trannslate cleanly). So
 the second of September was
zero six year nine month couple day
 but the twelfth will be
zero six year nine month ten two day

Asterisk currently does:

 two thousand six year nine month tenth second
 two thousand six year nine month second

My patch, as it is now, would do:

 two thousand six year nine month ten two day
 two thousand six year nine month two day

Is couple used instead of two anywhere else?  You use it for day and
minute.  Is it ever used for year, month, hour, or second?

How bad is it to say two instead of couple?  I could probably program
it to play couple if the recording exists, and fall back to two if
that is at all acceptable.

Are there any other numbers which might be expressed differently in some
circumstances?

 I hope that's not too confusing :-\ Do you want to know about times too?
 To express those in a general colloquial way, ten past two would be
 two time two
 Because the minutes part is expressed like the face of the clock, 10
 minutes is expressed as 2. You can say
 two time ten minute
 but it sounds a bit stuffy. Notice I didn't use the word for couple in
 that. If I wanted to say 2 minutes after two, I would say
 two time couple minute

Is a 12-hour or 24-hour clock preferred in chinese?

Asterisk currently uses a 24-hour clock by default (format HM):

  ten four time ten minute
  ten four time zero two minute
  zero two time zero zero minute  (02:00am)

and can add zero seven second to that if seconds are requested (HMS).
Is the zero zero minute very bad?  And how stuffy is zero two minute?

A 12-hour time format (IMP) would be expressed:

  two time ten minute p-m
  two time zero two minute p-m
  two time zero zero minute a-m

I am told that a-m and p-m are very similar sounds in chinese.

 Here I did use the couple word, and I expressed a time with greater
 precision than the clock face numbers permit.

 Those expressions might sound odd as English, because none of the words
 have a plural ending. That is how Chinese is expressed - no plurals.

I like it!

~ John Williams


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[asterisk-users] Call parking and RTP traffic

2006-09-06 Thread Dave Fullerton



Greetings

I've noticed something odd while messing around with a test system and 
I'm not sure if this is a bug or not. I have three phones connected to 
an asterisk system in a remote office over a point to point T1 (no nat) 
all set up with canreinvite=yes. The phones are a Polycom 601, 501 and a 
budgetone 102. Here's what I am seeing:


Phone A calls phone B, phone B answers. RTP traffic travels between A 
and B directly like it is supposed to. Phone B does an attended transfer 
(using the phone's transfer features) to the parking extension, waits 
for the parked number and then completes the transfer. RTP travels from 
phone A to asterisk and hold music is played like it is supposed to. 
Now, phone B calls the parked extension and the call is reconnected, 
except the RTP traffic is now traveling A - Asterisk - B. I would 
have thought traffic should resume going between A and B directly. Is 
this an incorrect assumption or is it a bug?


I've tested this on 1.2.11 and SVN-branch-1.2-r41989.

Thanks

-Dave
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Re: [asterisk-users] Conditional IF based on IP address?

2006-09-06 Thread Rushowr

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Steve Hsieh wrote:
 Greetings,
  
 Is it possible to create a conditional IF inside extensions.conf based
 on the source IP address of a SIP phone (as opposed to extension)?  What
 I would like to do is the following:
  
  
 1. If SIP phone IP belongs to 192.168.0.0/24 http://192.168.0.0/24
 subnet, set CALLERID=
 2. Else, set CALLERID=
  
 Thanks in advance for any examples or help.
  
 Steve
 
 
 
 
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Steve, yes you can do it.

You'll need to use the SIPPEER function (available in trunk only I
believe) to get ${SIPPEER(ipaddr)}. You can then use Set and If() to do
what you want.

SKM
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Re: [asterisk-users] Really bad phone line.. possible causes?

What codec are your sip phones using?  We'd have a similar, though 
immediate, degradation in audio quality using G.729 when zaptel was 
built with MMX optimizations.  We use an AMD CPU.


When zaptel was rebuilt without MMX optimizations we were back in business.

Jeff Turner wrote:

Hi,

I was wondering if those more familiar with PSTN-Asterisk installs
could take a listen to this (86k recording):

http://confluence.atlassian.com/download/attachments/185668/linegoingbad.mp3?version=1

It's what I hear dialling into our Asterisk box. As soon as the call
receiver makes a sound (clicking fingers in that recording), the line
makes a brief buzzing noise, then goes crazy with raw static.

There is a longer recording and configuration info at:

http://confluence.atlassian.com/display/TEST/Asterisk+phoneline+samples

The odd thing is, we have two phonelines, and the connection quality of
the second line coming into Asterisk is perfect.

Our Asterisk box is hooked up (in parallel with a NEC PBX) to our public
phone line via a TDM404P card.

Does this ring any bells for anyone?


Thanks,
Jeff
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!DSPAM:500,44fe5161224541822916521!



--
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Office Manager, Horan  Company, LLC
(907) 747- x112
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[asterisk-users] Call parking and ringbacks

2006-09-06 Thread J. Oquendo


Greets all and TIA

Here is my description short and simple:

Call comes in -- Gets parked -- parking time expires -- rings back 
person who parked the call


Is there a way for me to change the extension when ringing back?

Normally Asterisk does this:

12125551212 -- AsteriskPBX -- extension 1202 (parks call) -- time 
expires -- extension701 (parking extension) -- ring back 1202


Is there either a) a method for me to change the tone via a new 
extension, or change the callerid name on THAT ringback from the 701-720 
range?


--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 


The happiness of society is the end of government.
John Adams

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Re: [asterisk-users] Cisco MWI


Steve Kennedy wrote:

Phone itself.

[S-5200]
  

This is incorrect.  It should be:

[5200]


mailbox=5200
  


You're missing the @context on your mailbox line.  i.e. my phones are in 
the from-sip context, so:


[EMAIL PROTECTED]



Doug

--

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deserve neither Liberty nor Safety.


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Re: [asterisk-users] Conditional IF based on IP address?

2006-09-06 Thread Rushowr

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Rushowr wrote:
 Steve Hsieh wrote:
 Greetings,
 
 Is it possible to create a conditional IF inside extensions.conf based
 on the source IP address of a SIP phone (as opposed to extension)?  What
 I would like to do is the following:
 
 
 1. If SIP phone IP belongs to 192.168.0.0/24 http://192.168.0.0/24
 subnet, set CALLERID=
 2. Else, set CALLERID=
 
 Thanks in advance for any examples or help.
 
 Steve
 
 
 
 
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 Steve, yes you can do it.
 
 You'll need to use the SIPPEER function (available in trunk only I
 believe) to get ${SIPPEER(ipaddr)}. You can then use Set and If() to do
 what you want.
 
 SKM

I'm sorry, you need ${SIPPEER(${EXTEN}|ipaddr)} to retrieve the data you
want
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[asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

2006-09-06 Thread Dan Serban

I have a system running Asterisk 1.2.10 (Debian packaging) and about 50
Linksys SPA-942 phones, after the initial config and mass deployment of
the phones everything looks like it's configured well.

When an incoming call is answered and then attempted to be xfer'ed via
the soft button on the phone itself, it seems that if you hit the button
twice in quick succession, there is no problem (effectively a blind
transfer), if then I try to tell the other extension that Joe is
calling to sell you a fridge and hit xfer, the calling party cannot
hear what that person at the extension is saying.  Sometimes the tables
are fully turned, the caller can hear, but the operator can't hear a thing.

One thing's for sure, if you hit the button quickly (blind transfer) it
works no problem at all.

This is what I see asterisk saying when I transfer the call unsuccessfully.

== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/82-006d42a0ZOMBIE' in macro 'stdexten'
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/82-006d42a0ZOMBIE'

I've looked at the macro with a fine tooth comb, I cannot see any
problems with it whatsoever, (though that doesn't mean that my ignorance
isn't getting in the way).

I found some mention on the digium mantis bug tracker, here's the link:

http://bugs.digium.com/view.php?id=7421

Before I try and patch the source (which I'm hesitant to do since I run
the debian packages), is there another solution or maybe an unidentified
issue that I haven't been able to decipher?

If there's more information that I can provide to solve this problem,
I'd be happy to do so.

Thank you.
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Re: [asterisk-users] Native Chinese speaker needed

2006-09-06 Thread Steve Hsieh

John,

My patch, as it is now, would do:two thousand six year nine month ten two daytwo thousand six year nine month two day
Is couple used instead of two anywhere else?You use it for day andminute.Is it ever used for year, month, hour, or second?

For year, it should be couple in front of the thousand. In addition, you need to explicity say zero as well:

2006 = couple thousand zero six year

However, it gets interesting at 2010. I surveyed 5 native Chinese speaking colleagues (from both Taiwan and China) in addition to myself, and there isn't any agreement on what to say to make it sound right. The only agreement I found was to say the digits individually. Everyone felt comfortable with having a voicemail system speak the year as:

2006 = two zero zero six year
2010 = two zero one zero year

It's also simpler to implement.

For month day, youcan use two:

Feb 2 = two month 2 day

But it goes back to couple when speaking 2 o'clock:

2:22pm = coupletime twenty two PM

How bad is it to say two instead of couple?I could probably programit to play couple if the recording exists, and fall back to two if
that is at all acceptable.

It would sound very awkward. People will understand if two is said instead, but it sticks out badly.

Are there any other numbers which might be expressed differently in somecircumstances?

I think 2 is the only digit that changes.

Is a 12-hour or 24-hour clock preferred in chinese?

Civiliansgenerallyspeak in12-hr format.

Asterisk currently uses a 24-hour clock by default (format HM): ten four time ten minute ten four time zero two minute
zero two time zero zero minute(02:00am)and can add zero seven second to that if seconds are requested (HMS).Is the zero zero minute very bad?

If the minutes are zero, you'd drop it and just say the hour (same as English).
02:00am = coupletime AM

And how stuffy is zero two minute?

This is correct.

A 12-hour time format (IMP) would be expressed: two time ten minute p-m two time zero two minute p-m
two time zero zero minute a-m

You'd want to say couple time instead of two time in front.

Steve

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Re: [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

2006-09-06 Thread Alberto Sagredo

I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works 
fine. Are you canreinvite=yes ?.


I have not been notice any problem related to transferring calls (blind 
and attended)


Regards

Dan Serban escribió:

I have a system running Asterisk 1.2.10 (Debian packaging) and about 50
Linksys SPA-942 phones, after the initial config and mass deployment of
the phones everything looks like it's configured well.

When an incoming call is answered and then attempted to be xfer'ed via
the soft button on the phone itself, it seems that if you hit the button
twice in quick succession, there is no problem (effectively a blind
transfer), if then I try to tell the other extension that Joe is
calling to sell you a fridge and hit xfer, the calling party cannot
hear what that person at the extension is saying.  Sometimes the tables
are fully turned, the caller can hear, but the operator can't hear a thing.

One thing's for sure, if you hit the button quickly (blind transfer) it
works no problem at all.

This is what I see asterisk saying when I transfer the call unsuccessfully.

== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/82-006d42a0ZOMBIE' in macro 'stdexten'
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/82-006d42a0ZOMBIE'

I've looked at the macro with a fine tooth comb, I cannot see any
problems with it whatsoever, (though that doesn't mean that my ignorance
isn't getting in the way).

I found some mention on the digium mantis bug tracker, here's the link:

http://bugs.digium.com/view.php?id=7421

Before I try and patch the source (which I'm hesitant to do since I run
the debian packages), is there another solution or maybe an unidentified
issue that I haven't been able to decipher?

If there's more information that I can provide to solve this problem,
I'd be happy to do so.

Thank you.
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Re: [asterisk-users] Conditional IF based on IP address?

2006-09-06 Thread Steve Hsieh

Thanks, Russ!

Any suggestions on how to apply a subnet mask so that I can match an IP that belongs to 192.168.0.0/23, for example? Or would the only way be to matchthe string using REGEX?


On 9/6/06, Rushowr [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-Hash: SHA1Rushowr wrote: Steve Hsieh wrote: Greetings,
 Is it possible to create a conditional IF inside extensions.conf based on the source IP address of a SIP phone (as opposed to extension)?What I would like to do is the following:
 1. If SIP phone IP belongs to 192.168.0.0/24 http://192.168.0.0/24 subnet, set CALLERID= 2. Else, set CALLERID=
 Thanks in advance for any examples or help. Steve  ___
 --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users Steve, yes you can do it. You'll need to use the SIPPEER function (available in trunk only I believe) to get ${SIPPEER(ipaddr)}. You can then use Set and If() to do
 what you want. SKMI'm sorry, you need ${SIPPEER(${EXTEN}|ipaddr)} to retrieve the data youwant
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Re: [asterisk-users] Conditional IF based on IP address?

2006-09-06 Thread Rushowr

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Steve Hsieh wrote:
 Thanks, Russ!
  
 Any suggestions on how to apply a subnet mask so that I can match an IP
 that belongs to 192.168.0.0/23 http://192.168.0.0/23, for example? Or
 would the only way be to match the string using REGEX?
 
  
 On 9/6/06, *Rushowr* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:
 
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 Rushowr wrote:
  Steve Hsieh wrote:
  Greetings,
 
  Is it possible to create a conditional IF inside extensions.conf
 based
  on the source IP address of a SIP phone (as opposed to
 extension)?  What
  I would like to do is the following:
 
 
  1. If SIP phone IP belongs to 192.168.0.0/24
 http://192.168.0.0/24 http://192.168.0.0/24
  subnet, set CALLERID=
  2. Else, set CALLERID=
 
  Thanks in advance for any examples or help.
 
  Steve
 
 
 
 
 
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  Steve, yes you can do it.
 
  You'll need to use the SIPPEER function (available in trunk only I
  believe) to get ${SIPPEER(ipaddr)}. You can then use Set and If()
 to do
  what you want.
 
  SKM
 
 I'm sorry, you need ${SIPPEER(${EXTEN}|ipaddr)} to retrieve the data you
 want
 
 
 
 
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Your best bet would be to use the REGEX function to match the first
three octets :)

Rushowr
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=FT10
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Re: [asterisk-users] Different MOH between waiting calls and transfer

2006-09-06 Thread equis software

I know SetMusicOnHold command, but when a call is in the queue waiting, the queue function play the MOH configured 'myMOH', see the example below:exten = 90,1,Answerexten = 90,2,Ringingexten = 90,3,SetMusicOnHold(myMOH)
exten = 90,4,Queue(myQueue|tw|||300)When the agent who answer the call transfer it in Attended mode I can´t play a different music on hold to the caller.I try this, but obviously doesn't work.exten = _XXX,1,SetMusicOnHold(mySecondMOH)
exten = _XXX,2,Dial(Zap/g1/${EXTEN:1},20,t)ThanksEstebanOn 9/5/06, Doug Lytle [EMAIL PROTECTED]
 wrote:equis software wrote: Could I use different music on hold between waiting calls in queue and
 calls that are waiting to be tranfered?Yes, with the SetMusicOnHold command.http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetMusicOnHold
Doug--Ben Franklin quote:Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety.___
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Re: [asterisk-users] Different MOH between waiting calls and transfer


equis software wrote:
I know SetMusicOnHold command, but when a call is in the queue 
waiting, the queue function play the MOH configured 'myMOH', see the 
example below:


I don't think you can with the dialplan.  Maybe with the manager 
interface.  You'd have to set it before the call is answered.


Doug

--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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Re: [asterisk-users] app_rxfax Only Receives One Page

It has happened to me, with a x100p card, problems receiving fax that i've solved adjusting the gains.

Don't understand quite well why you say that...


On 9/6/06, Steve Underwood [EMAIL PROTECTED] wrote:
Marco Mouta wrote: Try to increase your rxgain, and check you have echocancel disabled
Better still, try leaving the gains alone. If the gain controls wereremoved completely from Asterisk, support issues would decreasedramatically.Steve pls post your results On 9/6/06, *Steve Totaro*  
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Doug Lytle wrote:
  Steve Totaro wrote:  I am trying to setup a fax server and all I get is the first page of  a multipage fax.The first page is perfect quality. 
  I am not sure how to debug this.I have an HP DL320 with a quad  Sangoma T1 board.   You'll be much happier moving it over to HylaFAX and iaxmodem.Very
  easy to setup, low impact on the server and has error correction.   Doug  I used the asterfax install script which installs iaxmodem.I will look
 into HylaFAX but I would love to find the solution to the one page problem while I pursue HylaFAX.___--Bandwidth and Colocation provided by 
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-- Com os melhores cumprimentos,Marco Mouta 
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[asterisk-users] Volume events causing talk off on Asterisk with Digium 411P

2006-09-06 Thread Servetas, Andrew











We are experiencing random talk off events when we hear a
loud volume event on the PSTN side of our calls. We do not always hear
the spurious DTMF, but I can see it in the console when I have the debug and
verbose levels turned up. We do however always have the associated brief
periods of silence that immediately follow. Sometimes they are only a
matter of seconds, other times they can be as long as a minute. We hear
it most often if the remote party is on a cellular phone with a lot of
background noise, or if a loud noise happens during the call. Neither
party can hear the other when this happens. It almost reacts like an AGC
circuit is muting the call.



We are using a Digium TE411P quad-span T1 card on
1.2.5. I called Digium support and we have played with the VPM_DEFAULT_DTMFTHRESHOLD
in the WCT4XXP.C driver as recommended, and all the RELAXDTMF and GAIN
settings in Zapata.conf are set according to their recommendations.



Has anyone else experienced this, and if so, what have you
done to correct it?



Andy
Servetas

CTI Support Engineer



Dirigosoft
Corporation

Portland, ME



www.dirigosoft.com









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Re: [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

2006-09-06 Thread Dan Serban

Alberto Sagredo wrote:
 I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works
 fine. Are you canreinvite=yes ?.
 
 I have not been notice any problem related to transferring calls (blind
 and attended)
 

Thank you for your response, it gave me a nudge to check the
configuration in the sip.conf file.  It seems that if I set
canreinvite=no for every SIP peer, it works!

And I have found no other adverse effects.  Strange issue...

 Regards
 
 Dan Serban escribió:
 I have a system running Asterisk 1.2.10 (Debian packaging) and about 50
 Linksys SPA-942 phones, after the initial config and mass deployment of
 the phones everything looks like it's configured well.

 When an incoming call is answered and then attempted to be xfer'ed via
 the soft button on the phone itself, it seems that if you hit the button
 twice in quick succession, there is no problem (effectively a blind
 transfer), if then I try to tell the other extension that Joe is
 calling to sell you a fridge and hit xfer, the calling party cannot
 hear what that person at the extension is saying.  Sometimes the tables
 are fully turned, the caller can hear, but the operator can't hear a
 thing.

 One thing's for sure, if you hit the button quickly (blind transfer) it
 works no problem at all.

 This is what I see asterisk saying when I transfer the call
 unsuccessfully.

 == Spawn extension (macro-stdexten, s, 1) exited non-zero on
 'SIP/82-006d42a0ZOMBIE' in macro 'stdexten'
 == Spawn extension (macro-stdexten, s, 1) exited non-zero on
 'SIP/82-006d42a0ZOMBIE'

 I've looked at the macro with a fine tooth comb, I cannot see any
 problems with it whatsoever, (though that doesn't mean that my ignorance
 isn't getting in the way).

 I found some mention on the digium mantis bug tracker, here's the link:

 http://bugs.digium.com/view.php?id=7421

 Before I try and patch the source (which I'm hesitant to do since I run
 the debian packages), is there another solution or maybe an unidentified
 issue that I haven't been able to decipher?

 If there's more information that I can provide to solve this problem,
 I'd be happy to do so.

 Thank you.
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Re: [asterisk-users] Cisco MWI

2006-09-06 Thread Steve Kennedy

On Wed, Sep 06, 2006 at 03:36:53PM -0400, Doug Lytle wrote:

 Steve Kennedy wrote:
 Phone itself.
 [S-5200]
 This is incorrect.  It should be:
 [5200]
 mailbox=5200

That bit seems to work, phones registers ok and can receive and make
calls.

 You're missing the @context on your mailbox line.  i.e. my phones are in 
 the from-sip context, so:
 [EMAIL PROTECTED]

But my mailbox (5200) is in default.


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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Re: [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

2006-09-06 Thread Alberto Sagredo

I use canreinvite=yes in my config files, and it does work, so maybe its 
a spa 941 misconfiguration.


I think if nat=no sometime it has problems if you are behind NAT, but 
under same network it must not fail.


Which firmware are you running on spas?

Dan Serban escribió:

Alberto Sagredo wrote:
  

I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works
fine. Are you canreinvite=yes ?.

I have not been notice any problem related to transferring calls (blind
and attended)




Thank you for your response, it gave me a nudge to check the
configuration in the sip.conf file.  It seems that if I set
canreinvite=no for every SIP peer, it works!

And I have found no other adverse effects.  Strange issue...

  

Regards

Dan Serban escribió:


I have a system running Asterisk 1.2.10 (Debian packaging) and about 50
Linksys SPA-942 phones, after the initial config and mass deployment of
the phones everything looks like it's configured well.

When an incoming call is answered and then attempted to be xfer'ed via
the soft button on the phone itself, it seems that if you hit the button
twice in quick succession, there is no problem (effectively a blind
transfer), if then I try to tell the other extension that Joe is
calling to sell you a fridge and hit xfer, the calling party cannot
hear what that person at the extension is saying.  Sometimes the tables
are fully turned, the caller can hear, but the operator can't hear a
thing.

One thing's for sure, if you hit the button quickly (blind transfer) it
works no problem at all.

This is what I see asterisk saying when I transfer the call
unsuccessfully.

== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/82-006d42a0ZOMBIE' in macro 'stdexten'
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/82-006d42a0ZOMBIE'

I've looked at the macro with a fine tooth comb, I cannot see any
problems with it whatsoever, (though that doesn't mean that my ignorance
isn't getting in the way).

I found some mention on the digium mantis bug tracker, here's the link:

http://bugs.digium.com/view.php?id=7421

Before I try and patch the source (which I'm hesitant to do since I run
the debian packages), is there another solution or maybe an unidentified
issue that I haven't been able to decipher?

If there's more information that I can provide to solve this problem,
I'd be happy to do so.

Thank you.
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Re: [asterisk-users] Cisco MWI


Steve Kennedy wrote:

But my mailbox (5200) is in default.
  


I'm pretty sure that you'll still need to include the @context for the 
MWI to work correctly.


Doug

--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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Re: [asterisk-users] Cisco MWI

2006-09-06 Thread Michiel van Baak

On 17:04, Wed 06 Sep 06, Doug Lytle wrote:
 Steve Kennedy wrote:
 But my mailbox (5200) is in default.
   
 
 I'm pretty sure that you'll still need to include the @context for the 
 MWI to work correctly.

Or switch the phone to SCCP. It will give you a lot of extra
power :)
-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?

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Re: [asterisk-users] Cisco MWI

2006-09-06 Thread Aaron Daniel

If you want to leave the @default off, if I remember correctly, you have
to set searchcontexts=yes in voicemail.conf.

On Wed, 2006-09-06 at 23:16 +0200, Michiel van Baak wrote:
 On 17:04, Wed 06 Sep 06, Doug Lytle wrote:
  Steve Kennedy wrote:
  But my mailbox (5200) is in default.

  
  I'm pretty sure that you'll still need to include the @context for the 
  MWI to work correctly.
 
 Or switch the phone to SCCP. It will give you a lot of extra
 power :)
-- 
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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[asterisk-users] faktortel

2006-09-06 Thread Dean Collins









Is there anyone on this list who uses faktortel?



Anyone having problems with incoming iax today?





Cheers,

Dean








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Re: [asterisk-users] Cisco MWI


Michiel van Baak wrote:

On 17:04, Wed 06 Sep 06, Doug Lytle wrote:
  
Or switch the phone to SCCP. It will give you a lot of extra

power :)
  


And segfaults, and phone lockups and grief.

Doug

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


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Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel