RE: [Asterisk-Users] Blank CIDName or CIDNum = asterisk

[gw]

Hello Yusuf,
The idea is I do not want to override the cid, but rather just not have
it show asterisk on unavailable calls.

greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yusuf
Iqbal
Sent: Thursday, August 11, 2005 1:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Blank CIDName or CIDNum = asterisk

Another thing you can do. In your dialplan you can define cid in
incoming call like this..

exten = s,1,SetCallerID(Unavailable)

Thanks
yusuf
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Re: [Asterisk-Users] Help me how to listen voicemail with SIP 7960

[Yusuf Iqbal]

In your settings button 1stly unlock config.Then find SIP
Configuration (option 4) after selecting that go to option 7 that is
Messages URI. Now edit or put the extension.
Thank you
Yusuf

On 8/11/05, Lokesh kumar [EMAIL PROTECTED] wrote:
 Hi,
 Everybody
 
 I am running asterisk successfully, I am having few
 couples of SIP 7960 phones, I am booting the phones
 with P0S-3-06-0-00 file. But i am unable to access
 voicemails through the phone, but i can send voicemail
 attachments with the email, which i mentioned in
 voicemail.conf file.
 The messages button never respond when i press it,
 suggest me how i have to access voicemail boxes
 through SIP 7960 phones.
 I will be very thank full to you
 
 Lokesh
 Portugal
 mail - [EMAIL PROTECTED]
 
 
 
 
 
 
 
 Send a rakhi to your brother, buy gifts and win attractive prizes. Log on to 
 http://in.promos.yahoo.com/rakhi/index.html
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Re: [Asterisk-Users] App_Queue strategy=ringallfree (feature request, possible bounty)

[Florian Overkamp]

Kevin P. Fleming wrote:

Kristian Kielhofner wrote:

Not having looked at the code (like I could make much sense out of 
it anyways), how hard would it be to add something like 
strategy=ringallfree, where only members of the queue not already on a 
call from that queue will receive incoming calls?



We have been suggesting that people implement this sort of thing by 
using Local channels and the dialplan, rather than trying to force more 
complicated logic into app_queue.


By using the dialplan, you can use any method you wish to decide that 
the agent is 'busy'... look in a database, run an AGI, etc.


Hi,

this is a viable option, I have actually defined persistent agents in 
older asterisk versions with that strategy. It does seem to have 
considerable effects on the logging of the calls though, so if queue 
analysis is important you may get more workload than you bargained for.


Florian
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[Asterisk-Users] tdm400p / outbound zap prob

[dan]

I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p
w/ 4 FXO.  Incoming calls work fine, outbound I get this:

-- Executing SetVar(SIP/231-af2b, OUTNUM=6643955) in new stack
-- Executing Cut(SIP/231-af2b, custom=OUT_1|:|1) in new stack
-- Executing GotoIf(SIP/231-af2b, 0?19) in new stack
-- Executing Dial(SIP/231-af2b, ZAP/g0/6643955) in new stack
-- Called g0/6643955
-- Hungup 'Zap/1-1'
  == No one is available to answer at this time
-- Executing Goto(SIP/231-af2b, s-NOANSWER|1) in new stack
-- Goto (macro-dialout-trunk,s-NOANSWER,1)
-- Executing NoOp(SIP/231-af2b, Dial failed due to NOANSWER) in
new stack
-- Executing Macro(SIP/231-af2b, outisbusy) in new stack
-- Executing Playback(SIP/231-af2b,
allison7/all-circuits-busy-now) in new stack
-- Playing 'allison7/all-circuits-busy-now' (language 'en')
-- Executing Playback(SIP/231-af2b, allison7/pls-try-call-later)
in new stack
-- Playing 'allison7/pls-try-call-later' (language 'en')

I'm not sure why (not reboot), but sometimes I get something like this:

-- Executing SetVar(SIP/231-7e98, OUTNUM=6643955) in new stack
-- Executing Cut(SIP/231-7e98, custom=OUT_1|:|1) in new stack
-- Executing GotoIf(SIP/231-7e98, 0?19) in new stack
-- Executing Dial(SIP/231-7e98, ZAP/g0/6643955) in new stack
-- Called g0/6643955
-- Zap/1-1 answered SIP/231-7e98
-- Hungup 'Zap/1-1'
  == Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on
'SIP/231-7e98' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 18005069511, 1) exited non-zero on
'SIP/231-7e98'
-- Executing Macro(SIP/231-7e98, hangupcall) in new stack
-- Executing ResetCDR(SIP/231-7e98, w) in new stack
-- Executing NoCDR(SIP/231-7e98, ) in new stack
-- Executing Wait(SIP/231-7e98, 5) in new stack
  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/231-7e98' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/231-7e98'


In this situation I can hear the call going through for a second or two,
sometimes even hear the other end answer before * hangs up the channel. 
I've tried adding a w before ${ARG2} on line exten = s,11 (in
extensions.conf below)  but this has no effect.  Here's some zapata.conf:

[channels]

language=en
context=from-pstn
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=10.0
txgain=3.0
group=0
callgroup=1
pickupgroup=1
immediate=no
faxdetect=incoming
channel = 1-4


And part of extensions.conf:

[macro-dialout-trunk]
exten = s,1,GotoIf($[foo${ARG3} = foo]?3:2))   ; arg3 is pattern password
exten = s,2,Authenticate(${ARG3})
exten = s,3,Macro(record-enable,${CALLERIDNUM},OUT)
exten = s,4,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?7)  ;check for CID
override for exten
exten = s,5,SetCallerID(${ECID${CALLERIDNUM}})
exten = s,6,Goto(9)
exten = s,7,GotoIf($[foo${OUTCID_${ARG1}} = foo]?9)  ;check for CID
override for trunk
exten = s,8,SetCallerID(${OUTCID_${ARG1}})
exten = s,9,SetGroup(OUT_${ARG1})
exten = s,10,CheckGroup(${OUTMAXCHANS_${ARG1}})
; if we've used up the max channels, continue at 109 (n+101)
exten = s,11,SetVar(DIAL_NUMBER=${ARG2})
exten = s,12,SetVar(DIAL_TRUNK=${ARG1})
exten = s,13,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper
dial string for this trunk
exten = s,14,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER})  ; OUTNUM
is the final dial number
exten = s,15,Cut(custom=OUT_${ARG1},:,1)  ; Custom trunks are prefixed
with AMP:
exten = s,16,GotoIf($[${custom} = AMP]?19)
exten = s,17,Dial(${OUT_${ARG1}}/${OUTNUM})  ; Regular Trunk Dial
exten = s,18,Goto(s-${DIALSTATUS},1)

; This is a custom trunk.  Substitute $OUTNUM$ with the actual number and
rebuild the dialstring
; example trunks: AMP:CAPI/:b$OUTNUM$,30,r,
AMP:OH323/[EMAIL PROTECTED]:
exten = s,19,Cut(pre_num=OUT_${ARG1},$,1)
exten = s,20,Cut(the_num=OUT_${ARG1},$,2)  ; this is where we expect to
find string OUTNUM
exten = s,21,Cut(post_num=OUT_${ARG1},$,3)
exten = s,22,GotoIf($[${the_num} = OUTNUM]?23:24) ; if we didn't find
OUTNUM, then skip to Dial
exten = s,23,SetVar(the_num=${OUTNUM}) ; replace OUTNUM with the actual
number to dial
exten = s,24,Dial(${pre_num:4}${the_num}${post_num})
exten = s,25,Goto(s-${DIALSTATUS},1)

exten = s,111,Noop(max channels used up)
exten = s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten = s-BUSY,2,Busy()
exten = s-BUSY,3,Wait(60)
exten = s-BUSY,4,NoOp()

exten = _s-.,1,NoOp(Dial failed due to ${DIALSTATUS})


Any ideas?

Thanks,

Dan

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[Asterisk-Users] No speech path

[arnab.dey]

Title: Message



Hi,

I am having the
following call configuration,

Xlite --- VOCAL
--- Asterisk --- Extension (Xlite)

I am able to call
from the Xlite SIP phone (registered with VOCAL) to the extension registered
with Asterisk. I hear a ring bank and the call gets connected. But speech path
is not established for some reason. 

Even, I tried out
the following configuration,

Extension1 (Xlite)
--- Asterisk --- Extension2 (Xlite)

Even here, I face
the same problem (no speech path on answering the call).

I checked and
re-checked my sip.conf and extensions.conf file. Everyting seems proper. Also, I
have not come across anybody, who has faced a similar
problem.

Any help /
information will be greatly appreciated.

Thanks,
Arnab.



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[Asterisk-Users] Error while calling

[Joel n.solanki]

Dear all, 

I am getting the below errors when using asterisk. 
I am using sjphone for testing purpose.
Below are the setting for sip.conf and 
extension.conf
When i dial the number it rings on the remote 
telephone. but after ringing 1 time it will disconnect.
Can anybody tell me what does this error means and 
the how to solve this issue.

Thanking You,
Joel

sip.conf
[general]
context=default
port=5060
binaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=g729
allow=g723
allow=ulaw
allow=ilbc

[voip]
type=peer
host=202.202.202.202

and here is the extension.conf. I have placed in 
the middle of extension.conf

exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])exten = 
_X.,2,Hangup

Aug 11 10:15:01 WARNING[11260]: channel.c:2127 
ast_channel_make_compatible: No path to translate from SIP/isphone-8213(256) to 
SIP/200-1264(4)Aug 11 10:15:02 NOTICE[11260]: channel.c:1736 
ast_set_read_format: Unable to find a path from g723 to g729Aug 11 10:15:02 
NOTICE[11260]: channel.c:1703 ast_set_write_format: Unable to find a path from 
g729 to g723 -- SIP/isphone-8213 is making progress 
passing it to SIP/200-1264Aug 11 10:15:02 WARNING[11260]: chan_sip.c:1836 
sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write 
= 256/256)Aug 11 10:15:02 WARNING[11260]: chan_sip.c:1836 sip_write: Asked 
to transmit frame type 4, while native formats is 1 (read/write = 
256/256)Aug 11 10:15:02 WARNING[11260]: chan_sip.c:1836 sip_write: Asked to 
transmit frame type 4, while native formats is 1 (read/write = 
256/256)
Aug 11 10:15:06 
WARNING[11260]: chan_sip.c:1836 sip_write: Asked to transmit frame type 1, while 
native formats is 4 (read/write = 4/4) -- SIP/isphone-8213 
answered SIP/200-1264Aug 11 10:15:06 WARNING[11260]: channel.c:2127 
ast_channel_make_compatible: No path to translate from SIP/200-1264(4) to 
SIP/isphone-8213(1)Aug 11 10:15:06 WARNING[11260]: app_dial.c:1024 
dial_exec: Had to drop call because I couldn't make SIP/200-1264 compatible with 
SIP/isphone-8213 == Spawn extension (default, 14025695651, 1) exited 
non-zero on 'SIP/200-1264'ast*CLI ast*CLI 

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RE: [Asterisk-Users] Error while calling

[Wei Kun]

see error below, try some non-license protocol, such gsm 
first

Aug 11 10:15:02 NOTICE[11260]: 
channel.c:1736 ast_set_read_format: Unable to find a path from g723 to 
g729Aug 11 10:15:06 
WARNING[11260]: app_dial.c:1024 dial_exec: Had to drop call because I couldn't 
make SIP/200-1264 compatible with SIP/isphone-8213 == Spawn extension 
(default, 14025695651, 1) exited non-zero on 'SIP/200-1264'


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[Asterisk-Users] Zultys ZIP 4x5

[scott kerschner]

Hi peoples



Can anyone tell me if the Zultys Zip 4x5 supports iax
protocols or if they have configured one before for iax.



If you have a sample config file that would be great.
Any assistance would be nice



Scott






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[Asterisk-Users] IAX setup

[scott kerschner]

Hi



Is there a preferred setup of IAX/2 with regards to silence
suppression or are there any hidden little secrets that I should
know about?



I have a configuration that works pretty well,
however there are at times silence on the phone and you are not
sure if there person is still connected.



I am using grandstream Budgetone-100 ip phones. Does
this make any sort of difference or is there a setting I need to look at?





Thank you in advance for your help



Scott K








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[Asterisk-Users] Supervised transfer problem with BudgetTone

[Nicolas Schmerber]

Hi all,

I'm quite new on this mailing list, and I discover the asterisk world.
I m experimenting a PBX with SIP phones, grandstream budgetone (not 
expensive for tests)


All the features I need work just not one : the supervised call 
transfers. I know there are a lot of posts about that, but none gave me 
the correct answer (unless I missed it).


Here is my config :

2 sip phones BT102 with firmware : 1.0.6.7 (the last at this day)
Asterisk from debian stable (1.0.7)
a Bri connection arives on the PBX, this works , and can reach the sip 
phones; but the call transfer only works in blind ( no ability to speak 
to the transfee to introduce the incoming call).

Here are config files :

extensions.conf :

NICO = SIP/nico
CEDRIC = SIP/cedric

[default]
include = incoming
exten = 22,1,Dial(${CEDRIC},20)
exten = 23,2,Dial(${NICO},20)

[incoming] ; the BRI stuff
exten = 9692,1,Dial(${CEDRIC},20)   ; if numerber arriving on bri 
finishes by 9692 dial Cedric

exten = _969X,1,Dial(${NICO},20)  ; else dial Nico


features. conf :

[general]
atxfer = *5

So I d like to know the params for the BT phones, the asterisk config , 
and the procedure ( for example should i press *5 when i want to release 
the line and etablish caller = transfee ) and so on .


Thanks in advance







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[Asterisk-Users] * behind NAT, client behind NAT(handytone 286), very strange behavior

[Ohad.Levy]

Hi
All,



I've
an Asterisk Server behind a NAT.

Using
DNAT, I've opened port 5060 and all 1:2 udp.

Sip configured
with externalip and subnet.



I've
another site, also with NAT, where I map the rtp port (as defined in the
client) to map to the local client (DNAT).

Using
Xlite, this configuration works, it requires using the quality=yes and NAT=yes/always
in the sip ext configuration but works quite well.

However,
lately I've purchased a Grandstream ATA Handytone 286 and tried to apply the
same settings but



When
doing an echo test, I can't hear myself, but I can hear the asterisk server
(meaning asterisk can reach the client behind the NAT).

When doing
some tcpdump, it looks like some packets are coming from the client to
asterisk, so the network setting looks ok.

When calling
to another sip device, with or without canreinvite (yes/no) the rtp stream is
unable to establish it self, no matter where the second client is
(inside/outside NAT).



But! When
calling using a zap channel (which is on the asterisk server) everything works!
I can hear the person I'm talking to and he can hear me.



I'm a
bit confused.. How could it be that this works and echo test doesnt?

Any
help would be appreciated!



Thanks,

Ohad






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[Asterisk-Users] app_voicemail.c still looking for config file even I try to configure the voicemail from database.

[Wei Kun]

Hi
I am trying to make asterisk load config from database, so far I get the
sip, extension working, but voicemail seems still looking for config file,
not from the database.

the extconfig.conf looks like
...
sipusers = mysql,asterisk,sip_buddies
sippeers = mysql,asterisk,sip_buddies
extensions = mysql,asterisk,extensions_table
voicemail = mysql,asterisk,voicemail_users
..

the table looks like
mysql select * from voicemail_users;
+--+-+-+-+--+--+
---+--+
| uniqueid | customer_id | context | mailbox | password | fullname | email
| page |
+--+-+-+-+--+--+
---+--+
|1 |2000 | local   |2000 | 4321 | Wei Kun  |
[EMAIL PROTECTED] |  |
|2 |2001 | local   |2001 | 8765 | Wei Kun  |
[EMAIL PROTECTED]  |  |
+--+-+-+-+--+--+
---+--+

But when I call, it prints

Aug 11 15:04:08 WARNING[9316] app_voicemail.c: No entry in voicemail config
file for '2001'

Do I need put something in voicemail.conf to instruct app_voicemail.c to
look it up from database?

Thanks
Kun

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Re: [Asterisk-Users] Calling Extension directly

[Michele \O-Zone\ Pinassi]

On Wednesday 10 August 2005 17:02, Niklas Larsson wrote:
 On Wed, 10 Aug 2005 10:54:20 +0200, O-Zone\ wrote:
  Hi all,
  i'm using Asterisk with several extensions with 7 PSTN lines. Is
  possible, for a caller, to dial directly an extensions ? For
  example, dial something like [PSTN number]*[ext number] ?
 
  Thanks !

 Nope.

 Unless * answers the call and you use a ivr menu with if u know the exten
 dial it now, othervice press

Have you an example how to make a ivr with that function ? And, if possible, 
with a timeout (something like after 15 second you'll be redirect to...)

Thanks a lot ! Oz

-- 

O-Zone ! No (C) 2005
www.zerozone.it
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[Asterisk-Users] Sip ports

[jonny hashem]

i have added port=5060 to sip client configuration but
it seems the same problem and in the same errors:

Aug 11 10:29:18 WARNING[9869]: chan_sip.c:843
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for
seqno 102 (Non-critical Response)


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Re: [Asterisk-Users] USB handset wanted

[Ondrej Valousek]

Matt,

You have forgotten the ringer.
In fact, I don't care that much about LCD  buttons. I want to use it 
with something like X-lite.
Initially, I used machine builtin soundcard with X-Lite (worked well) 
but then I realized that if the phone is supposed to compete with the 
standard analog phone, it must have a working ringer.


From what I see I suppose that every handset with builtin ringer must 
be recongized to the OS as 2 USB soundcards - one for speaker/mike, the 
second as a ringer.

But I could be wrong.

Our company is completely linux based and If I manage, it will have a 
linux based PBX as well (nothing against Windows, though).

Thanks,

Ondrej

Matt Riddell wrote:


Ondrej Valousek wrote:


Hello all asterisk users!

Question: Does anybody know about any good USB handset that would 
understand SIP and Asterisk and will run with Linux?



USB Phones don't understand anything.  They are effectively four 
components:


a) Microphone
b) Speaker
c) LCD Display
d) Buttons

You have to design everything on the client side.  If you don't 
understand USB extensively this would be rater a difficult task.


I have found tons of them, but they are mainly only supported in 
Windows environment.



Because people have written drivers for them (often the manufacturer)

I would like to set up new phone system in our company that would be 
based on asterisk acting as PBX and SIP.



With the clients or the server running Linux?

If you have any suggestions, please let me know. Any help would be much 



Well, it's definitely doable, I have written 2 stacks for usb phones, 
although writing it raw (just via usb access) in Linux would be a 
considerable undertaking.


I would recommend that you:

1) Find a phone where the usb audio device is recognised in Linux, and 
then move towards controlling the LCD and buttons.  If you're lucky, 
the LCD will have something like an HD44870 chip controlling it, but 
bear in mind you're obviously going to need to open it up to check the 
chip.


2) Run a usb sniffer and see what you can get out of the keypad.

3) Write an IAXClient based softphone and include hardware control 
with it.


4) Rinse, Repeat.

:)



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Re: [Asterisk-Users] error compiling asterisk on solaris

[Ondrej Valousek]

www.sunfreeware.com might (and probably will) help
I have just found out that in Solaris 10, it is installed by default in 
/usr/sfw/lib


Ondrej

Rollin Weeks wrote:


Chris,

The problem is that your compiler can't find a library called
libcrypt.so.0.9.7.  This library is apparently needed by
libssl.so.  These are both runtime, shared libraries.  The
result is that you end up with undefined symbols (probably
variables used in services the libraries provide).  You need
to find the encryption library for Solaris 9.

Rollin Weeks

On 8/9/05, *chris* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


hello,
 
 
can anyone help me? im gettitng this error when i tried runnin

make on solaris 9
 
rm -f include/asterisk/version.h.tmp

make[1]: `ast_expr.a' is up to date.
make[1]: Leaving directory `/export/home/fst/chris/cvs/asterisk'
gcc -g  -o asterisk  io.o sched.o logger.o frame.o loader.o
config.o channel.o t
ranslate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o

callerid.o fskmod em.o
image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o dsp.o
chanvars.o 
indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o

dns.o aescrypt. o
aestab.o aeskey.o utils.o config_old.o plc.o jitterbuf.o dnsmgr.o
devicestate. o
netsock.o slinfactory.o strcompat.o ast_expr.a editline/libedit.a
db1-ast/libd b1.a
stdtime/libtime.a -lncurses -lm -lpthread -ldl -lnsl -lsocket
-lresolv -L/u
sr/local/ssl/lib -lssl

/usr/local/sparc-sun-solaris2.8/bin/ld: warning:
libcrypto.so.0.9.6, needed by
/
usr/local/ssl/lib/libssl.so, not found (try using -rpath or

-rpath-link)
utils.o: In function `vasprintf':
/export/home/fst/chris/cvs/asterisk/utils.c:623: undefined
reference to `va_copy '
/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_DigestInit'
/usr/local/ssl/lib/libssl.so: undefined reference to `BIO_find_type'
/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_enc_null'
/usr/local/ssl/lib/libssl.so: undefined reference to
`EVP_CIPHER_CTX_init'
/usr/local/ssl/lib/libssl.so: undefined reference to `X509_NAME_dup'
/usr/local/ssl/lib/libssl.so: undefined reference to
`COMP_compress_block'
/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_rc2_cbc'
/usr/local/ssl/lib/libssl.so: undefined reference to `sk_new_null'
/usr/local/ssl/lib/libssl.so: undefined reference to
`X509_STORE_get_by_subject'
/usr/local/ssl/lib/libssl.so: undefined reference to `lh_free'
/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_VerifyFinal'
/usr/local/ssl/lib/libssl.so: undefined reference to `COMP_CTX_new'
/usr/local/ssl/lib/libssl.so: undefined reference to `sk_dup'
/usr/local/ssl/lib/libssl.so: undefined reference to
`X509_STORE_CTX_set_ex_data
'

/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_DigestFinal'
/usr/local/ssl/lib/libssl.so: undefined reference to `X509_free'
/usr/local/ssl/lib/libssl.so: undefined reference to
`CRYPTO_get_ex_data'
/usr/local/ssl/lib/libssl.so: undefined reference to `BN_bin2bn'
/usr/local/ssl/lib/libssl.so: undefined reference to
`CRYPTO_get_ex_new_index'
/usr/local/ssl/lib/libssl.so: undefined reference to
`PEM_read_bio_RSAPrivateKey
'

/usr/local/ssl/lib/libssl.so: undefined reference to `BN_bn2bin'
/usr/local/ssl/lib/libssl.so: undefined reference to `RAND_add'
/usr/local/ssl/lib/libssl.so: undefined reference to `BIO_s_socket'
/usr/local/ssl/lib/libssl.so: undefined reference to `asn1_add_error'
/usr/local/ssl/lib/libssl.so: undefined reference to
`d2i_RSAPrivateKey'
/usr/local/ssl/lib/libssl.so: undefined reference to `sk_num'
/usr/local/ssl/lib/libssl.so: undefined reference to `BIO_free_all'
/usr/local/ssl/lib/libssl.so: undefined reference to
`BIO_get_retry_reason'
/usr/local/ssl/lib/libssl.so: undefined reference to `X509_STORE_new'
/usr/local/ssl/lib/libssl.so: undefined reference to `SHA1_Init'
/usr/local/ssl/lib/libssl.so: undefined reference to `HMAC_Final'
/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_md5'
/usr/local/ssl/lib/libssl.so: undefined reference to
`ASN1_object_size'
/usr/local/ssl/lib/libssl.so: undefined reference to
`EVP_get_cipherbyname'
/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_rc4'
/usr/local/ssl/lib/libssl.so: 

Re: [Asterisk-Users] Help with calling Perl AGI interface

[Jean-Michel Hiver]

Dan Marino wrote:


I have installed the Perl library from
http://asterisk.gnuinter.net/asterisk-perl and am wondering how I
reference agi-test.agi from extensions.conf

I have added 
exten = s,1,AGI,agi-test.agi

but that doesn't seem to do it.

Is there a certain directory .agi files should be, is that the problem?
 

Depending on your asterisk install, the agi-bin directory can be 
somewhere like /var/lib/asterisk/agi-bin or /usr/share/asterisk/agi-bin


locate agi-bin is your friend :)

Cheers,
Jean-Michel.

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Re: [Asterisk-Users] Supervised transfer problem with BudgetTone

[steve]

On Thu, 11 Aug 2005, Nicolas Schmerber wrote:

 All the features I need work just not one : the supervised call 
 transfers. I know there are a lot of posts about that, but none gave me 
 the correct answer (unless I missed it).


Hi,

You'll need to switch to the CVS-HEAD version of Asterisk in order to have 
supervised transfers.

Steve

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[Asterisk-Users] How to determine elapsed time of a call in progress?

[Michel Koenen]

Hi all,

I need to be able to determining the elapsed time of a call.
I tried commands like 'show channel' or 'zap show channel',  this
outputs a list of parameters including 'elapsed time' but for some
reason this is always '0h0m0s'.
Is this normal or am I looking at the wrong place or using the wrong command?

My configuration:
Asterisk 1.0.7 + Bristuffed
The channel which I tried (and need the information for) is the
channel of an agent who is handling a call via the Queue where agent
is member of. This channel is running via the Zaphfc driver via HFC
PCI in NT mode.

Can you give me a hint to my problem?

Thank you!
Michel Koenen
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Re: [Asterisk-Users] call load balancing

[tim panton]

On 10 Aug 2005, at 16:48, Michiel van Baak wrote:On 08:45, Wed 10 Aug 05, Jean-Michel Hiver wrote: 1) your provider is voluntarily screwing up VoIP traffic2) some idiot purposingly fills up your pipe with UDP traffic If they fill the pipe with TCP traffic, UDP will be dead aswell. Protocols don't matter, bandwidth does.Actually they do. A smart router/firewall can manage inbound TCP traffic by delaying or dropping outbound acks. This will cause anycorrect TCP implementation to back off.Clearly this isn't perfect, it won't help you if you are being DOS'dbut it will throttle inbound http/smtp.Tim.http://www.westhawk.co.uk/  ___
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Re: [Asterisk-Users] error compiling asterisk on solaris

[chris]

hi rollin,

idownloaded openssl from sunfreware.com

i change openssl pkg from openssl-0.9.7g to openssl-0.9.6i hoping that i am
only using the wrong version, but i'm still getting the error,

thnks for the reply rollin, but i believe i have the libcrypto.so.0.9.6 that
is needed.

bash-2.05# cd /usr/local/ssl
bash-2.05# ls
bin  doc  lib  misc private
certsinclude  man  openssl.cnf
bash-2.05# cd lib
bash-2.05# ls
libcrypto.a libcrypto.so.0  libssl.alibssl.so.0
libcrypto.solibcrypto.so.0.9.6  libssl.so   libssl.so.0.9.6
bash-2.05#

i also tried including /usr/local/ssl:/usr/local/ssl/lib on path but i'm
still getting the error.

pls advice if i'm doing the right thing and where can i get encryption
library for Solaris

thnks.
- Original Message -
From: Ondrej Valousek [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, August 11, 2005 3:35 PM
Subject: Re: [Asterisk-Users] error compiling asterisk on solaris


 www.sunfreeware.com might (and probably will) help
 I have just found out that in Solaris 10, it is installed by default in
 /usr/sfw/lib

 Ondrej

 Rollin Weeks wrote:

  Chris,
 
  The problem is that your compiler can't find a library called
  libcrypt.so.0.9.7.  This library is apparently needed by
  libssl.so.  These are both runtime, shared libraries.  The
  result is that you end up with undefined symbols (probably
  variables used in services the libraries provide).  You need
  to find the encryption library for Solaris 9.
 
  Rollin Weeks
 
  On 8/9/05, *chris* [EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED] wrote:
 
  hello,
 
 
  can anyone help me? im gettitng this error when i tried runnin
  make on solaris 9
 
  rm -f include/asterisk/version.h.tmp
  make[1]: `ast_expr.a' is up to date.
  make[1]: Leaving directory `/export/home/fst/chris/cvs/asterisk'
  gcc -g  -o asterisk  io.o sched.o logger.o frame.o loader.o
  config.o channel.o t
  ranslate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o
  callerid.o fskmod em.o
  image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o dsp.o
  chanvars.o
  indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o
  dns.o aescrypt. o
  aestab.o aeskey.o utils.o config_old.o plc.o jitterbuf.o dnsmgr.o
  devicestate. o
  netsock.o slinfactory.o strcompat.o ast_expr.a editline/libedit.a
  db1-ast/libd b1.a
  stdtime/libtime.a -lncurses -lm -lpthread -ldl -lnsl -lsocket
  -lresolv -L/u
  sr/local/ssl/lib -lssl
  /usr/local/sparc-sun-solaris2.8/bin/ld: warning:
  libcrypto.so.0.9.6, needed by
  /
  usr/local/ssl/lib/libssl.so, not found (try using -rpath or
  -rpath-link)
  utils.o: In function `vasprintf':
  /export/home/fst/chris/cvs/asterisk/utils.c:623: undefined
  reference to `va_copy '
  /usr/local/ssl/lib/libssl.so: undefined reference to
`EVP_DigestInit'
  /usr/local/ssl/lib/libssl.so: undefined reference to `BIO_find_type'
  /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_enc_null'
  /usr/local/ssl/lib/libssl.so: undefined reference to
  `EVP_CIPHER_CTX_init'
  /usr/local/ssl/lib/libssl.so: undefined reference to `X509_NAME_dup'
  /usr/local/ssl/lib/libssl.so: undefined reference to
  `COMP_compress_block'
  /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_rc2_cbc'
  /usr/local/ssl/lib/libssl.so: undefined reference to `sk_new_null'
  /usr/local/ssl/lib/libssl.so: undefined reference to
  `X509_STORE_get_by_subject'
  /usr/local/ssl/lib/libssl.so: undefined reference to `lh_free'
  /usr/local/ssl/lib/libssl.so: undefined reference to
`EVP_VerifyFinal'
  /usr/local/ssl/lib/libssl.so: undefined reference to `COMP_CTX_new'
  /usr/local/ssl/lib/libssl.so: undefined reference to `sk_dup'
  /usr/local/ssl/lib/libssl.so: undefined reference to
  `X509_STORE_CTX_set_ex_data
  '
  /usr/local/ssl/lib/libssl.so: undefined reference to
`EVP_DigestFinal'
  /usr/local/ssl/lib/libssl.so: undefined reference to `X509_free'
  /usr/local/ssl/lib/libssl.so: undefined reference to
  `CRYPTO_get_ex_data'
  /usr/local/ssl/lib/libssl.so: undefined reference to `BN_bin2bn'
  /usr/local/ssl/lib/libssl.so: undefined reference to
  `CRYPTO_get_ex_new_index'
  /usr/local/ssl/lib/libssl.so: undefined reference to
  `PEM_read_bio_RSAPrivateKey
  '
  /usr/local/ssl/lib/libssl.so: undefined reference to `BN_bn2bin'
  /usr/local/ssl/lib/libssl.so: undefined reference to `RAND_add'
  /usr/local/ssl/lib/libssl.so: 

[Asterisk-Users] help on receive text

[someshwarak]

Hi * 
users,

I am only seeing 
SendText in the available asterisk applications. But I have not seen Receive 
Text application. I tried on asterisk-1.0.7 and 1.0.9. Can anyonetell me 
how to use this receive text command. 

I want to use 
receivetext command and get text information from an softphone so that that can 
be routed to some other phone supporting text message. (my soft phones are 
SIP/IAX based).


where I can get the 
receive text application for asterisk?
if avialable what 
should be the syntax

Kindly let me know 
howthe configuaration should look like?

thanks,
Somesh
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[Asterisk-Users] music on hold problem

[rkvalmiki]

Hello list

While trying to test the music on hold it shows in the
verbose that it music on hold started and in
background mpg123 also started with that specific file
name

but we could able to listen only the ringing sound
rather instead of music 

in verbose it show s this error 

-- Executing SetMusicOnHold(SIP/6060-08225640,
default) in new stack
-- Executing WaitMusicOnHold(SIP/6060-08225640,
60) in new stack
-- Started music on hold, class 'default', on
SIP/6060-08225640

Aug 11 14:40:42 WARNING[1102653504]: channel.c:1597
ast_prod: Prodding channel 'SIP/6060-08225640' failed

you r help will be highly appreciated.

with regards
rk







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Re: [Asterisk-Users] realtime odbc/mysql eating connections

[Frank Sautter]

Matthew Boehm wrote:
Since you are using ODBC, this seems more likely to be an ODBC issue. If 
you are concerned, you should just use the native MySQL RealTime driver. 
It does not exibit the behavior you mentioned.


Frank Sautter wrote:

our asterisk is configured to retrieve sippeers and iaxpeers via odbc 
from a mysql database. after each call show processlist; within the 
mysql console shows 2 more persistent connections which are showing no 
further activity and will not go away even after restaring asterisk.


well after changing from res_odbc to res_mysql and cdr_odbc to cdr_mysql 
this problem was gone. but after i looked if everything was working ok, 
i found my real problem: the cdr database was somehow corrupted and i 
had to make a 'myisamchk --recover'!


regards
 frank
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Re: [Asterisk-Users] Supervised transfer problem with BudgetTone

[Nicolas Schmerber]

[EMAIL PROTECTED] a écrit :


On Thu, 11 Aug 2005, Nicolas Schmerber wrote:

 

All the features I need work just not one : the supervised call 
transfers. I know there are a lot of posts about that, but none gave me 
the correct answer (unless I missed it).
   




Hi,

You'll need to switch to the CVS-HEAD version of Asterisk in order to have 
supervised transfers.


Steve

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When looking at a recent firmware changelog of Grandstream , it says BT 
should support supervised transfer, so shouldnt it work ?

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[Asterisk-Users] Ignoring the called number in the INVITE message

[Tomáš Komárek]

Hello,

I've got such a problem. I'm configuring Asterisk as a backup server, if 
call to the first one fails.


My problem is, that the redirection from the sending machine work so, 
that in the INVITE line of the INVITE message is the presentation number 
of the Asterisk server and in the To line is the real called number.


So I need to setup Asterisk so, that it will ignore the number in the 
INVITE line and takes care about the To line.


Thanks a lot for the advices

Tomas


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[Asterisk-Users] Sipura-3000 IP-PSTN scenrio

[Arsen Chaloyan]

Hello,
I'm configured Sipura-3000  to forward IP calls to
PSTN number on no answer (In User1 tab Cfwd No Ans
Dest: [EMAIL PROTECTED])

IPPhone  ---IP---  Sipura-3000  ---PSTN---  PSTN
User

Generally it works fine, but Sipura sends back SIP OK
to IPPhone just prior to dialing to PSTN number.
How to configure Sipura to detect that the remote side
on PSTN picks up the phone and only then to send SIP
OK back to IPPhone?

Can you recommend any other device that has such
detection?
How it works?

Thanks, Arsen.
P.S. Sipura-3000 Software Version:3.1.5(GWb)

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[Asterisk-Users] RE: GrandStream GSX-2000 strangeness

[Faris Raouf]

Thanks to all who replied on this.

But amazingly I think I've solved the problem.

Basically I did a factory reset (select reset via the Menu key then enter
the MAC address [as shown on the white label under the phone], then press
Menu key again) and re-entered the necessary config details on both phones.

And this has solved the problem completely (so far). Can you possibly give
it a go to see if it solves your problem too Mark?

What I don't (yet) know is the cause. It could be that the last firmware
update somehow corrupted some of the existing settings, or it could be that
prolonged use causes the problem, requiring a factory reset.

I can still duplicate the sound problems in a way though. If you login to
the phone's web config page, while listening to the phone giving a
dial-tone, I can hear the same type of glitches happening every time I click
on any links.

So it seems that the source of the glitches is probably the phone doing
something internal and getting stuck in a loop.

GrandStream support also replied to my email on this subject, suggesting the
possibility that it may be a hardware problem and asking for me to send them
a copy of the phone's config. 

Faris.


Message: 19
Date: Wed, 10 Aug 2005 22:37:13 +0100
From: Mark Brown [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] GrandStream GSX-2000 strangeness
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain;  charset=us-ascii

I have exactly the same problem with my GSX-2000. And am running the
latest firmware.
Although they seem like really cool phones in theary, practically I
think they still have a far way to go. I personally can't believe they
actually launched the 2000's with all the problems they actually have.
Many of the advertised features on the GS website have still never been
implemented in the actual phones themselves.





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Re: [Asterisk-Users] Supervised transfer problem with BudgetTone

[Olle E. Johansson]

Nicolas Schmerber wrote:
 [EMAIL PROTECTED] a écrit :
 
 On Thu, 11 Aug 2005, Nicolas Schmerber wrote:
 All the features I need work just not one : the supervised call
 transfers. I know there are a lot of posts about that, but none gave
 me the correct answer (unless I missed it).
   

 You'll need to switch to the CVS-HEAD version of Asterisk in order to
 have supervised transfers.

 Steve
 When looking at a recent firmware changelog of Grandstream , it says BT
 should support supervised transfer, so shouldnt it work ?

CVS head of Asterisk supports attended transfers native in Asterisk, not
really SIP attended transfers. Work is in progress in that area, but
will require quite a lot of changes to the SIP channel so I am not sure
whether we will be able to support it in 1.2 or not. Definitely in the
1.4 release.

/Olle
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Re: [Asterisk-Users] Ignoring the called number in the INVITE message

[Olle E. Johansson]

Tomáš Komárek wrote:
 Hello,
 
 I've got such a problem. I'm configuring Asterisk as a backup server, if
 call to the first one fails.
 
 My problem is, that the redirection from the sending machine work so,
 that in the INVITE line of the INVITE message is the presentation number
 of the Asterisk server and in the To line is the real called number.
 
 So I need to setup Asterisk so, that it will ignore the number in the
 INVITE line and takes care about the To line.
 
In CVS head you can reach the To: header with the SIP_HEADER function.

/Olle
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RE: [Asterisk-Users] RE: GrandStream GSX-2000 strangeness

[Mark Brown]

I did a factory reset a while ago, but it didn't make any difference.
This is my second 2000 since the previous one was sent back to the
supplier for intermittent hanging or freezing up during use.






-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: 11 August 2005 11:17
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: GrandStream GSX-2000 strangeness

Thanks to all who replied on this.

But amazingly I think I've solved the problem.

Basically I did a factory reset (select reset via the Menu key then
enter
the MAC address [as shown on the white label under the phone], then
press
Menu key again) and re-entered the necessary config details on both
phones.

And this has solved the problem completely (so far). Can you possibly
give
it a go to see if it solves your problem too Mark?

What I don't (yet) know is the cause. It could be that the last firmware
update somehow corrupted some of the existing settings, or it could be
that
prolonged use causes the problem, requiring a factory reset.

I can still duplicate the sound problems in a way though. If you login
to
the phone's web config page, while listening to the phone giving a
dial-tone, I can hear the same type of glitches happening every time I
click
on any links.

So it seems that the source of the glitches is probably the phone doing
something internal and getting stuck in a loop.

GrandStream support also replied to my email on this subject, suggesting
the
possibility that it may be a hardware problem and asking for me to send
them
a copy of the phone's config. 

Faris.


Message: 19
Date: Wed, 10 Aug 2005 22:37:13 +0100
From: Mark Brown [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] GrandStream GSX-2000 strangeness
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:

[EMAIL PROTECTED]
Content-Type: text/plain;  charset=us-ascii

I have exactly the same problem with my GSX-2000. And am running the
latest firmware.
Although they seem like really cool phones in theary, practically I
think they still have a far way to go. I personally can't believe they
actually launched the 2000's with all the problems they actually have.
Many of the advertised features on the GS website have still never been
implemented in the actual phones themselves.





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[Asterisk-Users] re: how to set the voice message as email attachment ?

[larry lin]

Hi there,

  I am using redhat 9.0 with asterisk 1.0.7.
  I created an user and was be able to leave voice messages to that user 
and retrieve the voice message. I looked the wiki and setup the voice 
message as the email attachment. However, I have never received email with 
the voice attachment. Here is the setting for voicemail.conf:


;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail
;format=g723sf|wav49|wav
;format=wav49|gsm|wav
format=wav
; Who the e-mail notification should appear to come from
[EMAIL PROTECTED]
;[EMAIL PROTECTED]
; Should the email contain the voicemail as an attachment
attach=yes
;Turn on/off envelope playback before message playback. [ON by default]
envelope=yes
; Maximum length of a voicemail message in seconds
;maxmessage=180
; Minimum length of a voicemail message in seconds
;minmessage=3
; Maximum length of greetings in seconds
;maxgreet=60
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred' M 
'hours'


[default]
3114 = 3114,larry lin,[EMAIL PROTECTED],,attach=yes


Do I missing anything ? Thanks in advance.

Larry


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[Asterisk-Users] MS Live Communication Server

[bubuk]

Hi List!

does anyone played around with the LCS and Asterisk? Because the LCS is 
doing no RFC compliant SIP, i wonder if it can work. Google couldn't 
tell me. If someon heared about that, please let me know.


The fact i figured out is that the Border Controler from Jasomi can be 
used as a gateway from MS-LCS-SIP to regular SIP. But that is not really 
handy and expensive too.


Thank you
Volker
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Re: [Asterisk-Users] Ignoring the called number in the INVITE message

[Tomáš Komárek]

Well,

that is great, but I'm not a good programmer, so I would need some 
furher details. Probably I will need to edit the file chan_sip.c and 
then recompile Asterisk.


Is it true??

Would you please advice me?

Thanks in advance.

Tomas

Olle E. Johansson napsal(a):

Tomáš Komárek wrote:


Hello,

I've got such a problem. I'm configuring Asterisk as a backup server, if
call to the first one fails.

My problem is, that the redirection from the sending machine work so,
that in the INVITE line of the INVITE message is the presentation number
of the Asterisk server and in the To line is the real called number.

So I need to setup Asterisk so, that it will ignore the number in the
INVITE line and takes care about the To line.



In CVS head you can reach the To: header with the SIP_HEADER function.

/Olle
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Re: [Asterisk-Users] Ignoring the called number in the INVITE message

[Olle E. Johansson]

Tomáš Komárek wrote:
 Well,
 
 that is great, but I'm not a good programmer, so I would need some
 furher details. Probably I will need to edit the file chan_sip.c and
 then recompile Asterisk.
 
 Is it true??
No, it's a dialplan function in CVS head. You do not need to program
anything. CVS head (the future 1.2) has dialplan applications and
dialplan functions. Please check the sample configuration files and the
help system for more information.

/Olle

---
Astricon 2005 - Early bird registration ends soon!
http://www.astricon.net/2005/
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Re: [Asterisk-Users] MS Live Communication Server

[Jacky]

LCS 2005 just support SIP TCP or TLS right now.
so you must patch asterisk chan_sip.c support TCP,
look http://bugs.digium.com/view.php?id=4903

I have successful call to asterisk's SIP peer or PSTN use Office
Communicator 2005(sign-in my LCS 2005)
but I can't use Dial(SIP/[EMAIL PROTECTED]) , let asterisk's SIP user invite
LCS's user.

Need any input.


2005/8/11, bubuk [EMAIL PROTECTED]:
 Hi List!
 
 does anyone played around with the LCS and Asterisk? Because the LCS is
 doing no RFC compliant SIP, i wonder if it can work. Google couldn't
 tell me. If someon heared about that, please let me know.
 
 The fact i figured out is that the Border Controler from Jasomi can be
 used as a gateway from MS-LCS-SIP to regular SIP. But that is not really
 handy and expensive too.
 
 Thank you
 Volker
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[Asterisk-Users] More then one Tormenta 2 E1/T1 card on system.

[Jarek Jarzebowski]

Hi all,

I am interested in your opinions about using more then one Tormenta 2 
card on asterisk server based on Debian - but distribution does not 
matter in this case I suppose.


--
Jarek
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Re: [Asterisk-Users] Calling Extension directly

[Adam Lewis]

The general idea is that when * answers the call it would play back a
recording of something like If you know the extension of the person
you are trying to reach, you may enter it at any time.  Mike Smith
201, John Michaels 202, Smithy Doe 203... etc.

You can then include the internal extensions context in the IVR answer
context so that all the internal extensions works (be sure that the
internal extensions context doesn't include outgoing dialing,
otherwise people calling you will be able to make long distance calls
on your lines)

There are some sample configs on the wiki (voip-info.org).  One sample
actually does the directory through an AGI script that goes to
database:
http://www.sbuehl.com/projects/asterisk/



On 11/08/05, Michele O-Zone Pinassi [EMAIL PROTECTED] wrote:
 On Wednesday 10 August 2005 17:02, Niklas Larsson wrote:
  On Wed, 10 Aug 2005 10:54:20 +0200, O-Zone\ wrote:
  Hi all,
  i'm using Asterisk with several extensions with 7 PSTN lines. Is
  possible, for a caller, to dial directly an extensions ? For
  example, dial something like [PSTN number]*[ext number] ?
  
  Thanks !
 
  Nope.
 
  Unless * answers the call and you use a ivr menu with if u know the exten
  dial it now, othervice press
 
 Have you an example how to make a ivr with that function ? And, if possible,
 with a timeout (something like after 15 second you'll be redirect to...)
 
 Thanks a lot ! Oz
 
 --
 
 O-Zone ! No (C) 2005
 www.zerozone.it
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[Asterisk-Users] Comedian mail ignores mailbox greetings

[Steve Blair]

Hello:

 We just upgraded to the CVS HEAD release of Asterisk. We migrated over
configuration files from our previous system and most things work as
expected. On issue is that a caller does not get the mailbox specific 
greetings

when they are redirected to voicemail. Instead they get the general system
default greeting. Individual users can record a custom greeting and it is
stored in their mailbox but Comedian mail seems to ignore it.

 Any ideas what might be happening?

Thanks,Steve

--
 
ISC Network Engineering

The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104  



voice: 215-573-8396 


  215-746-8001

fax: 215-898-9348


sip:[EMAIL PROTECTED]

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Re: [Asterisk-Users] real-time priority

[Elwin Andriol]

Joseph wrote:


How to list real-time priority in Linux for an application (example
asterisk)?

 

What do you mean with listing real-time priority? You can list process 
priorities with commands like top or ps -eo pri,nice,%cpu,pid,args 
--sort pri (for example).


If you're interrested in asterisk's real-time responsiveness, the 
following might be of interrest.


Real-time priority actually doesn't exist in Linux (you'll need to use a 
real RTOS for that). Still, Linux makes a destinction between processes 
that need sort of real-time response times and processes that don't. 
Controlling this in a direct way is a difficult, if possible at all. 
Prioritizing processes is done on the fly (in real time) by the 
scheduling process in the Linux core.


However, there is a way to manipulate the prioritizing of processes with 
a command called 'nice'. Normally you use this command (with a positive 
adjustment value) to make a process to behave 'nice' to other processes. 
That is, it gives the process a lower priority that it would normally 
get, thus making it a relative low priority process. By using nice with 
a negative adjustment (you'll need to be root for that), you're able to 
give a certain process a higher priority than it would normally get, 
thus giving the process more of a 'real-time' priority.


In my experience it proved to be more usefull to give all the processes, 
that stood in the way of asterisk performance, a positive nice 
adjustment, rather than giving asterisk a negative nice adjustment. I 
haven't tested this thoroughly, so I'm not sure about the reasons for 
this. It could have something to with asterisk getting in the way of 
Linux's core processes when incresing it's priority. Still, it's nothing 
more than a guess.

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Re: [Asterisk-Users] Comedian mail ignores mailbox greetings

[Julian Lyndon-Smith]

trying changing the permissions on the files in 
/var/spool/asterisk/vm/mailbox


Failing that, remove all the files in /var/spool/asterisk/vm/mailbox

Julian.

Steve Blair wrote:



Hello:

 We just upgraded to the CVS HEAD release of Asterisk. We migrated over
configuration files from our previous system and most things work as
expected. On issue is that a caller does not get the mailbox specific 
greetings
when they are redirected to voicemail. Instead they get the general 
system

default greeting. Individual users can record a custom greeting and it is
stored in their mailbox but Comedian mail seems to ignore it.

 Any ideas what might be happening?

Thanks,Steve



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Re: [Asterisk-Users] does SIP works behind the NAT

[Esben Stien]

Tom Rymes [EMAIL PROTECTED] writes:

 forward port 5060 

Yup, configurable in sip.conf

 ports 1-2

Yup, configurable in rtp.conf

 it could be more complicated than that.

Nope. 

-- 
Esben Stien is [EMAIL PROTECTED] s  a 
 http://www. s tn m
  irc://irc.  b  -  i  .   e/%23contact
  [sip|iax]:   e e 
   jid:b0ef@n n
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Re: [Asterisk-Users] Sip ports

[Esben Stien]

jonny hashem [EMAIL PROTECTED] writes:

 i have 

You really don't say much about what you have. 

-- 
Esben Stien is [EMAIL PROTECTED] s  a 
 http://www. s tn m
  irc://irc.  b  -  i  .   e/%23contact
  [sip|iax]:   e e 
   jid:b0ef@n n
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Re: [Asterisk-Users] USB handset wanted

[Joseph]

On Thu, 2005-08-11 at 09:29 +0200, Ondrej Valousek wrote:
 Matt,
 
 You have forgotten the ringer.
 In fact, I don't care that much about LCD  buttons. I want to use it 
 with something like X-lite.
 Initially, I used machine builtin soundcard with X-Lite (worked well) 
 but then I realized that if the phone is supposed to compete with the 
 standard analog phone, it must have a working ringer.
 
  From what I see I suppose that every handset with builtin ringer must 
 be recongized to the OS as 2 USB soundcards - one for speaker/mike, the 
 second as a ringer.
 But I could be wrong.
 
 Our company is completely linux based and If I manage, it will have a 
 linux based PBX as well (nothing against Windows, though).


With initial testing, it seems like the USB-CS50 from Plantronics should
work. Not sure how you would go about doing the ringer part, but if they
have the headset on, they could here it.

Using the sound card of your system, radio shack sells a switch device
to either send the sound to your headset or to your speakers. It makes
it easy to change back and forth.

-- 
respectfully, Joseph


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Re: [Asterisk-Users] Comedian mail ignores mailbox greetings

[Steve Blair]

Julian Lyndon-Smith wrote:

trying changing the permissions on the files in 
/var/spool/asterisk/vm/mailbox



What should the permissions be?


Failing that, remove all the files in /var/spool/asterisk/vm/mailbox

Julian.

Steve Blair wrote:



Hello:

 We just upgraded to the CVS HEAD release of Asterisk. We migrated over
configuration files from our previous system and most things work as
expected. On issue is that a caller does not get the mailbox specific 
greetings
when they are redirected to voicemail. Instead they get the general 
system
default greeting. Individual users can record a custom greeting and 
it is

stored in their mailbox but Comedian mail seems to ignore it.

 Any ideas what might be happening?

Thanks,Steve



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--
 
ISC Network Engineering

The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104  



voice: 215-573-8396 


  215-746-8001

fax: 215-898-9348


sip:[EMAIL PROTECTED]

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[Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone

[Sean Rima]

I have a brief from a local hotel to build a PBX using Asterisk but they
want to use their exisiting telephones and wiring from an old PBX that
no longer works.

Basically, I can build the system but an looking for a card that will
allow for upto 20 extensions to be wired into the back of the PC. Doeas
anyone know of a solution to this

Sean--
ICQ: 679813FidoNet: 2:263/950
Jabber: [EMAIL PROTECTED] AOL: tcobone
Vodafone Messenger: thecivvie


smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [Asterisk-Users] Supervised transfer problem with BudgetTone

[Eric Wieling aka ManxPower]

[EMAIL PROTECTED] wrote:


On Thu, 11 Aug 2005, Nicolas Schmerber wrote:


All the features I need work just not one : the supervised call 
transfers. I know there are a lot of posts about that, but none gave me 
the correct answer (unless I missed it).




Hi,

You'll need to switch to the CVS-HEAD version of Asterisk in order to have 
supervised transfers.


Not quite correct.  You can do supervised transfers with 1.0.x if your 
phone supports it.  Last I heard GS Budgetone does not support 
supervised transgers.



--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120

r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. r makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.

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Re: [Asterisk-Users] Comedian mail ignores mailbox greetings

[Julian Lyndon-Smith]

I always make them 777 ;)

Your best bet is to remove the files - then when they are recreated, 
they have the default permissions.


Julian

Steve Blair wrote:




Julian Lyndon-Smith wrote:

trying changing the permissions on the files in 
/var/spool/asterisk/vm/mailbox



What should the permissions be?


Failing that, remove all the files in /var/spool/asterisk/vm/mailbox

Julian.

Steve Blair wrote:



Hello:

 We just upgraded to the CVS HEAD release of Asterisk. We migrated over
configuration files from our previous system and most things work as
expected. On issue is that a caller does not get the mailbox 
specific greetings
when they are redirected to voicemail. Instead they get the general 
system
default greeting. Individual users can record a custom greeting and 
it is

stored in their mailbox but Comedian mail seems to ignore it.

 Any ideas what might be happening?

Thanks,Steve



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Re: [Asterisk-Users] Realtime + MYSQL

[Timur V. Elzhov]

On Thu, Aug 11, 2005 at 09:20:36AM -0400, Nathan Alberti wrote:

 I'm having a few issues with the MySQL realtime configuration in 
 CVS-HEAD. I tested it initially with realtime extensions (realtime_ext 
 = mysql,asterisk,extensions) and a realtime switch in extensions.conf 
 and that works fine, So I though I'd go back and test a static 
 configuration mapping.
 
 I used the table structure from the asterisk guru postgres howto to 
 create something similar in MySQL (shown below) and included the 
 following in extconfig;
 
 voicemail.conf = mysql,asterisk,voicemail_users

No, this way is used to store NOT voicemail users info, but ANY
configuration file. To create such a table from the existing config
filr use ast2sql.pl script. Link is given here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Static

But you're probably want to create just the voicemail users config
table:

 MySQL Table
 CREATE TABLE voicemail_users (
 id int NOT NULL auto_increment,
 customer_id varchar(255) NOT NULL default '0',
 context varchar(255) NOT NULL default '',
 mailbox varchar(255) NOT NULL default '',
 password varchar(4) NOT NULL default '0',
 fullname varchar(50) NOT NULL default '',
 email varchar(50) NOT NULL default '',
 pager varchar(50) NOT NULL default '',
 stamp datetime NOT NULL default '-00-00 00:00:00',
 PRIMARY KEY  (`id`)
 );
 ###

So the correct line in extconfig.conf must be

voicemail = mysql,asterisk,voicemail_users

not voicemail.conf = mysql,asterisk,voicemail_users


--
Best regards,
Timur.

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Re: [Asterisk-Users] Supervised transfer problem with BudgetTone

[Eric Wieling aka ManxPower]

Olle E. Johansson wrote:


CVS head of Asterisk supports attended transfers native in Asterisk, not
really SIP attended transfers. Work is in progress in that area, but
will require quite a lot of changes to the SIP channel so I am not sure
whether we will be able to support it in 1.2 or not. Definitely in the
1.4 release.


What is the specific problem?  We hav been doing supervised transfers 
with 1.0.x and Polycom phones for several months.


--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120

r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. r makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.

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RE: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone

[Chad Osmond]

To use the old phones and existing wiring you'll need some E1/T1 FXS
Channel banks  and a T1/E1 Card. Each bank will handle 30/24 phones and
pipe them into a single E1/T1 connection.

You can connect up to 4 (or 8 soon from Sangoma) T1's per card. I really
like the Sangoma cards, there are also Digium cards as well.


The Wiki will have a lot more information regarding Channel Banks and
FXS adapters, I would suggest starting there.

Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Rima
Sent: August 11, 2005 8:34 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Newbie Question: Building an Asterisk system
to replace an old PBX but using existing phone

I have a brief from a local hotel to build a PBX using Asterisk but they
want to use their exisiting telephones and wiring from an old PBX that
no longer works.

Basically, I can build the system but an looking for a card that will
allow for upto 20 extensions to be wired into the back of the PC. Doeas
anyone know of a solution to this

Sean--
ICQ: 679813FidoNet: 2:263/950
Jabber: [EMAIL PROTECTED] AOL: tcobone
Vodafone Messenger: thecivvie
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Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone

[Tom Hayden]

Well, it's unlikely you're going to find a PCI card that can handle
twenty analog lines, however I suggest you look at purchasing a call
bank such as the adit 600.  You then can link up your * server with
the call bank using a T1 card and control and route calls using that
method.

--
Tom Hayden
Astoria Telecom, LLC
www.astoriatelecom.net

On 8/11/05, Sean Rima [EMAIL PROTECTED] wrote:
 I have a brief from a local hotel to build a PBX using Asterisk but they
 want to use their exisiting telephones and wiring from an old PBX that
 no longer works.
 
 Basically, I can build the system but an looking for a card that will
 allow for upto 20 extensions to be wired into the back of the PC. Doeas
 anyone know of a solution to this
 
 Sean--
 ICQ: 679813FidoNet: 2:263/950
 Jabber: [EMAIL PROTECTED] AOL: tcobone
 Vodafone Messenger: thecivvie
 
 
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-- 
Tom
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Re: [Asterisk-Users] Help with calling Perl AGI interface

[Tom Hayden]

I'll second that. Make sure your script is in
/var/lib/asterisk/agi-bin and you have the right permissions on it. I
really just wanted to reply to your post though to congraduate you,
Dan Marino, on your recent induction into the Pro Football Hall of
Fame ;)

--
Tom Hayden
Astoria Telecom, LLC
www.astoriatelecom.net

On 8/11/05, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
 Dan Marino wrote:
 
 I have installed the Perl library from
 http://asterisk.gnuinter.net/asterisk-perl and am wondering how I
 reference agi-test.agi from extensions.conf
 
 I have added
 exten = s,1,AGI,agi-test.agi
 but that doesn't seem to do it.
 
 Is there a certain directory .agi files should be, is that the problem?
 
 
 Depending on your asterisk install, the agi-bin directory can be
 somewhere like /var/lib/asterisk/agi-bin or /usr/share/asterisk/agi-bin
 
 locate agi-bin is your friend :)
 
 Cheers,
 Jean-Michel.
 
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-- 
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Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone

[Andrew Kohlsmith]

On Thursday 11 August 2005 08:34, Sean Rima wrote:
 I have a brief from a local hotel to build a PBX using Asterisk but they
 want to use their exisiting telephones and wiring from an old PBX that
 no longer works.

Can you plug one of the phones into a REGULAR telephone line and get dialtone 
and take and place calls?

If the answer is yes, you can use a te110p (T1 card) and an FXS channel bank 
to connect the phones to Asterisk.

If not, you're SOL unless you can find some kind of proprietary-to-standard 
phone interface, and the chances of that are slim to none.

-A.
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[Asterisk-Users] RE: Sip Ports

[Gene Willingham]

I believe that this message is a failed MWI for voicemail.  I get them on
Cisco phones that have not been configured correctly.  It also could be an
indication of a NAT issue.  The NAT device is shutting down the ports for
the client, and the MWI message could not be delivered.  

The reason I believe it is a MWI is the message says Non-Critical Response.
You could be getting messages that say Critical Response, if so these are
failed inbound calls to the device.  If you give a little more information
about which SIP device, whether it is NATed or Not, whether you have send
similar messages that say Critical Response, we could give more information
about how to debug.

 
 Message: 8
 Date: Thu, 11 Aug 2005 00:24:01 -0700 (PDT)
 From: jonny hashem [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Sip ports
 To: asterisk-users@lists.digium.com
 Message-ID: [EMAIL PROTECTED]
 Content-Type: text/plain; charset=iso-8859-1
 
 i have added port=5060 to sip client configuration but
 it seems the same problem and in the same errors:
 
 Aug 11 10:29:18 WARNING[9869]: chan_sip.c:843
 retrans_pkt: Maximum retries exceeded on call
 [EMAIL PROTECTED] for
 seqno 102 (Non-critical Response)
 
 


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Re: [Asterisk-Users] More then one Tormenta 2 E1/T1 card on system.

[izo]

On 8/11/05, Jarek Jarzebowski [EMAIL PROTECTED] wrote:
 Hi all,
 
 I am interested in your opinions about using more then one Tormenta 2
 card on asterisk server based on Debian - but distribution does not
 matter in this case I suppose.

Its not recommend setup, especially when you need to have echo
cancelation turned on or doing some codec compression.

regards
m.
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RE: [Asterisk-Users] Firewall will definately increase jittersinyourvoice conversation

[Jonathan k. Creasy]

I didn't necessarily mean a separate firewall device, but I wouldn't put
a machine out there without a firewall either between it and the net or
on it (iptables for example)

As far as If I know what I am doing goes, I have not read the source
of everything that *is* required in my environment so how am I to know
it's secure enough not to allow for the creation of a backdoor, rootkit
or anything else? 

Therefore, even when I have taken all other security measures I also
lock down a box with a firewall. Usually, iptables, not a separate
firewall device. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Wednesday, August 10, 2005 11:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Firewall will definately increase
jittersinyourvoice conversation

That's a crack of crap sold by the marketing (not sales) people selling
firewalls. If you know what you're doing, one can very easily secure
any
linux system to function on the Internet (etc) without a firewall. It
all
depends on your level of knowledge/skills on how to disable those items
that are not really needed in your environment. Start with a 'netstat
-a'
to identify those ports that are listening, and shut those items down
that
you don't want exposed.

You can do the same for any MS system as well.



 Wiley is definitely right. It would be dangerous not to have a
firewall
 for security reasons. 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Wiley
 Siler
 Sent: Wednesday, August 10, 2005 2:27 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Firewall will definately increase
jitters
 inyourvoice conversation
 
 Lokesh,
 
 While adding a firewall may add a tiny bit of latency (non-noticeable
by
 the way) it in no way means you are gonna get jitter.  An over
utilized
 data line might cause that but a firewall in and of itself will not.
I
 use a Pix to route my VoIP to an ITSP and I could not be happier.  To
 say that using a firewall causes high latency is incorrect.
 
 Thanks,
 Wiley
 
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Lokesh
 kumar
 Sent: Wednesday, August 10, 2005 10:57 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Firewall will definately increase jitters in
 yourvoice conversation
 
 Hi,
 
 If you will put firewall, then i think you will get high latency and
 consequently you will hear voice jitter in your conversation. so avoid
 putting firewall.
 
 Regards
 Lokesh
 Portugal
 mail [EMAIL PROTECTED]
 
 
   
 
   
   
 
 Send a rakhi to your brother, buy gifts and win attractive prizes. Log
 on to http://in.promos.yahoo.com/rakhi/index.html
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---End of Original Message-


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RE: [Asterisk-Users] Is it mandatory to give power supply toTDM400Pcard

[Jonathan k. Creasy]

Ok, I just unplugged my power connector to a card with 4 FXO modules and
they no longer work. 

Plug it back in and it works.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Thursday, August 11, 2005 12:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Is it mandatory to give power supply
toTDM400Pcard

BS! Fxo modules do not require the power connector, period. The
connector
is only used to develop ringing voltage for fxs modules. It's very easy
to
trace the power connector wiring and find that it goes nowhere when fxo
modules
are used.



 Is it not for a card with 4 FXO? I spent several hours the other day
 trying to figure out what I had done wrong and I ahd forgotten to
 connect the power cable. 
 
 I setup several of these before and couldn't figure out why this one
 didn't work. It appears that's all it waqs. 
 
 Without the power connecter the card will probe, and even appear to be
 working but when the lines ring (coming into the FXO port) it will not
 indicate the ring status to asterisk. 
 
 -Jonathan
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Time
 Bandit
 Sent: Wednesday, August 10, 2005 3:04 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Is it mandatory to give power supply to
 TDM400Pcard
 
  I can modprobe TDM400P card without feeding power
  supply , so what is the purpose of providing power
  supply in that card.
  Can any body tell me
 It is needed if you have FXS ports on it, because the card will need
 to provide ringing voltage to the phone.
 
 hth
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---End of Original Message-


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RE: [Asterisk-Users] will a firewall slow down asterisk?

[Jonathan k. Creasy]

There is pfSense (based on monowall) which I like also. www.pfsense.com

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Sent: Wednesday, August 10, 2005 11:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] will a firewall slow down asterisk?

 I recommend m0n0wall (http://m0n0.ch/wall/)  which is a NetBSD based
 firewall that includes traffic shaping. Easily managed via a web
 interface. Runs on any decent PC with 2 or more NICs. Also on Soekris
 or WRAP embedded platforms.
I recommend IpCop www.ipcop.org
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Re: [Asterisk-Users] Supervised transfer problem with BudgetTone

[Nicolas Schmerber]

Eric Wieling aka ManxPower a écrit :


Olle E. Johansson wrote:


CVS head of Asterisk supports attended transfers native in Asterisk, not
really SIP attended transfers. Work is in progress in that area, but
will require quite a lot of changes to the SIP channel so I am not sure
whether we will be able to support it in 1.2 or not. Definitely in the
1.4 release.



What is the specific problem? We hav been doing supervised transfers 
with 1.0.x and Polycom phones for several months.



Thanks for answering me all, but seems it s a debate to see if it works :)
I m not able to have other phones for the moment, so if this kind of 
transfer doesnt work with Budgetones it doesn't matter, but if someone 
had successfull story with it , I would apreciate much.


Any other idea maybe ?

Nicolas S.
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Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone

[Sean Rima]

Chad Osmond wrote:
 To use the old phones and existing wiring you'll need some E1/T1 FXS
 Channel banks  and a T1/E1 Card. Each bank will handle 30/24 phones and
 pipe them into a single E1/T1 connection.
 
 You can connect up to 4 (or 8 soon from Sangoma) T1's per card. I really
 like the Sangoma cards, there are also Digium cards as well.
 
 
 The Wiki will have a lot more information regarding Channel Banks and
 FXS adapters, I would suggest starting there.

Thanks for this info, I forgot to check the wiki, I am trying to get
them to use IP phones and ditch the old wiring anyway

Sean

-- 
ICQ: 679813FidoNet: 2:263/950
Jabber: [EMAIL PROTECTED] AOL: tcobone
Vodafone Messenger: thecivvie


smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone

[Sean Rima]

Tom Hayden wrote:
 Well, it's unlikely you're going to find a PCI card that can handle
 twenty analog lines, however I suggest you look at purchasing a call
 bank such as the adit 600.  You then can link up your * server with
 the call bank using a T1 card and control and route calls using that
 method.
 

I told them it would be easier and cheaper to ditch the old phones and
wiring to go for dedicated Asterisk phones, I may still go this method
as I need a few for myself anyway

Sean

-- 
ICQ: 679813FidoNet: 2:263/950
Jabber: [EMAIL PROTECTED] AOL: tcobone
Vodafone Messenger: thecivvie


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Description: S/MIME Cryptographic Signature
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Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone

[Sean Rima]

Andrew Kohlsmith wrote:
 On Thursday 11 August 2005 08:34, Sean Rima wrote:
 I have a brief from a local hotel to build a PBX using Asterisk but they
 want to use their exisiting telephones and wiring from an old PBX that
 no longer works.
 
 Can you plug one of the phones into a REGULAR telephone line and get dialtone 
 and take and place calls?
 
 If the answer is yes, you can use a te110p (T1 card) and an FXS channel bank 
 to connect the phones to Asterisk.
 
 If not, you're SOL unless you can find some kind of proprietary-to-standard 
 phone interface, and the chances of that are slim to none.
 

They are standard phones but I also want them to have all the features
that Asterisk does provide, so I may build a bos for my house and show
them that as well

Sean

-- 
ICQ: 679813FidoNet: 2:263/950
Jabber: [EMAIL PROTECTED] AOL: tcobone
Vodafone Messenger: thecivvie


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Description: S/MIME Cryptographic Signature
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Re: [Asterisk-Users] TE205P installation problem - ZT_SPANCONFIG failed on span 1 [Virus checked]

[DRi]

have you checked if the card is recognized by the kernel
...loaded the needed module for the card

to see which modules are actually loaded: lsmod
to see which pci-cards are recognized by the kernel: lspci
...the digium cards are usually detected as an unknown network device

the needed module should be wct2xxp - maybe wct4xxp will do this as well
modules should be installed within /lib/modules/YOUR_KERNEL_VERSION/misc/

 Wichtige Vorabinformation 
bw computer bezieht am 01.09.2005 neue Geschäftsräume. Unsere neue 
Adresse+Rufnummer: 
bw computer
Fangdieckstr. 64
(1. Stock)
22547 Hamburg
T: +49 40 / 49 296 - 0
F: +49 40 / 49 296 - 100 
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RE: [Asterisk-Users] Supervised transfer problem with BudgetTone

[The VoIP Connection]

Section 4.3.7.2 from the Bugetone Manual:

The user can transfer an active call to a third party with announcement.
The user presses the “flash” button and hears a dial tone, then dial the 3rd
party’s phone number followed by pressing send button. If the call is
answered, press “flash” to complete the transfer operation, if the call is
not
answered, pressing “flash” button to resume the original call.

Notes:

• If attended Transfer fails, the BudgeTone phone will ring the user to
remind that
another party is still on the call, the user can then pick up the call using
handset
or speaker.

Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
 

 -Original Message-
 From: Nicolas Schmerber [mailto:[EMAIL PROTECTED] 
 Sent: Thursday, August 11, 2005 5:59 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Supervised transfer problem 
 with BudgetTone
 
 [EMAIL PROTECTED] a écrit :
 
 On Thu, 11 Aug 2005, Nicolas Schmerber wrote:
 
   
 
 All the features I need work just not one : the supervised call 
 transfers. I know there are a lot of posts about that, but 
 none gave 
 me the correct answer (unless I missed it).
 
 
 
 
 Hi,
 
 You'll need to switch to the CVS-HEAD version of Asterisk in 
 order to 
 have supervised transfers.
 
 Steve
 
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 When looking at a recent firmware changelog of Grandstream , 
 it says BT should support supervised transfer, so shouldnt it work ?
 
 

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[Asterisk-Users] Suggestion for VoIP router with QoS

[Bastian Schern]

Hello,

I'm searching for a router for our company. Does anybody has a 
suggestion for a router with a SIP Application Layer Gateway and good 
working QoS (Upstream AND Downstream).


Regards
Bastian
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Re: [Asterisk-Users] Blank CIDName or CIDNum = asterisk

[Hugh L. Johnson]

That worked.  The following line also got rid of asterisk without
entering any custom info:

callerid=

Thank you,
Hugh

On Thu, 2005-08-11 at 03:19 +0100, Tony Hoyle wrote:
 In the [default] section of sip.conf put:
 
 callerid=unavailable


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Re: [Asterisk-Users] error compiling asterisk on solaris

[Derek Whitten]

have you tried compiling openssl by hand?

have you ran 'crle' (http://tinyurl.com/2t9zr)
crle - configure runtime linking environment

you may have to add '/usr/local/ssl' to crle to get solaris to find
those libraries or compile ssl by hand into a 'standard' location


On Thu, 2005-08-11 at 01:34, chris wrote:
 hi rollin,
 
 idownloaded openssl from sunfreware.com
 
 i change openssl pkg from openssl-0.9.7g to openssl-0.9.6i hoping that i am
 only using the wrong version, but i'm still getting the error,
 
 thnks for the reply rollin, but i believe i have the libcrypto.so.0.9.6 that
 is needed.
 
 bash-2.05# cd /usr/local/ssl
 bash-2.05# ls
 bin  doc  lib  misc private
 certsinclude  man  openssl.cnf
 bash-2.05# cd lib
 bash-2.05# ls
 libcrypto.a libcrypto.so.0  libssl.alibssl.so.0
 libcrypto.solibcrypto.so.0.9.6  libssl.so   libssl.so.0.9.6
 bash-2.05#
 
 i also tried including /usr/local/ssl:/usr/local/ssl/lib on path but i'm
 still getting the error.
 
 pls advice if i'm doing the right thing and where can i get encryption
 library for Solaris
 
 thnks.
 - Original Message -
 From: Ondrej Valousek [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Thursday, August 11, 2005 3:35 PM
 Subject: Re: [Asterisk-Users] error compiling asterisk on solaris
 
 
  www.sunfreeware.com might (and probably will) help
  I have just found out that in Solaris 10, it is installed by default in
  /usr/sfw/lib
 
  Ondrej
 
  Rollin Weeks wrote:
 
   Chris,
  
   The problem is that your compiler can't find a library called
   libcrypt.so.0.9.7.  This library is apparently needed by
   libssl.so.  These are both runtime, shared libraries.  The
   result is that you end up with undefined symbols (probably
   variables used in services the libraries provide).  You need
   to find the encryption library for Solaris 9.
  
   Rollin Weeks
  
   On 8/9/05, *chris* [EMAIL PROTECTED]
   mailto:[EMAIL PROTECTED] wrote:
  
   hello,
  
  
   can anyone help me? im gettitng this error when i tried runnin
   make on solaris 9
  
   rm -f include/asterisk/version.h.tmp
   make[1]: `ast_expr.a' is up to date.
   make[1]: Leaving directory `/export/home/fst/chris/cvs/asterisk'
   gcc -g  -o asterisk  io.o sched.o logger.o frame.o loader.o
   config.o channel.o t
   ranslate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o
   callerid.o fskmod em.o
   image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o dsp.o
   chanvars.o
   indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o
   dns.o aescrypt. o
   aestab.o aeskey.o utils.o config_old.o plc.o jitterbuf.o dnsmgr.o
   devicestate. o
   netsock.o slinfactory.o strcompat.o ast_expr.a editline/libedit.a
   db1-ast/libd b1.a
   stdtime/libtime.a -lncurses -lm -lpthread -ldl -lnsl -lsocket
   -lresolv -L/u
   sr/local/ssl/lib -lssl
   /usr/local/sparc-sun-solaris2.8/bin/ld: warning:
   libcrypto.so.0.9.6, needed by
   /
   usr/local/ssl/lib/libssl.so, not found (try using -rpath or
   -rpath-link)
   utils.o: In function `vasprintf':
   /export/home/fst/chris/cvs/asterisk/utils.c:623: undefined
   reference to `va_copy '
   /usr/local/ssl/lib/libssl.so: undefined reference to
 `EVP_DigestInit'
   /usr/local/ssl/lib/libssl.so: undefined reference to `BIO_find_type'
   /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_enc_null'
   /usr/local/ssl/lib/libssl.so: undefined reference to
   `EVP_CIPHER_CTX_init'
   /usr/local/ssl/lib/libssl.so: undefined reference to `X509_NAME_dup'
   /usr/local/ssl/lib/libssl.so: undefined reference to
   `COMP_compress_block'
   /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_rc2_cbc'
   /usr/local/ssl/lib/libssl.so: undefined reference to `sk_new_null'
   /usr/local/ssl/lib/libssl.so: undefined reference to
   `X509_STORE_get_by_subject'
   /usr/local/ssl/lib/libssl.so: undefined reference to `lh_free'
   /usr/local/ssl/lib/libssl.so: undefined reference to
 `EVP_VerifyFinal'
   /usr/local/ssl/lib/libssl.so: undefined reference to `COMP_CTX_new'
   /usr/local/ssl/lib/libssl.so: undefined reference to `sk_dup'
   /usr/local/ssl/lib/libssl.so: undefined reference to
   `X509_STORE_CTX_set_ex_data
   '
   /usr/local/ssl/lib/libssl.so: undefined reference to
 `EVP_DigestFinal'
   /usr/local/ssl/lib/libssl.so: undefined reference to `X509_free'
   /usr/local/ssl/lib/libssl.so: undefined reference to
   `CRYPTO_get_ex_data'
   

Re: [Asterisk-Users] Ignoring the called number in the INVITE message

[Tomáš Komárek]

Well,

I suppose that the dialplan is only in the configuration file called 
extensions.conf.


So I've tried to add the line

exten = _246079020,1,DIAL(SIP/${SIP_HEADER(To)},30,t)

Where I've supposed it would work the way, that when there is an 
incoming call, where in the INVITE line of the message is sip:246079020, 
then there will be ringed the line with the number that is contained in 
the field To. Problem is, this way it does not work. I can not find any 
litereture concerning SIP_HEADER.


Where do i do the mistake???

Thank

Tomas


Olle E. Johansson napsal(a):

Tomáš Komárek wrote:


Well,

that is great, but I'm not a good programmer, so I would need some
furher details. Probably I will need to edit the file chan_sip.c and
then recompile Asterisk.

Is it true??


No, it's a dialplan function in CVS head. You do not need to program
anything. CVS head (the future 1.2) has dialplan applications and
dialplan functions. Please check the sample configuration files and the
help system for more information.

/Olle

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Re: [Asterisk-Users] Is it mandatory to give power supply toTDM400Pcard

[Andrew Kohlsmith]

On Thursday 11 August 2005 09:30, Jonathan k. Creasy wrote:
 Ok, I just unplugged my power connector to a card with 4 FXO modules and
 they no longer work.

You're *sure* you've got FXO modules and not FXS ones?  FXO plug into regular 
phone lines, FXS plug into telephones... 

Unless Digium changed something and now powers both off of that 12V supply...  

-A.
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RE: [Asterisk-Users] USB handset wanted

[Bates, Curtis]

I have been playing with an MV100 from mvox (www.mvox.com) and a Phoenix Audio 
Duet (www.phnxaudio.com).  Both are USB Audio Devices.  With X-Lite, I use them 
like a speakerphone.  I had X-Lite play the ring to the audio device.  I also 
used X-Lite's interface for all interaction with it.  

I like the MV100 for my personal use best.  It is small and cheap ($40 at Radio 
Shack), it also had better echo cancellation.

-Original Message-
From: Ondrej Valousek [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 11, 2005 2:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] USB handset wanted


Matt,

You have forgotten the ringer.
In fact, I don't care that much about LCD  buttons. I want to use it 
with something like X-lite.
Initially, I used machine builtin soundcard with X-Lite (worked well) 
but then I realized that if the phone is supposed to compete with the 
standard analog phone, it must have a working ringer.

 From what I see I suppose that every handset with builtin ringer must 
be recongized to the OS as 2 USB soundcards - one for speaker/mike, the 
second as a ringer.
But I could be wrong.

Our company is completely linux based and If I manage, it will have a 
linux based PBX as well (nothing against Windows, though).
Thanks,

Ondrej

Matt Riddell wrote:

 Ondrej Valousek wrote:

 Hello all asterisk users!

 Question: Does anybody know about any good USB handset that would 
 understand SIP and Asterisk and will run with Linux?


 USB Phones don't understand anything.  They are effectively four 
 components:

 a) Microphone
 b) Speaker
 c) LCD Display
 d) Buttons

 You have to design everything on the client side.  If you don't 
 understand USB extensively this would be rater a difficult task.

 I have found tons of them, but they are mainly only supported in 
 Windows environment.


 Because people have written drivers for them (often the manufacturer)

 I would like to set up new phone system in our company that would be 
 based on asterisk acting as PBX and SIP.


 With the clients or the server running Linux?

 If you have any suggestions, please let me know. Any help would be much 


 Well, it's definitely doable, I have written 2 stacks for usb phones, 
 although writing it raw (just via usb access) in Linux would be a 
 considerable undertaking.

 I would recommend that you:

 1) Find a phone where the usb audio device is recognised in Linux, and 
 then move towards controlling the LCD and buttons.  If you're lucky, 
 the LCD will have something like an HD44870 chip controlling it, but 
 bear in mind you're obviously going to need to open it up to check the 
 chip.

 2) Run a usb sniffer and see what you can get out of the keypad.

 3) Write an IAXClient based softphone and include hardware control 
 with it.

 4) Rinse, Repeat.

 :)


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Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone

[Andrew Kohlsmith]

On Thursday 11 August 2005 09:31, Sean Rima wrote:
 They are standard phones but I also want them to have all the features
 that Asterisk does provide, so I may build a bos for my house and show
 them that as well

Standard phones can still do MWI (if they have a light), call transfers, 
three-way calling... all the good stuff that any Zap channel can provide.

If they have displays and conform to ADSI they can even have soft buttons and 
so on.  I have that at my house.

-A.
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RE: [Asterisk-Users] Blank CIDName or CIDNum = asterisk

[Damon Estep]

So caller ID name is passed when available and nothing is passed when
not?

 
 That worked.  The following line also got rid of asterisk without
 entering any custom info:
 
 callerid=
 
 Thank you,
 Hugh
 
 On Thu, 2005-08-11 at 03:19 +0100, Tony Hoyle wrote:
  In the [default] section of sip.conf put:
 
  callerid=unavailable
 
 
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Re: [Asterisk-Users] dialplan defenition (closer)

[Joao Pereira]

The IP - pbx extension calls are already workin fine.
Now Im just configuring the pbx extension - IP calls this way:

[pbx extensions] --- [SIEMENS PBX]  [ASTERISK] --- [SER] --- [sip 
clients]


Thats why the Dial is for SIP only.

Now Im going to try to get the 118 in Asterisk, because the 74 part is 
being eaten somewere.


Joao Pereira

Armin Schindler wrote:


On Wed, 10 Aug 2005, Joao Pereira wrote:
 


Ok, I m getting to the point,
This route:
exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
Isn't working because the dialed number isnt maching _74XXX

I putted Asterisk in capi debug mode and when I dial 74118 he says:


gnugk*CLI capi debug
CAPI Debugging Enabled
-- CONNECT_IND ID=001 #0x0004 LEN=0078
Controller/PLCI/NCCI= 0x401
CIPValue= 0x10
CalledPartyNumber   = 81118
CallingPartyNumber  = 01 83118
   


...
 


--
I believe that someware 74118 is being transformed in 118... but the number
that apears in this debug is
CalledPartyNumber   = 81118
   



Yes, your number is 'transformed' somewhere. CAPI only gets the '118' to 
dial. 81 is just the numbering plan.


 


How do I get this call?
I already tried:
exten = _81118,1,Dial(SIP/[EMAIL PROTECTED],30,r)
exten = 81118,1,Dial(SIP/[EMAIL PROTECTED],30,r)
   



Where is your dial() for the CAPI line?
Here you dial SIP only?!

Armin

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Re: [Asterisk-Users] USB handset wanted

[Ondrej Valousek]

Ok,

Here is my more detailed vision:
The company has 20-30 engineers sitting behind thin clients powered by 
LTSP using one common login server via XDMCP.

USB handsets are connected to the thin clients.

Now I would like them to use phones so I have 2 options:
- use more advanced sound systems like ESD or maybe ALSA that is able to 
connect to the remote sound server (running on each thin client) and run 
softphone directly on the login server
- forget about ESD/ALSA, stick with OSS (most softphones are only OSS 
aware anyway) and launch softpone software locally on each thin client.


The better is option 1, I think because all apps use the same sound 
device and users can have 1 common headset for everything. No switches, 
no remote logons, no hassle. But it does not solve the ringing issue. So 
I have to forget it.


But these my thoughts assume the headset appear to the system as another 
USB soundcard. That's the bottom line.

Any other oppinions?

Ondrej

Joseph wrote:


On Thu, 2005-08-11 at 09:29 +0200, Ondrej Valousek wrote:
 


Matt,

You have forgotten the ringer.
In fact, I don't care that much about LCD  buttons. I want to use it 
with something like X-lite.
Initially, I used machine builtin soundcard with X-Lite (worked well) 
but then I realized that if the phone is supposed to compete with the 
standard analog phone, it must have a working ringer.


From what I see I suppose that every handset with builtin ringer must 
be recongized to the OS as 2 USB soundcards - one for speaker/mike, the 
second as a ringer.

But I could be wrong.

Our company is completely linux based and If I manage, it will have a 
linux based PBX as well (nothing against Windows, though).
   




With initial testing, it seems like the USB-CS50 from Plantronics should
work. Not sure how you would go about doing the ringer part, but if they
have the headset on, they could here it.

Using the sound card of your system, radio shack sells a switch device
to either send the sound to your headset or to your speakers. It makes
it easy to change back and forth.

 



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Re: [Asterisk-Users] request for clarification on Asterisk T.38 bounty

[Rosario Pingaro]

is it possible to test some patch about T38 passthrough?
In fact we have a t38 tested prvider and a t38 tested ata.

Would you like to share the code?

Thanks
Rosario


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

Sent: Sunday, August 07, 2005 8:40 PM
Subject: Re: [Asterisk-Users] request for clarification on Asterisk T.38 
bounty




Adam Megacz wrote:


The bounty stands at $5,500.  I'm seriously considering taking a shot
at it if I can find a decent T.38 provider to test with (I'm still
hoping for reliable PAYG T.38).

It looks like a lot of very smart people have done a lot of very hard
work (t38modem, spandsp) that would go towards getting this working.
At this point it appears to be mostly a matter of integration
(libspandsp+asterisk), encapsulating T.38 inside IAX2 (not too hard),
and testing (tedious and time-consuming).  Basically the easier but
less-fun part of the big-picture task.

t38modem is of little use for this. It is purely a terminating program. 
spandsp is, of course, applicable as its modems are a core requirement. 
Doing a quick botch up of T.38 isn't too hard. A solid reliable 
implementation takes considerably more effort. Some real R as well as D is 
needed to do it properly. The bounties give no indication of criteria for 
judging completeness.



My main question is this: how is the bounty divided?  Does the person
who does this grunt work get the whole $5,500, or does part of it go
to the authors of t38modem/spandsp (which would surely be a large part
of any solution)?

I think you should forget these bounties. There is nobody administering 
them, so I think the chances of a payout are minimal.



I guess on one hand it would be unjust *not* to divide the bounty with
them, but on the other hand, if the bounty is to be divided, I think
the uncertainty about exactly how that would happen might be a factor
in why the bounty has gone unclaimed for so long.

It has gone unclaimed for so long because the problem is not trivial, and 
I have been too busy with other things to complete my implementation. It 
has been sitting here half finished since the beginning of the year. 
Passthrough is simple, but the interesting things are termination, and 
PSTN gateway operation. The code I have, tidied up, would provide 
UDPTL-to-UDPTL passthrough operation for SIP, which many would find 
useful. Maybe I should tidy and commit it as an interm step. It implements 
the UDPTL transport, with full FEC handling, and offer some simple botches 
to sip.c to make it udptl and T.38 aware. I have most of a gateway and 
termination implementation, too, but it isn't close to being ready to 
commit. I find sip.c is currently too messy to produce anything more than 
a botch for it. A couple of people have said they are reworking sip.c to 
make the addition of new codecs, transports, etc. and their renegotiation 
function smoothly. I haven't seen any results so far. I did only minimal 
work on sip.c in the hope that one those efforts would bear fruit in * 
1.2.


As with many things in *, the licencing forced me to do rather more work 
than necessary. If * were GPL, I could have used some GPL'ed ASN.1 code I 
found. To make code that could be committed to CVS I had to spend quite 
some time rolling my own routines. The final result is faster, but it took 
a lot more effort.


Regards,
Steve

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RE: [Asterisk-Users] Is it mandatory to give power supplytoTDM400Pcard

[Jonathan k. Creasy]

No I got confusedyes they are FXO modules with POTS lines coming
from bell attached. 

The only thing I can think of is that since the card supports 3.3v or 5v
PCI slots that maybe on a 3.3v slot it requires the other connection all
the time because it really does need the 5v and is just not picky about
where it comes from. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Thursday, August 11, 2005 9:44 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Is it mandatory to give power
supplytoTDM400Pcard

On Thursday 11 August 2005 09:30, Jonathan k. Creasy wrote:
 Ok, I just unplugged my power connector to a card with 4 FXO modules
and
 they no longer work.

You're *sure* you've got FXO modules and not FXS ones?  FXO plug into
regular 
phone lines, FXS plug into telephones... 

Unless Digium changed something and now powers both off of that 12V
supply...  

-A.
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Re: [Asterisk-Users] USB handset wanted

[Ondrej Valousek]

Question:
Did they behave like 1 audio device (for the speaker and mike) or 2 
audiodevices (creating /dev/dsp1, /dev/dsp2) with the second one for the 
ringer?

That's what I am unable to find out...
Thanks a lot for the tips

Ondrej

Bates, Curtis wrote:

I have been playing with an MV100 from mvox (www.mvox.com) and a Phoenix Audio Duet (www.phnxaudio.com).  Both are USB Audio Devices.  With X-Lite, I use them like a speakerphone.  I had X-Lite play the ring to the audio device.  I also used X-Lite's interface for all interaction with it.  


I like the MV100 for my personal use best.  It is small and cheap ($40 at Radio 
Shack), it also had better echo cancellation.

-Original Message-
From: Ondrej Valousek [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 11, 2005 2:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] USB handset wanted


Matt,

You have forgotten the ringer.
In fact, I don't care that much about LCD  buttons. I want to use it 
with something like X-lite.
Initially, I used machine builtin soundcard with X-Lite (worked well) 
but then I realized that if the phone is supposed to compete with the 
standard analog phone, it must have a working ringer.


From what I see I suppose that every handset with builtin ringer must 
be recongized to the OS as 2 USB soundcards - one for speaker/mike, the 
second as a ringer.

But I could be wrong.

Our company is completely linux based and If I manage, it will have a 
linux based PBX as well (nothing against Windows, though).

Thanks,

Ondrej

Matt Riddell wrote:

 


Ondrej Valousek wrote:

   


Hello all asterisk users!

Question: Does anybody know about any good USB handset that would 
understand SIP and Asterisk and will run with Linux?
 

USB Phones don't understand anything.  They are effectively four 
components:


   a) Microphone
   b) Speaker
   c) LCD Display
   d) Buttons

You have to design everything on the client side.  If you don't 
understand USB extensively this would be rater a difficult task.


   

I have found tons of them, but they are mainly only supported in 
Windows environment.
 


Because people have written drivers for them (often the manufacturer)

   

I would like to set up new phone system in our company that would be 
based on asterisk acting as PBX and SIP.
 


With the clients or the server running Linux?

   

If you have any suggestions, please let me know. Any help would be much 
 

Well, it's definitely doable, I have written 2 stacks for usb phones, 
although writing it raw (just via usb access) in Linux would be a 
considerable undertaking.


I would recommend that you:

1) Find a phone where the usb audio device is recognised in Linux, and 
then move towards controlling the LCD and buttons.  If you're lucky, 
the LCD will have something like an HD44870 chip controlling it, but 
bear in mind you're obviously going to need to open it up to check the 
chip.


2) Run a usb sniffer and see what you can get out of the keypad.

3) Write an IAXClient based softphone and include hardware control 
with it.


4) Rinse, Repeat.

:)

   



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Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone

[Sean Rima]

Andrew Kohlsmith wrote:
 On Thursday 11 August 2005 09:31, Sean Rima wrote:
 They are standard phones but I also want them to have all the features
 that Asterisk does provide, so I may build a bos for my house and show
 them that as well
 
 Standard phones can still do MWI (if they have a light), call transfers, 
 three-way calling... all the good stuff that any Zap channel can provide.
 
 If they have displays and conform to ADSI they can even have soft buttons and 
 so on.  I have that at my house.
 

Nope nothing like that only basic telephones

Sean

-- 
ICQ: 679813FidoNet: 2:263/950
Jabber: [EMAIL PROTECTED] AOL: tcobone
Vodafone Messenger: thecivvie


smime.p7s
Description: S/MIME Cryptographic Signature
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[Asterisk-Users] TE205P installation problem - ZT_SPANCONFIG failed on span 1

[Cavanna, Richard]

Checked modules and wct4xxp and zaptel are loaded. (if you check the makefile 
wct2xxp is an alias for wct4xxp)

Then did a lspci and it is not sharing any IRQs

Now I am doing a zttest and it hangs on  Opened pseudo zap interface, 
measuring accuracy...

Richard 
--


have you checked if the card is recognized by the kernel
...loaded the needed module for the card

to see which modules are actually loaded: lsmod
to see which pci-cards are recognized by the kernel: lspci
...the digium cards are usually detected as an unknown network device

the needed module should be wct2xxp - maybe wct4xxp will do this as well
modules should be installed within /lib/modules/YOUR_KERNEL_VERSION/misc/

 Wichtige Vorabinformation 
bw computer bezieht am 01.09.2005 neue Geschäftsräume. Unsere neue 
Adresse+Rufnummer: 
bw computer
Fangdieckstr. 64
(1. Stock)
22547 Hamburg
T: +49 40 / 49 296 - 0
F: +49 40 / 49 296 - 100 

--


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Re: [Asterisk-Users] dialplan defenition (goooooooal)

[Joao Pereira]

I got it
The siemens PBX is cutting the 74XXX in XXX, and thats why it wasnt 
working. Now, to implement my dialplan in witch all the SIP phones are 
74XXX, I must put the 74 manually, and the line is:


exten = _XXX,1,Dial(SIP/[EMAIL PROTECTED],5,r)

Thank you to everyone that helped me.
Cheers
Joao Pereira

Joao Pereira wrote:


The IP - pbx extension calls are already workin fine.
Now Im just configuring the pbx extension - IP calls this way:

[pbx extensions] --- [SIEMENS PBX]  [ASTERISK] --- [SER] --- [sip 
clients]


Thats why the Dial is for SIP only.

Now Im going to try to get the 118 in Asterisk, because the 74 part is 
being eaten somewere.


Joao Pereira

Armin Schindler wrote:


On Wed, 10 Aug 2005, Joao Pereira wrote:
 


Ok, I m getting to the point,
This route:
exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
Isn't working because the dialed number isnt maching _74XXX

I putted Asterisk in capi debug mode and when I dial 74118 he says:


gnugk*CLI capi debug
CAPI Debugging Enabled
-- CONNECT_IND ID=001 #0x0004 LEN=0078
Controller/PLCI/NCCI= 0x401
CIPValue= 0x10
CalledPartyNumber   = 81118
CallingPartyNumber  = 01 83118
  


...
 

-- 

I believe that someware 74118 is being transformed in 118... but the 
number

that apears in this debug is
CalledPartyNumber   = 81118
  



Yes, your number is 'transformed' somewhere. CAPI only gets the '118' 
to dial. 81 is just the numbering plan.


 


How do I get this call?
I already tried:
exten = _81118,1,Dial(SIP/[EMAIL PROTECTED],30,r)
exten = 81118,1,Dial(SIP/[EMAIL PROTECTED],30,r)
  



Where is your dial() for the CAPI line?
Here you dial SIP only?!

Armin

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Re: [Asterisk-Users] Supervised transfer problem with BudgetTone

[Nicolas Schmerber]

The VoIP Connection a écrit :


Section 4.3.7.2 from the Bugetone Manual:

The user can transfer an active call to a third party with announcement.
The user presses the “flash” button and hears a dial tone, then dial the 3rd
party’s phone number followed by pressing send button. If the call is
answered, press “flash” to complete the transfer operation, if the call is
not
answered, pressing “flash” button to resume the original call.

Notes:

• If attended Transfer fails, the BudgeTone phone will ring the user to
remind that
another party is still on the call, the user can then pick up the call using
handset
or speaker.

Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]


 


-Original Message-
From: Nicolas Schmerber [mailto:[EMAIL PROTECTED] 
Sent: Thursday, August 11, 2005 5:59 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Supervised transfer problem 
with BudgetTone


[EMAIL PROTECTED] a écrit :

   


On Thu, 11 Aug 2005, Nicolas Schmerber wrote:



 

All the features I need work just not one : the supervised call 
transfers. I know there are a lot of posts about that, but 
   

none gave 
   


me the correct answer (unless I missed it).
  

   


Hi,

You'll need to switch to the CVS-HEAD version of Asterisk in 
 

order to 
   


have supervised transfers.

Steve

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When looking at a recent firmware changelog of Grandstream , 
it says BT should support supervised transfer, so shouldnt it work ?



   



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Tried this manipulation a few minutes ago :

A calls B , B pushes flash button ( A is waiting with a mp3 played)
B calls C pressing Send ;
C answers
B presses flash button again ;
C is so on hold (with a mp3 played)
B hangs up
But A and C arent in connect ; the phoneof B rings ( to tell someone is 
in wait : A)


So it seems to fail

What should i put in grandstream config for the next item :
/Enable Call Features: Y/ N ?
//Disable Call-Waiting: Y/N ?
//Send DTMF: / in-audio / via RTP (RFC2833) / via SIP INFO
/Send Flash Event: Y / N ? /
Any others Ideas ?.

Thx

Nicolas S.
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Re: [Asterisk-Users] Suggestion for VoIP router with QoS

[c waddy]

We have been using the Ingate Firewalls and they work very well with SIP  QOS.

On 8/11/05, Bastian Schern [EMAIL PROTECTED] wrote:
 Hello,
 
 I'm searching for a router for our company. Does anybody has a
 suggestion for a router with a SIP Application Layer Gateway and good
 working QoS (Upstream AND Downstream).
 
 Regards
Bastian
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Re: [Asterisk-Users] Realtime + MYSQL

[Rollin Weeks]

Damon,

You may be querying the wrong table, because the following fields in your Select statement do not exit
in the table, voicemail_users, that you created:

 category,
 var_name,
 var_val,
 cat_metric,
 filename,
 commented

Every item mentioned in a Select query must exist in the table that is being queried.

Rollin Weeks

 
 On 8/10/05, Damon Estep [EMAIL PROTECTED] wrote:
 I'm having a few issues with the MySQL realtime configuration in CVS-HEAD. I tested it initially with realtime extensions (realtime_ext = mysql,asterisk,extensions) and a realtime switch in extensions.conf
 and that works fine, So I though I'd go back and test a static configuration mapping. I used the table structure from the asterisk guru postgres howto to create something similar in MySQL (shown below) and included the
 following in extconfig; voicemail.conf = mysql,asterisk,voicemail_users The result is that app_voicemail fails to load and it appears from the debug that it is not happy with the table structure... however the
names it has for the fields seem strange (to me that is :)) If anyone has gone through the process of creating the correct tablesin MySQL and doesn't mind sharing I would be most appreciative.
 Regards, Nathan. MySQL Table CREATE TABLE voicemail_users ( id int NOT NULL auto_increment, customer_id varchar(255) NOT NULL default '0',
 context varchar(255) NOT NULL default '', mailbox varchar(255) NOT NULL default '', password varchar(4) NOT NULL default '0', fullname varchar(50) NOT NULL default '', email varchar(50) NOT NULL default '',
 pager varchar(50) NOT NULL default '', stamp datetime NOT NULL default '-00-00 00:00:00', PRIMARY KEY(`id`) ); ### res_mysql.conf [general]
 dbhost = localhost dbname = asterisk dbuser = asterisk dbpass =  dbport = 3306 dbsock = /var/run/mysqld/mysqld.sock  Debug Log
 Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Static SQL: SELECT category, var_name, var_val, cat_metric FROMvoicemail_users WHERE filename='voicemail.conf' and commented=0 ORDER BY filename,
 cat_metric desc, var_metric asc, category, var_name, var_val, id Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Everything is fine. Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Query:
 SELECT category, var_name, var_val, cat_metric FROM voicemail_users WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id
 Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Query Failed because: Unknown column 'category' in 'field list'  ___
This works for voicemail in CVS-HEADCREATE TABLE `voicemail` (`uniqueid` int(11) NOT NULL auto_increment,`customer_id` int(11) NOT NULL default '0',`context` varchar(50) NOT NULL default '',`mailbox` varchar(10) NOT NULL default '0',
`password` varchar(4) NOT NULL default '0',`fullname` varchar(50) NOT NULL default '',`email` varchar(50) NOT NULL default '',`pager` varchar(50) NOT NULL default '',`stamp` timestamp NOT NULL default CURRENT_TIMESTAMP on update
CURRENT_TIMESTAMP,PRIMARY KEY(`uniqueid`),KEY `mailbox_context` (`mailbox`,`context`)) ENGINE=MyISAM DEFAULT CHARSET=latin1;___Asterisk-Users mailing list
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Re: [Asterisk-Users] Is it mandatory to give power supplytoTDM400Pcard

[Kevin P. Fleming]

Jonathan k. Creasy wrote:


The only thing I can think of is that since the card supports 3.3v or 5v
PCI slots that maybe on a 3.3v slot it requires the other connection all
the time because it really does need the 5v and is just not picky about
where it comes from. 


No, that is not correct. The +5V power pin on the Molex connector is not 
even wired to anything on the board, it is completely ignored.


As the others have already said, the FXO modules do not use the +12V 
power feed at all. The TDM400P with only FXO modules installed works 
just fine without the auxiliary power connector connected to a power supply.

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Re: [Asterisk-Users] dialplan defenition (goooooooal)

[Eric Wieling aka ManxPower]

Joao Pereira wrote:

I got it
The siemens PBX is cutting the 74XXX in XXX, and thats why it wasnt 
working. Now, to implement my dialplan in witch all the SIP phones are 
74XXX, I must put the 74 manually, and the line is:


exten = _XXX,1,Dial(SIP/[EMAIL PROTECTED],5,r)


Don't use r.

r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. r makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.



--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120

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[Asterisk-Users] is there cdrs for sip

[jonny hashem]

i have used astcc to open accounts to clients but now
i dont want to use astcc and i want to use sip cdrs.
 

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Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone

[Tom Rymes]

On Aug 11, 2005, at 10:35 AM, Sean Rima wrote:


Andrew Kohlsmith wrote:


On Thursday 11 August 2005 09:31, Sean Rima wrote:

They are standard phones but I also want them to have all the  
features
that Asterisk does provide, so I may build a bos for my house and  
show

them that as well



Standard phones can still do MWI (if they have a light), call  
transfers,
three-way calling... all the good stuff that any Zap channel can  
provide.


If they have displays and conform to ADSI they can even have soft  
buttons and

so on.  I have that at my house.


Nope nothing like that only basic telephones

Sean


This may be heresy for some, but  I would look into [EMAIL PROTECTED] for a  
reasonably sized hotel. It has wakeup calls  weather built-in, easy  
for the hotel to configure, etc, and despite the home in the name,  
it is solid and robust. Contrary to popular belief, you can also  
extend it as needed by using the extensions_custom.conf file.


Tom
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[Asterisk-Users] Firefly Problem

[David Choo]

Hi All,

I'm facing a very funny situtation when dealing with Firefly. When the
firefly extensions are being dialed, Firefly will hear 1 ring, before
hearing the called party's voice, all while the called party is hearing the
dialing tones.

When Firefly picks up the calls accordingly, the calls will be able to go
through like normal, but * don't seem to detect that the called has gone
through. After 20 seconds, the calls will be dropped for some reasons. As
though its not correct. Do note that it don't seem to be a protocol
problem, as IAXComm don't have this issue.

Here is the iax debug

 Start IAX Debug =

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
   Timestamp: 00016ms  SCall: 00042  DCall: 2 [202.156.XXX.XXX:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
   Timestamp: 0ms  SCall: 00042  DCall: 2 [202.156.XXX.XXX:4569]
   FORMAT  : 2
asterisk*CLI
-- Call accepted by 202.156.XXX.XXX (format gsm)
-- Format for call is gsm
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
   Timestamp: 00062ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass:
RINGING
   Timestamp: 0ms  SCall: 00042  DCall: 2 [202.156.XXX.XXX:4569]
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 00066ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: VOICE   Subclass: 2
   Timestamp: 00080ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 002 Type: VOICE   Subclass: 2
   Timestamp: 00080ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 00080ms  SCall: 00042  DCall: 2 [202.156.XXX.XXX:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass:
ANSWER
   Timestamp: 04892ms  SCall: 00042  DCall: 2 [202.156.XXX.XXX:4569]
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK
   Timestamp: 04892ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 002 Type: VOICE   Subclass: 2
   Timestamp: 04941ms  SCall: 00042  DCall: 2 [202.156.XXX.XXX:4569]
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass: ACK
   Timestamp: 04941ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass:
LAGRQ
   Timestamp: 10032ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass:
LAGRP
   Timestamp: 10032ms  SCall: 00042  DCall: 2 [202.156.XXX.XXX:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: ACK
   Timestamp: 10032ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: PING
   Timestamp: 20021ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass:
LAGRQ
   Timestamp: 20024ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
Rx-Frame Retry[ No] -- OSeqno: 005 ISeqno: 004 Type: IAX Subclass: PONG
   Timestamp: 20021ms  SCall: 00042  DCall: 2 [202.156.XXX.XXX:4569]
Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 006 Type: IAX Subclass: ACK
   Timestamp: 20021ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
Rx-Frame Retry[ No] -- OSeqno: 006 ISeqno: 005 Type: IAX Subclass:
LAGRP
   Timestamp: 20024ms  SCall: 00042  DCall: 2 [202.156.XXX.XXX:4569]
Tx-Frame Retry[-01] -- OSeqno: 005 ISeqno: 007 Type: IAX Subclass: ACK
   Timestamp: 20024ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
-- Nobody picked up in 2 ms
-- Hungup 'IAX2/892-2'
Tx-Frame Retry[000] -- OSeqno: 005 ISeqno: 007 Type: IAX Subclass:
HANGUP
-- Executing Goto(Zap/2-1, s-NOANSWER|1) in new stack
   Timestamp: 21081ms  SCall: 2  DCall: 00042 [202.156.XXX.XXX:4569]
-- Goto (macro-stdexten,s-NOANSWER,1)
   CAUSE CODE  : 0

 End IAX Debug =

Best Regards,

==
David Choo
Sales Engineer
Business  Technology Division
Engineered for Changing Businesses
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-6842 2725, Ext - 404
Fax : 65-6842 2724
E-mail :[EMAIL PROTECTED]
=

Privileged/Confidential information may be contained in this message. If
you are not the intended recipient, you must not copy it or use it for any
purpose, nor deliver this message to anyone. Instead, please delete this
message and destroy any other record of it immediately and kindly notify
the sender by return email. Thank you for your co-operation.

Internet 

Re: [Asterisk-Users] Zultys ZIP 4x5

[Eric Wieling aka ManxPower]

scott kerschner wrote:

Hi peoples

 


Can anyone tell me if the Zultys Zip 4x5 supports iax protocols or if they
have configured one before for iax.


Zultys products do not support IAX.  What in the world made you think 
they did?


--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120

r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. r makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.

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Re: [Asterisk-Users] re: how to set the voice message as email attachment ?

[Gurminder Arora]

I think you can add these lines in voicemail.conf


emailsubject=[VMBOX]:New message ${VM_DATE}
emailbody=Hello ${VM_NAME}:\n\tYou have a new voice
message.\n\tMessage Duration: ${VM_DUR} mins\n\tCaller ID:
${VM_CALLERID}\n\t( !)\n\t Date: ${VM_DATE}. \nThanks!\n--The Netlabs
SoftCall Service\n
filename='voicemail'
attach=yes
saycid=yes
sendvoicemail=yes
review=yes
operator=yes
delete=yes



Do tell me it works 
Bye
Gurminder

On 8/11/05, larry lin [EMAIL PROTECTED] wrote:
 Hi there,
 
I am using redhat 9.0 with asterisk 1.0.7.
I created an user and was be able to leave voice messages to that user
 and retrieve the voice message. I looked the wiki and setup the voice
 message as the email attachment. However, I have never received email with
 the voice attachment. Here is the setting for voicemail.conf:
 
 ;
 ; Voicemail Configuration
 ;
 [general]
 ; Default formats for writing Voicemail
 ;format=g723sf|wav49|wav
 ;format=wav49|gsm|wav
 format=wav
 ; Who the e-mail notification should appear to come from
 [EMAIL PROTECTED]
 ;[EMAIL PROTECTED]
 ; Should the email contain the voicemail as an attachment
 attach=yes
 ;Turn on/off envelope playback before message playback. [ON by default]
 envelope=yes
 ; Maximum length of a voicemail message in seconds
 ;maxmessage=180
 ; Minimum length of a voicemail message in seconds
 ;minmessage=3
 ; Maximum length of greetings in seconds
 ;maxgreet=60
 central=America/Chicago|'vm-received' Q 'digits/at' IMp
 central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred' M
 'hours'
 
 [default]
 3114 = 3114,larry lin,[EMAIL PROTECTED],,attach=yes
 
 
 Do I missing anything ? Thanks in advance.
 
 Larry
 
 
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[Asterisk-Users] SIP_HEADER help

[Tomáš Komárek]

Hello,

I have a problem with the SIP_HEADER function. Can anybody help me with 
the usage?


I need to dial an extension with the number that is in the To field 
instead of the one, that in THE INVITE field.


I'm trying something like
exten = 246.,1,DIAL(SIP/${SIP_HEADER(To)},30,t)

but it does not work. I get the result

-- Executing Dial(SIP/195.122.207.106-f4103dc8, SIP/|30|t) in new stack

which is wrong..

Thanks for the advices.

Tomas
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Re: [Asterisk-Users] app_voicemail.c still looking for config file even I try to configure the voicemail from database.

[Matthew Boehm]

Wei Kun wrote:

Hi
I am trying to make asterisk load config from database, so far I get the
sip, extension working, but voicemail seems still looking for config file,
not from the database.
Aug 11 15:04:08 WARNING[9316] app_voicemail.c: No entry in voicemail config
file for '2001'


How exactly are you calling it? Are you specifying the right voicemail 
context? What did the debug log say?


-Matthew

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Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone

[Sean Rima]

Tom Rymes wrote:
 On Aug 11, 2005, at 10:35 AM, Sean Rima wrote:
 
 Andrew Kohlsmith wrote:

 On Thursday 11 August 2005 09:31, Sean Rima wrote:

 They are standard phones but I also want them to have all the  features
 that Asterisk does provide, so I may build a bos for my house and  show
 them that as well


 Standard phones can still do MWI (if they have a light), call 
 transfers,
 three-way calling... all the good stuff that any Zap channel can 
 provide.

 If they have displays and conform to ADSI they can even have soft 
 buttons and
 so on.  I have that at my house.

 Nope nothing like that only basic telephones

 Sean
 
 This may be heresy for some, but  I would look into [EMAIL PROTECTED] for a 
 reasonably sized hotel. It has wakeup calls  weather built-in, easy 
 for the hotel to configure, etc, and despite the home in the name,  it
 is solid and robust. Contrary to popular belief, you can also  extend it
 as needed by using the extensions_custom.conf file.
 

I will have a look at that and see if it helps, byt the sounds itmay

Sean

-- 
ICQ: 679813FidoNet: 2:263/950
Jabber: [EMAIL PROTECTED] AOL: tcobone
Vodafone Messenger: thecivvie


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Re: [Asterisk-Users] Is it mandatory to give power supplytoTDM400Pcard

[Steve Maroney]

The power connector is used to supply ringing voltage when fxs
modules are used.

Thank you,
Steve Maroney

On Thu, 11 Aug 2005, Kevin P. Fleming wrote:

 Jonathan k. Creasy wrote:

  The only thing I can think of is that since the card supports 3.3v or 5v
  PCI slots that maybe on a 3.3v slot it requires the other connection all
  the time because it really does need the 5v and is just not picky about
  where it comes from.

 No, that is not correct. The +5V power pin on the Molex connector is not
 even wired to anything on the board, it is completely ignored.

 As the others have already said, the FXO modules do not use the +12V
 power feed at all. The TDM400P with only FXO modules installed works
 just fine without the auxiliary power connector connected to a power supply.
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Re: [Asterisk-Users] Press # to continue / Findme

[Time Bandit]

 Any ideas?  Using background dosen;t work, because you hit # and it
 hangs up.
I think you have to define a # extension in your macro, something like
exten = #,1,Playback(not-available)
exten = #,2,Goto(somewhere)

If I'm wrong, please someone correct me

hth
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Re: [Asterisk-Users] Realtime + MYSQL

[Matthew Boehm]

Timur V. Elzhov wrote:


So the correct line in extconfig.conf must be

voicemail = mysql,asterisk,voicemail_users


Yes, Timur is correct. By stating that you want to bind voicemail.conf 
you mean you want to store the config file itself. This is not what you 
are looking for. Change the line above to what Timur says and it should 
work fine.


-Matthew

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[Asterisk-Users] meetme.conf and realtime

[Dave Kettmann]

Hi all,

I am kinda of confused on how the table should look. For the sip.conf it isnt 
too hard to figure out the layout of the database. Has anyone used realtime 
with meetme.conf? I cant figure out the layout of the DB as it doesnt have 
multiple entries like the sip.conf does. I have searched the archives and 
havent found any help with this.

Any help is appreciated,

Thanks,

Dave Kettmann
NetLogic
314-266-4000
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RE: [Asterisk-Users] Firewall will definately increase jittersinyourvoice conversation

[Wiley Siler]

The question was not can I secure a Linux box without a hardware
firewall.  The question (or statement really) was will a firewall add
jitter and lower performance.  That answer is obviously a big NO.  Can
you secure a Linux (or even Windows) machine by closing ports?  Sure.
It helps immensely.  However, an advantage of hardware is that you are
physically separating the traffic from the end point.  Sure, all the
ports closed on a Linux box can protect that machine.  However, having
only web (for example) traffic going to your Apache server is really
beneficial.  The server can focus on delivering pages and not spend any
CPU cycles on is this a good packet?  Should I drop it?.  A firewall
(software or hardware) should also be able to better deal with DOS and
things of that nature. Port securing does nothing to assist with DOS.

So...  You are totally right, you can secure a box that way.  However, a
firewall (be it software or hardware) is far superior a method.  I
prefer the hardware method myself as it is a matter of management and
additional features.  However, for some, a software method may be
better.  I ran Mandrake SNF (a shorewall implementation) for a long time
so I have been there.  Considering you can run a Linux firewall on a 386
machine worth $20 makes the fact that so many people don't have
firewalls seem just ridiculous.

W


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Wednesday, August 10, 2005 8:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Firewall will definately increase
jittersinyourvoice conversation

That's a crack of crap sold by the marketing (not sales) people selling
firewalls. If you know what you're doing, one can very easily secure
any linux system to function on the Internet (etc) without a firewall.
It all depends on your level of knowledge/skills on how to disable those
items that are not really needed in your environment. Start with a
'netstat -a'
to identify those ports that are listening, and shut those items down
that you don't want exposed.

You can do the same for any MS system as well.



 Wiley is definitely right. It would be dangerous not to have a 
 firewall for security reasons.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Wiley 
 Siler
 Sent: Wednesday, August 10, 2005 2:27 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Firewall will definately increase 
 jitters inyourvoice conversation
 
 Lokesh,
 
 While adding a firewall may add a tiny bit of latency (non-noticeable 
 by the way) it in no way means you are gonna get jitter.  An over 
 utilized data line might cause that but a firewall in and of itself 
 will not.  I use a Pix to route my VoIP to an ITSP and I could not be 
 happier.  To say that using a firewall causes high latency is
incorrect.
 
 Thanks,
 Wiley
 
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Lokesh 
 kumar
 Sent: Wednesday, August 10, 2005 10:57 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Firewall will definately increase jitters in

 yourvoice conversation
 
 Hi,
 
 If you will put firewall, then i think you will get high latency and 
 consequently you will hear voice jitter in your conversation. so avoid

 putting firewall.
 
 Regards
 Lokesh
 Portugal
 mail [EMAIL PROTECTED]
 
 
   
 
   
   
 
 Send a rakhi to your brother, buy gifts and win attractive prizes. Log

 on to http://in.promos.yahoo.com/rakhi/index.html
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---End of Original Message-


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Re: [Asterisk-Users] ISDN DID

[Panitaxx]

Hello ,

Thank you for every response. It was the telcos fault. They told me
they were sending it, but they wer not.

regards,

ia

On 8/10/05, Johann Steinwendtner [EMAIL PROTECTED] wrote:
 There is no called party ie but sending complete ie included in the
 setup message. Hence, it tries to terminate.
 
 
 Best regards
 Hans
 
 Paul Belanger schrieb:
  Where are your calls coming from?  Are you connected to the Telco or PBX?
 
  PB
 
  Panitaxx wrote:
 
  Hi,
 
  thanks for your response. here is the log of one call:
 
  Enabled debugging on span 1
 
  Asterisk*CLI
   Protocol Discriminator: Q.931 (8)  len=33
   Call Ref: len= 2 (reference 72/0x48) (Originator)
   Message type: SETUP (5)
   [a1]
   Sending Complete (len= 1)
   [04 03 90 90 a3]
   Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
  capability: 3.1kHz audio (16)
Ext: 1  Trans mode/rate: 64kbps,
  circuit-mode (16)
Ext: 1  User information layer 1: A-Law
  (35)
   [18 03 a9 83 8d]
   Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
  Exclusive Dchan: 0
  ChanSel: Reserved
 Ext: 1  Coding: 0   Number Specified   Channel
  Type: 3
 Ext: 1  Channel: 13 ]
   [1e 02 84 83]
   Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
  (0) 0: 0   Location: Public network serving the remote user (4)
 Ext: 1  Progress Description: Calling
  equipment is non-ISDN. (3) ]
   [6c 0b 00 83 39 31 35 34 35 31 39 30 30]
   Calling Number (len=13) [ Ext: 0  TON: Unknown Number Type (0)  NPI:
  Unknown Number Plan (0)
 Presentation: Presentation allowed of
  network provided number (3) '915451900' ]
  -- Making new call for cr 72
  -- Processing Q.931 Call Setup
  -- Processing IE 161 (cs0, Sending Complete)
  -- Processing IE 4 (cs0, Bearer Capability)
  -- Processing IE 24 (cs0, Channel Identification)
  -- Processing IE 30 (cs0, Progress Indicator)
  -- Processing IE 108 (cs0, C
  alling Party Number)
  -- Going to extension s|1 because of Complete received
 
 
  Protocol Discriminator: Q.931 (8)  len=10
  Call Ref: len= 2 (reference 32840/0x8048) (Terminator)
  Message type: CALL PROCEEDING (2)
  [18 03 a9 83 8d]
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
  Exclusive Dchan: 0
ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel
  Type: 3
   Ext: 1  Channel: 13 ]
 
 
  -- Accepting call from '915451900' to 's' on channel 0/13, span 1
 
  Asterisk*CLI -- Executing Playback(Zap/13-1,
  vm-intro|noanswer) in new stack
 
 
  Protocol Discriminator: Q.931 (8)  len=9
  Call Ref: len= 2 (reference 32840/0x8048) (Terminator)
  Message type: PROGRESS (3)
  [1e 02 81 88]
  Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
  (0) 0: 0   Location: Private network serving the local user (1)
   Ext: 1  Progress Description: Inband
  information or appropriate pattern now available. (8) ]
 
 
  -- Playing 'vm-intro' (language 'es')
 
  Asterisk*CLI -- Executing Playback(Zap/13-1, vm-goodbye) in
  new stack
 
 
  Protocol Discriminator: Q.931 (8)  len=14
  Call Ref: len= 2 (reference 32840/0x8048) (Terminator)
  Message type: CONNECT (7)
  [18 03 a9 83 8d]
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
  Exclusive Dchan: 0
ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel
  Type: 3
   Ext: 1  Channel: 13 ]
  [1e 02 81 82]
  Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
  (0) 0: 0   Location: Private network serving the local user (1)
   Ext: 1  Progress Description: Called
  equipment is non-ISDN. (2) ]
 
 
  -- Playing 'vm-goodbye' (language 'es')
 
  Asterisk*CLI
   Protocol Discriminator: Q.931 (8)  len=5
   Call Ref: len= 2 (reference 72/0x48) (Originator)
   Message type: CONNECT ACKNOWLEDGE (15)
  -- Executing NoOp(Zap/13-1, ) in new stack
  -- Executing Hangup(Zap/13-1, ) in new stack
== Spawn extension (primario, s, 4) exited non-zero on 'Zap/13-1'
 
  NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active
 
  Protocol Discriminator: Q.931 (8)  len=9
  Call Ref: len= 2 (reference 32840/0x8048) (Terminator)
  Message type: DISCONNECT (69)
  [08 02 81 90]
  Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
  Location: Private network serving the local user (1)
  Ext: 1  Cause: Normal Clearing (16), class = Normal
  Event (1) ]
 
 
  -- Hungup 'Zap/13-1'
 
  Asterisk*CLI
  On 8/9/05, jj [EMAIL PROTECTED] wrote:
 
  What does pri debug span 1 show?
 
  On Aug 9, 2005, at 5:02 PM, Panitaxx wrote:
 
 
  Hello,
 
  I have an ISDN PRI E1. For some reason I am not receiving