date:20051124


Hello

Whan starting astersik(1.2) (asterisk -vvc), I get this message :

 [res_config_mysql.so] => (MySQL RealTime Configuration Driver)
/libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so:
Undefined s
ymbol "ast_config_load"

What did I forgot to do?

Olivier

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RE: [Asterisk-Users] Looking for Windows based Asterisk

2005-11-24 Thread Guido Hecken

also add winscp and ultraedit to your windows system, it works great.
http://winscp.net/eng/index.php
http://www.ultraedit.com/

Regards 

Guido Hecken

> > Without putty, my windows would be meaningless.
> >
> > PaulH
> >
> Subtle Paul! but nice! :)
> Mike
> UK

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RE : RE : RE : [Asterisk-Users] What does it mean?

Merci

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 23:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: RE : RE : [Asterisk-Users] What does it mean?


Je ne donne pas de réponse !
Il me semble t'avoir suggèrer asterisk comme système
de messagerie vocale au lieu d'SEMS, avoir fourni
quelques fichiers de configuration, ce n'étaient pas
des devinettes.

Conbien de fois on ma répondu "personne n'est obligé
de faire ton tavail, tu n'as qu'a payé pour ce que tu
demandes.

IL me semble même me souvenir avoir lu un développeur
te faire la remarque "les utilisateurs de nos projets
vous ne profitez que de notre travail !".


Pour répondre à ton problème configure logger.conf .

Harry

  
--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:

> Cela veut simplement dire que tu te plains de ne pas
> avoir de réponses, mais
> qu'en fait tu n'en donnes pas non plus, sauf sous
> forme de devinette.
> Auquel cas, il est plus simple de ne pas répondre,
> 
> merci
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De
> la part de harry gaillac
> Envoyé : jeudi 24 novembre 2005 17:54
> À : Asterisk Users Mailing List - Non-Commercial
> Discussion
> Objet : RE: RE : [Asterisk-Users] What does it mean?
> 
> 
> Je ne connais pas la signification de "sybillines".
> Harry
> --- Olivier Taylor <[EMAIL PROTECTED]> a
> écrit
> :
> 
> > Tes réponses sont aussi sybillines que tes
> questions
> > :)
> > 
> > Olivier
> > 
> > -Message d'origine-
> > De : [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED]
> De
> > la part de harry gaillac
> > Envoyé : jeudi 24 novembre 2005 16:45
> > À : Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > Objet : RE: [Asterisk-Users] What does it mean?
> > 
> > 
> > Hello,
> > 
> > Read the Makefile in apps.
> > Harry
> > --- Olivier Taylor <[EMAIL PROTECTED]> a
> > écrit
> > :
> > 
> > > Hello,
> > > 
> > > I have compiled asterisk cvs under freebsd, no
> > > problems.
> > > 
> > > When starting asterisk, I get :
> > > 
> > > [res_config_mysql.so] => (MySQL RealTime
> > > Configuration Driver)
> > > /libexec/ld-elf.so.1:
> > > /usr/lib/asterisk/modules/res_config_mysql.so:
> > > Undefined symbol "ast_config_load"
> > > 
> > > What's wrong?
> > > 
> > > Olivier
> > > 
> > > ___
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> > > 
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> > >
> >
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> > >   
> > >
> >
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> > > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> >
>
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> > 
> > Appel audio GRATUIT partout dans le monde avec le
> > nouveau Yahoo! Messenger
> > Téléchargez cette version sur
> > http://fr.messenger.yahoo.com 
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Re: [Asterisk-Users] Fax sending problems

2005-11-24 Thread Lee Howard


Lee Archer wrote:


Nov 24 10:50:15.02: [ 8222]: <-- data [1031]
Nov 24 10:51:15.01: [ 8222]: MODEM TIMEOUT: writing to modem
Nov 24 10:51:15.01: [ 8222]: MODEM WRITE SHORT: sent 1031, wrote 984
Nov 24 10:51:15.01: [ 8222]: SEND end page



What's going on with the iaxmodem output (on stdout/stderr) at this 
point in time?


Lee.
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Re: [Asterisk-Users] HELP! on disconnecting stale calls.

2005-11-24 Thread Paradise Dove

as i said before, i've ran "soft hangup" on both sip and zap channels
on this call several times but no success.
by exploring the code in chan_sip.c it shows that * also attempts to
run softhangup on this call.
is this probably be a bug?

thanks,
paradise dove

On 11/25/05, tracinet <[EMAIL PROTECTED]> wrote:
> Have you tried the "soft hangup" command?
>
>
> On 11/24/05, Paradise Dove <[EMAIL PROTECTED]> wrote:
> >
> > hi,
> > how can i hangup such calls without restarting asterisk?
> > the Zap channel on this case is busy for more than 7 hours
> > some logs are followed.
> >
> > thanks,
> > Paradise Dove
> > -
> > Nov 23 16:59:49 NOTICE[3752] chan_sip.c: Disconnecting call
> > 'SIP/2378-740f' for lack of RTP activity in 25788 seconds
> > Nov 23 16:59:49 NOTICE[3752] chan_sip.c: Disconnecting call
> > 'SIP/2378-740f' for lack of RTP activity in 25788 seconds
> > Nov 23 16:59:50 NOTICE[3752] chan_sip.c: Disconnecting call
> > 'SIP/2378-740f' for lack of RTP activity in 25789 seconds
> > Nov 23 16:59:51 NOTICE[3752] chan_sip.c: Disconnecting call
> > 'SIP/2378-740f' for lack of RTP activity in 25790 seconds
> > Nov 23 16:59:51 NOTICE[3752] chan_sip.c: Disconnecting call
> > 'SIP/2378-740f' for lack of RTP activity in 25790 seconds
> > Nov 23 16:59:52 NOTICE[3752] chan_sip.c: Disconnecting call
> > 'SIP/2378-740f' for lack of RTP activity in 25791 seconds
> > Nov 23 16:59:52 NOTICE[3752] chan_sip.c: Disconnecting call
> > 'SIP/2378-740f' for lack of RTP activity in 25791 seconds
> > Nov 23 16:59:53 NOTICE[3752] chan_sip.c: Disconnecting call
> > 'SIP/2378-740f' for lack of RTP activity in 25792 seconds
> > Nov 23 16:59:53 NOTICE[3752] chan_sip.c: Disconnecting call
> > 'SIP/2378-740f' for lack of RTP activity in 25792 seconds
> > -
> > Channel  Location State
> Application(Data)
> > Zap/15-1 [EMAIL PROTECTED]:1 Up  Bridged
> Call(SIP/2378-740f)
> > 1 active channel
> > 1 active call
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Re: [Asterisk-Users] Asterisk and Japanese Caller ID

2005-11-24 Thread isamar



Actually, exactly now I am trying to do that also...

Isamar


On Fri, 25 Nov 2005, Aaron Anderson wrote:


Are there any kind of patches or experimental libraries that I can use
to pull caller ID info off a japanese pots line?



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Re: [Asterisk-Users] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...

2005-11-24 Thread Francesco Peeters

On Fri, November 25, 2005 3:27, Tzafrir Cohen said:
> On Thu, Nov 24, 2005 at 10:16:40PM +0100, Francesco Peeters wrote:

>> > Seems to me there's an issue in that area: chan_zap, maybe libpri,
>> etc.
>
> So what do you have in zapata.conf?
>

I posted that a few posts back in this thread... No need to repeat it,
especially because of below developments!

>> >
>>
>> I keep replying to myself...  ;-)
>>

Yes, I really do!

I compiled 1.2 and bristuff 0.3.0 Pre1 yesterday late and that now seems
to work! * is up and running *with* 2nd card in NT mode...

I'll test with connecting a phone to it tonight to see whether that'll
work too, but at least * now continues to load correctly *and* both D
channels come up!

Progress at last!

I'll keep ya'll updated!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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Re: [Asterisk-Users] asterisk.conf question

2005-11-24 Thread Leif Neland


Adrian A wrote:


Does anyone know what exactly the option
transmit_silence_during_record in asterisk.conf does? Is this useful
for voicemail recording?


Could the option be named any more explicitly? It does _exactly_ what
it says it does.


Some providers terminate the connection if nothing is transmitted for x 
seconds.
If asterisk sends nothing while the caller speaks his message, the provider 
might terminate the call.

So asterisk can transmit silence (which is not "nothing") during record.

Similarly you might have to say "yes dear" regularly to avoid having the 
connection terminated while talking to your SO. :-)


Leif

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[Asterisk-Users] NewBie to Ast Server, help need for the configuration

2005-11-24 Thread ram

Hi all
 
iam setting PBX for outgoing calls at this moment
once iam success this , iam planning to do config inbound to
 
So iam start configuring with Outbound calls
 
Ring now my config looks like follow
 
Lan Users-- Astrix--- VoIP provider
 
I have one account with VoIP provider, i can make multiple calls using that accounts
 
i have 20 Lan users, who start making called to out going
 
all of the them connected to Lan Swtich where astrix connected
 
 
I have downloaded Asterisk+addons+sounds
and comipled with any errors
 
now iam looking what are the files need to configured to achieve the following setup.
 
here my question about the config
 
1. where should i config this Account of VoIP to register, so i can make calls out
2. how do i create 20 users and register them and start making calls
3. where can i see which user called where, and duration
4. how do i configure 20 users can talk each other using extensions.
5. the user side can be Soft Phone using PC or Any cisco ATA Box.
 
what are the config files i need to look
 
any suggestions will be appriciated.
 
ram
 
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[Asterisk-Users] Asterisk and Japanese Caller ID

2005-11-24 Thread Aaron Anderson


Are there any kind of patches or experimental libraries that I can use
to pull caller ID info off a japanese pots line?



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[Asterisk-Users] Recommended PCI latency time?

2005-11-24 Thread Boris Bakchiev

Hi,

What would be a recommended PCI latency timing for server running TE406P
card?

Thanks
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Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

2005-11-24 Thread Tom Rymes


On Nov 24, 2005, at 12:14 PM, Bharath wrote:

I found out that I have a faulty Belkin Router which was causing  
the problem. I tried forwarding ports as well as DMZ'd the Sip  
device but still could'nt not hear the voice. So i plugged the sip  
device directly to the cable modem & it worked fine. So my guess is  
that though I have set up the router to forwards port to the sip  
device there is something happening at the router that is blocking  
the RTP ports (1-2).

Thanks


Before you blame the router, make sure that you forwarded UDP ports  
5060 and 1-2, not TCP. (Though I guess the DMZ setup would  
have taken care of that...)


Tom


Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

"Intelligent technology solutions for small businesses."


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Re: [Asterisk-Users] HELP! on disconnecting stale calls.

2005-11-24 Thread tracinet

Have you tried the "soft hangup" command?On 11/24/05, Paradise Dove <[EMAIL PROTECTED]> wrote:
hi,how can i hangup such calls without restarting asterisk?the Zap channel on this case is busy for more than 7 hours
some logs are followed.thanks,Paradise Dove-Nov 23 16:59:49 NOTICE[3752] chan_sip.c: Disconnecting call'SIP/2378-740f' for lack of RTP activity in 25788 secondsNov 23 16:59:49 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25788 secondsNov 23 16:59:50 NOTICE[3752] chan_sip.c: Disconnecting call'SIP/2378-740f' for lack of RTP activity in 25789 secondsNov 23 16:59:51 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25790 secondsNov 23 16:59:51 NOTICE[3752] chan_sip.c: Disconnecting call'SIP/2378-740f' for lack of RTP activity in 25790 secondsNov 23 16:59:52 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25791 secondsNov 23 16:59:52 NOTICE[3752] chan_sip.c: Disconnecting call'SIP/2378-740f' for lack of RTP activity in 25791 secondsNov 23 16:59:53 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25792 secondsNov 23 16:59:53 NOTICE[3752] chan_sip.c: Disconnecting call'SIP/2378-740f' for lack of RTP activity in 25792 seconds-Channel  Location
State   Application(Data)Zap/15-1
[EMAIL PROTECTED]:1
Up  Bridged Call(SIP/2378-740f)1 active channel1 active call___--Bandwidth and Colocation sponsored by Easynews.com --
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[Asterisk-Users] Asterisk + SER problem,ua cann't hangup

2005-11-24 Thread WXFList

hi everybody:
   I use Asterisk and SER(with nathelp moudle) in on box, SER as sip
registrar and sip proxy, Asterisk as media gw and pstn connector. Here
is my configuration: SER use 192.168.2.10:5060,Asterisk use
192.168.2.10:5065,my pstn gw is 192.168.2.20:5060 
  in ser.cfg
if (method=="INVITE") {
if (!(uri=~"^sip:[EMAIL PROTECTED]")) {
   rewritehostport("192.168.2.10:5065");
   forward(uri:host, uri:port);
   break;
};
};
  in extensions.conf
[Out]
exten => _9XXX,1,Dial(SIP/[EMAIL PROTECTED],,rT)
exten => _9XXX,n,hangup
I use xlite dial 9001 and the 9001 phone ring,then i pick up the 9001 phone,
all seem good.but when i drop the 9001 phone,xlite doesn't disconnect.xlite
still show connected status till i manual huangup it.it seem the BYE message
cann't be sended to the xlite,what can i do? 

Regards.

here is some asterisk sip debug info:

set_destination: Parsing  for 
address/port to send to
set_destination: set destination to 192.168.2.10, port 5060
Reliably Transmitting (NAT) to 192.168.2.10:5060:
BYE sip:[EMAIL PROTECTED]:9753 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5065;branch=z9hG4bK37f9a0d6;rport
Route: 
From: ;tag=as1f67b017
To: 2002;tag=4b08853d
Contact: 
Call-ID: 3d4b5923d31edb66
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
 
 
---
Retransmitting #1 (NAT) to 192.168.2.10:5060:
BYE sip:[EMAIL PROTECTED]:9753 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5065;branch=z9hG4bK37f9a0d6;rport
Route: 
From: ;tag=as1f67b017
To: 2002;tag=4b08853d
Contact: 
Call-ID: 3d4b5923d31edb66
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
 

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Re: [Asterisk-Users] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...

2005-11-24 Thread Tzafrir Cohen

On Thu, Nov 24, 2005 at 10:16:40PM +0100, Francesco Peeters wrote:
> On Thu, November 24, 2005 20:09, Francesco Peeters said:
> > On Wed, November 23, 2005 20:29, Francesco Peeters said:
> 
> >
> > Just made myself a crossed NT1 connection to the NT mode card (as
> > described on the PBX4linux site) and connected my phone.
> >
> > The zaphfc driver shows that layer 1 is activated (G3) once the phone is
> > connected, but that is where it stops, as anything above that should be
> > handled in chan_zap.
> >
> > However when I leave the card in bri_cpe_ptmp in zapata.conf, the layer2+
> > protocols are not correct (TE mode) and when I put in bri_net_ptmp, the
> > chan_zap somehow doesn't complete loading or exits in an unexpected
> > manner, resulting in a situation where Asterisk stops loading it's
> > configs, and thus runs without a dialplan and other modules...
> >
> > Seems to me there's an issue in that area: chan_zap, maybe libpri, etc.

So what do you have in zapata.conf?

> >
> 
> I keep replying to myself...  ;-)
> 
> Some extra info:
> /proc/zaptel/*:
> Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)" AMI/CCS
> 
>1 ZTHFC1/0/1 Clear (In use)
>2 ZTHFC1/0/2 Clear (In use)
>3 ZTHFC1/0/3 HDLCFCS (In use)
> Span 2: ZTHFC2 "HFC-S PCI A ISDN card 1 [NT] layer 1 ACTIVATED (G3)" AMI/CCS
> 
>4 ZTHFC2/0/1 Clear (In use)
>5 ZTHFC2/0/2 Clear (In use)
>6 ZTHFC2/0/3 HDLCFCS (In use)
> 
> /proc/pci:
>   Bus  0, device   7, function  2:
> USB Controller: Intel Corp. 82371AB/EB/MB PIIX4 USB (rev 1).
>   IRQ 11.
>   Master Capable.  Latency=32.
>   I/O at 0xc000 [0xc01f].
>   Bus  0, device   9, function  0:
> Network controller: Cologne Chip Designs GmbH ISDN network controller
> [HFC-P
> CI] (rev 2).
>   IRQ 11.
>   Master Capable.  Latency=16.  Max Lat=16.
>   I/O at 0xc400 [0xc407].
>   Non-prefetchable 32 bit memory at 0xe3001000 [0xe30010ff].
>   Bus  0, device  17, function  0:
> Network controller: Cologne Chip Designs GmbH ISDN network controller
> [HFC-P
> CI] (#2) (rev 2).
>   IRQ 11.
>   Master Capable.  Latency=16.  Max Lat=16.
>   I/O at 0xcc00 [0xcc07].
>   Non-prefetchable 32 bit memory at 0xe3002000 [0xe30020ff].
> 
> I have tried all I can to assign different IRQ's to the HFC-PCI cards, but
> they *always* take IRQ 11  :-(
> 
> -- 
> Francesco Peeters
> 
> GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
> If your program doesn't recognize my signature, please visit
> http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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-- 
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http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
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Re: [Asterisk-Users] asterisk.conf question

2005-11-24 Thread Kevin P. Fleming


Adrian A wrote:


Does anyone know what exactly the option transmit_silence_during_record in
asterisk.conf does? Is this useful for voicemail recording?


Could the option be named any more explicitly? It does _exactly_ what it 
says it does.

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RE: [Asterisk-Users] harry's project

2005-11-24 Thread Jonathan k. Creasy

http://www.automated.it/guidetoasterisk.htm

I don't think you even require SER in that case. 

That will be $100. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac
Sent: Thursday, November 24, 2005 7:11 PM
To: users@openser.org; asterisk-users@lists.digium.com
Subject: [Asterisk-Users] harry's project

Hello,

here is an other  diagram for people who don't yet
understand what i expect to do.

Look at sip_call_flow.png file i wish to substitute
ondo sip server with ser and ondo pbx with asterisk .

ondo sip server is able to do far-end near-end nat I
guess ser too.

I do hope i will find some people who help me to
configure that .

Regards 
Harry 






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Re: [Asterisk-Users] Preventing long-distance call forwarding

Well, I kinda answered this myself, and I'll post how I did it in case 
a) it might cause other problems and 2) anyone else finds it useful.


To recap, a Polycom phone will let you enter anything as a call 
diversion number - this is obviously a problem in that someone can 
forward their phone to Australia, go home and phone their extension to 
make the call to Australia for free. There doesn't seem to be anyway to 
configure the phone to restrict call diversion numbers.


Here's what happens:

-- Called 2471
-- Got SIP response 302 "Moved Temporarily" back from 172.16.16.100
-- Now forwarding SIP/2293-2461 to 'Local/[EMAIL PROTECTED]' 
(thanks to SIP/2471-d428)


The context is set from the called phone which, in the ordinary course 
of events, is allowed to make long-distance calls. For diverted calls, 
what I have done is this:


exten => s,1,GotoIf($[$["${CHANNEL:0:5}" = "Local"] & 
$[$["${MACRO_CONTEXT}" = "long-distance"] | $["${MACRO_CONTEXT}" = 
"international"]]]?:3)

exten => s,2,Goto(internal,${ARG2},1)
exten => s,3,.. rest of dialplan

This basically moves the call into a context from which only local 
calls are allowed, based on the channel starting with 'Local/'.


Does anyone see any glaring problems with this?

On Nov 24, 2005, at 4:04 PM, Anthony Rodgers wrote:


Hi there,

We have PolyCom IP501s in a context that allows long-distance dialing,
but we want to prevent those same phones from being forwarded to
long-distance numbers using the softkey on the phone (without disabling
the key itself).

Does anyone have any PolyCom/dialplan tricks to accomplish this? Even a
way to reliably detect a call-forward in the dialplan would help

TIA.

Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp

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Re: [Asterisk-Users] mISDN and chan_isdn for 1.2

2005-11-24 Thread Vidar


This is how I just did it (finally):

### First grab the mqueue branch of mISDN to the folder which is hard-coded 
in the chan_misdn Makefile

mkdir /usr/src/mqueue
cd /usr/src/mqueue
cvs -d :pserver:[EMAIL PROTECTED]:/i4ldev login
(password: readonly)

cvs -d :pserver:[EMAIL PROTECTED]:/i4ldev co -r mqueue mISDN
cvs -d :pserver:[EMAIL PROTECTED]:/i4ldev co -r mqueue mISDNuser


### Patch kernel with mISDN
cd /usr/src/mqueue/mISDN
./std2kern


### (Recompile and install kernel with the usual capi/mISDN modules 
reboot)  ###



### Make sure you update your linux with the header files of mISDN (not 
doing this has cause me lots of head-ache and wasted hours)


cp /usr/src/mqueue/mISDN/include/linux/*.h /usr/include/linux


### Make mISDNuser

cd /usr/src/mqueue/mISDNuser
make


### Download and install chan_misdn

cd ..
wget http://www.beronet.com/downloads/chan_misdn/unstable/chan_misdn.tar.gz
tar zxf chan_misdn.tar.gz
cd chan_misdn
make
make install
/etc/init.d/misdn-init config
/etc/init.d/misdn-init start


At last I configured mISDN using instructions available in README and in 
various web sites ###

That includes...
* changing my card to NT mode in /etc/misdn-init.conf
* modifying /etc/asterisk/misdn.conf
* modifying /etc/asterisk/extensions.conf
* make sure the module loads in asterisk /etc/asterisk/modules.conf
* Start asterisk

Regards,
Vidar

- Original Message - 
From: John Martin

To: asterisk-users@lists.digium.com
Sent: Friday, November 18, 2005 8:04 PM
Subject: [Asterisk-Users] mISDN and chan_isdn for 1.2


Hi All,
 Can anyone recommend a version of mISDN and mISDNuser (dates of CVS or 
archive held on someone's server) that will work with the chan_isdn in 
Asterisk 1.2.


Many thanks.

John
www.AuPix.com




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[Asterisk-Users] Pros and Cons of T1/E1 cards

2005-11-24 Thread John Daragon


Hi;

We're looking to standardise on a single family of E1 PRI cards.

I guess our options are :

Digium / Zaptel / libpri
Sangoma/ Zaptel / Wanpipe
AVM/ CAPI
eIcon  / CAPI
Junghanns  / Bristuff

Can anyone share any comparative experience of these, please ? Do they 
differ much in terms of interrupt requirement, CPU load &c ?


Any info gratefully received.

jd

--

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argv[0] limited
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v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127


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Re: [Asterisk-Users] Asterisk 1.2.0 Broken staged install

2005-11-24 Thread Alfie Viechweg


Found auth problem!

Installing asterisk 1.2.0 with INSTALL_PREFIX set will copy this 
variables into your config file - asterisk.conf and result int things 
like failed sip user information etc. If you do you own install (LFS :) 
poeple) beware! Try using DESTDIR instead. The docs and Makefile is not 
too clear on the difference between these two.

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[Asterisk-Users] harry's project

Hello,

here is an other  diagram for people who don't yet
understand what i expect to do.

Look at sip_call_flow.png file i wish to substitute
ondo sip server with ser and ondo pbx with asterisk .

ondo sip server is able to do far-end near-end nat I
guess ser too.

I do hope i will find some people who help me to
configure that .

Regards 
Harry 






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Re: [Asterisk-Users] chan_misdn crashes : init_stack: success but entitylist not empty

2005-11-24 Thread Yoann Le Bihan

Jose,

I met so many problems these last 8 days that I don't remember exactly
which config was mine at that time, so I can't testify the answer...
(just for fun : my linux box is having 3 hd with a different distro on
each of them and I plug the cable on the hd I want to boot depending
on my mood ;o)).

I think I was running 1.0.9. The main things I did were :

  - deinstalling everything (asterisk, misdn, misdnuser, chan_misdn, ...)
  - compiling and installing asterisk 1.2.0 (make ; make install)
  - downloading the install_misdn script on beronet
(http://www.beronet.com/download/install-misdn.tar.gz) and executing
the make install (be careful : you need kernel headers)

And now, I'm done : Asterisk runs without chan_misdn, but crashes with
it :-( But it's installed :-)

Good luck ! ;)

Cheers,

YLB.


2005/11/25, Jose Limeres <[EMAIL PROTECTED]>:
> Yoann,
> I am going through a similar problem you reported in a past posting:
>
> Nov 24 17:49:31 ERROR[9326] chan_misdn.c: Unable to initialize mISDN
> Nov 24 17:49:31 WARNING[9326] loader.c: chan_misdn.so: load_module
> failed, returning -1
> Nov 24 17:49:31 WARNING[9326] chan_misdn.c: cb_log called with
> out-of-range port number! (0)
> Nov 24 17:49:31 WARNING[9326] loader.c: Loading module chan_misdn.so failed!
>
> How did you solve it?
> Thanks,  jose
>
> On 25/11/05, Yoann Le Bihan <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > Asterisk 1.2 on FC4, all is right, I'm happy. But when I try to load
> > chan_misdn after a successful install, I get it :
> >
> > # asterisk -vvvgc
> > [...]
> >  [chan_features.so] => (Feature Proxy Channel)
> >   == Registered channel type 'Feature' (Feature Proxy Channel Driver)
> >  [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri))
> >   == Parsing '/etc/asterisk/misdn.conf': Found
> > Got: 1 from get_ports
> > Init. Stack on port:1
> > No Connect port:1
> > init_stack: Success
> > #
> >
> > Nothing else. Asterisk crashes. If I look at /var/log/messages :
> >
> > # tail /var/log/messages
> > Nov 25 00:22:39 toto kernel: Debug: sleeping function called from
> > invalid context at arch/i386/lib/usercopy.c:634
> > Nov 25 00:22:39 toto kernel: in_atomic():0, irqs_disabled():1
> > Nov 25 00:22:39 toto kernel:  [] copy_from_user+0x18/0x80
> > Nov 25 00:22:39 toto kernel:  [] mISDN_write+0x318/0x7c5 
> > [mISDN_core]
> > Nov 25 00:22:39 toto kernel:  [] mISDN_write+0x0/0x7c5 
> > [mISDN_core]
> > Nov 25 00:22:39 toto kernel:  [] vfs_write+0xa2/0x15a
> > Nov 25 00:22:39 toto kernel:  [] sys_write+0x41/0x6a
> > Nov 25 00:22:39 toto kernel:  [] syscall_call+0x7/0xb
> > Nov 25 00:22:39 toto kernel: MISDN free_device: entitylist not empty
> > #
> >
> > Any idea ?... I've been on it for 1 whole week... I'm exhausted :-(
> >
> > Cheers,
> >
> > YLB.
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[Asterisk-Users] Preventing long-distance call forwarding


Hi there,

We have PolyCom IP501s in a context that allows long-distance dialing, 
but we want to prevent those same phones from being forwarded to 
long-distance numbers using the softkey on the phone (without disabling 
the key itself).


Does anyone have any PolyCom/dialplan tricks to accomplish this? Even a 
way to reliably detect a call-forward in the dialplan would help


TIA.

Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp

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Re: [Asterisk-Users] Grandstream problem

2005-11-24 Thread Alfie Viechweg


Michel Belleau (malaiwah.com) wrote:


Hi Alfie.

Did you try setting up a "username=100" in your [100] context and a
"username=101" in your [101] context?
That should do the trick..

Michel Belleau
SERVICES INFORMATIQUES MALAIWAH.COM
(418) 261-6412 -- http://www.malaiwah.com



Alfie Viechweg a écrit :

 


Can some on help me find the problem here please:
I'm using asterisk 1.2.0 with Grandstream GXP-2000

This is the debugging output from asterisk:

<-- SIP read from 10.0.3.21:5060:
REGISTER sip:10.0.3.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.21;branch=z9hG4bK5c77f205e9f991de
From: ;tag=aea38200ad3c1539
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 10001 REGISTER
Expires: 3600
User-Agent: Grandstream GXP2000 1.0.1.9
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0


--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 10.0.3.21 : 5060 (non-NAT)
Transmitting (no NAT) to 10.0.3.21:5060:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP
10.0.3.21;branch=z9hG4bK5c77f205e9f991de;received=10.0.3.21
From: ;tag=aea38200ad3c1539
To: ;tag=as248942d8
Call-ID: [EMAIL PROTECTED]
CSeq: 10001 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: 
Content-Length: 0


---
Nov 24 19:23:43 NOTICE[9700]: chan_sip.c:10815
handle_request_register: Registration from '' failed
for '10.0.3.21' - Username/auth name mismatch
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Destroying call '[EMAIL PROTECTED]'

* This is the relevant parts of my sip.conf:

[100]
type=friend
secret=test
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal

[101]
type=friend
secret=test
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal

 This is the relevant part of my extensions.conf:

[internal]
exten => 100,1,Dial(SIP/100)
exten => 101,1,Dial(SIP/101)
exten => 611,1,Echo()



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I tried adding username=xxx and that did not solve the problem.

What is the 'sip show users' command (using CLI) suppose to show in a 
properly configured server?

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Re: [Asterisk-Users] chan_misdn crashes : init_stack: success but entitylist not empty

2005-11-24 Thread Jose Limeres

Yoann,
I am going through a similar problem you reported in a past posting:

Nov 24 17:49:31 ERROR[9326] chan_misdn.c: Unable to initialize mISDN
Nov 24 17:49:31 WARNING[9326] loader.c: chan_misdn.so: load_module
failed, returning -1
Nov 24 17:49:31 WARNING[9326] chan_misdn.c: cb_log called with
out-of-range port number! (0)
Nov 24 17:49:31 WARNING[9326] loader.c: Loading module chan_misdn.so failed!

How did you solve it?
Thanks,  jose

On 25/11/05, Yoann Le Bihan <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Asterisk 1.2 on FC4, all is right, I'm happy. But when I try to load
> chan_misdn after a successful install, I get it :
>
> # asterisk -vvvgc
> [...]
>  [chan_features.so] => (Feature Proxy Channel)
>   == Registered channel type 'Feature' (Feature Proxy Channel Driver)
>  [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri))
>   == Parsing '/etc/asterisk/misdn.conf': Found
> Got: 1 from get_ports
> Init. Stack on port:1
> No Connect port:1
> init_stack: Success
> #
>
> Nothing else. Asterisk crashes. If I look at /var/log/messages :
>
> # tail /var/log/messages
> Nov 25 00:22:39 toto kernel: Debug: sleeping function called from
> invalid context at arch/i386/lib/usercopy.c:634
> Nov 25 00:22:39 toto kernel: in_atomic():0, irqs_disabled():1
> Nov 25 00:22:39 toto kernel:  [] copy_from_user+0x18/0x80
> Nov 25 00:22:39 toto kernel:  [] mISDN_write+0x318/0x7c5 
> [mISDN_core]
> Nov 25 00:22:39 toto kernel:  [] mISDN_write+0x0/0x7c5 [mISDN_core]
> Nov 25 00:22:39 toto kernel:  [] vfs_write+0xa2/0x15a
> Nov 25 00:22:39 toto kernel:  [] sys_write+0x41/0x6a
> Nov 25 00:22:39 toto kernel:  [] syscall_call+0x7/0xb
> Nov 25 00:22:39 toto kernel: MISDN free_device: entitylist not empty
> #
>
> Any idea ?... I've been on it for 1 whole week... I'm exhausted :-(
>
> Cheers,
>
> YLB.
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Re: [Asterisk-Users] Grandstream problem

2005-11-24 Thread Michel Belleau (malaiwah.com)

Hi Alfie.

Did you try setting up a "username=100" in your [100] context and a
"username=101" in your [101] context?
That should do the trick..

Michel Belleau
SERVICES INFORMATIQUES MALAIWAH.COM
(418) 261-6412 -- http://www.malaiwah.com



Alfie Viechweg a écrit :

> Can some on help me find the problem here please:
> I'm using asterisk 1.2.0 with Grandstream GXP-2000
>
> This is the debugging output from asterisk:
>
> <-- SIP read from 10.0.3.21:5060:
> REGISTER sip:10.0.3.1 SIP/2.0
> Via: SIP/2.0/UDP 10.0.3.21;branch=z9hG4bK5c77f205e9f991de
> From: ;tag=aea38200ad3c1539
> To: 
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 10001 REGISTER
> Expires: 3600
> User-Agent: Grandstream GXP2000 1.0.1.9
> Max-Forwards: 70
> Allow:
> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
> Content-Length: 0
>
>
> --- (12 headers 0 lines)---
> Using latest REGISTER request as basis request
> Sending to 10.0.3.21 : 5060 (non-NAT)
> Transmitting (no NAT) to 10.0.3.21:5060:
> SIP/2.0 404 Not found
> Via: SIP/2.0/UDP
> 10.0.3.21;branch=z9hG4bK5c77f205e9f991de;received=10.0.3.21
> From: ;tag=aea38200ad3c1539
> To: ;tag=as248942d8
> Call-ID: [EMAIL PROTECTED]
> CSeq: 10001 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Max-Forwards: 70
> Contact: 
> Content-Length: 0
>
>
> ---
> Nov 24 19:23:43 NOTICE[9700]: chan_sip.c:10815
> handle_request_register: Registration from '' failed
> for '10.0.3.21' - Username/auth name mismatch
> Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
> Destroying call '[EMAIL PROTECTED]'
>
> * This is the relevant parts of my sip.conf:
>
> [100]
> type=friend
> secret=test
> qualify=yes
> nat=no
> host=dynamic
> canreinvite=no
> context=internal
>
> [101]
> type=friend
> secret=test
> qualify=yes
> nat=no
> host=dynamic
> canreinvite=no
> context=internal
>
>  This is the relevant part of my extensions.conf:
>
> [internal]
> exten => 100,1,Dial(SIP/100)
> exten => 101,1,Dial(SIP/101)
> exten => 611,1,Echo()
>
>
>
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begin:vcard
fn:Michel Belleau (malaiwah.com)
n:Belleau;Michel
org:MALAIWAH.COM - Services Informatiques
adr;quoted-printable:;;6374, avenue Royale;L'Ange-Gardien;Qu=C3=A9bec;G0A 2K0;Canada
email;internet:[EMAIL PROTECTED]
tel;work:(418) 261-6412
x-mozilla-html:TRUE
url:http://www.malaiwah.com/
version:2.1
end:vcard

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[Asterisk-Users] Grandstream problem

2005-11-24 Thread Alfie Viechweg


Can some on help me find the problem here please:
I'm using asterisk 1.2.0 with Grandstream GXP-2000

This is the debugging output from asterisk:

<-- SIP read from 10.0.3.21:5060:
REGISTER sip:10.0.3.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.21;branch=z9hG4bK5c77f205e9f991de
From: ;tag=aea38200ad3c1539
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 10001 REGISTER
Expires: 3600
User-Agent: Grandstream GXP2000 1.0.1.9
Max-Forwards: 70
Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK

Content-Length: 0


--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 10.0.3.21 : 5060 (non-NAT)
Transmitting (no NAT) to 10.0.3.21:5060:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 10.0.3.21;branch=z9hG4bK5c77f205e9f991de;received=10.0.3.21
From: ;tag=aea38200ad3c1539
To: ;tag=as248942d8
Call-ID: [EMAIL PROTECTED]
CSeq: 10001 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: 
Content-Length: 0


---
Nov 24 19:23:43 NOTICE[9700]: chan_sip.c:10815 handle_request_register: 
Registration from '' failed for '10.0.3.21' - 
Username/auth name mismatch

Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Destroying call '[EMAIL PROTECTED]'

* This is the relevant parts of my sip.conf:

[100]
type=friend
secret=test
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal

[101]
type=friend
secret=test
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal

 This is the relevant part of my extensions.conf:

[internal]
exten => 100,1,Dial(SIP/100)
exten => 101,1,Dial(SIP/101)
exten => 611,1,Echo()



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[Asterisk-Users] chan_misdn crashes : init_stack: success but entitylist not empty

2005-11-24 Thread Yoann Le Bihan

Hi,

Asterisk 1.2 on FC4, all is right, I'm happy. But when I try to load
chan_misdn after a successful install, I get it :

# asterisk -vvvgc
[...]
 [chan_features.so] => (Feature Proxy Channel)
  == Registered channel type 'Feature' (Feature Proxy Channel Driver)
 [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri))
  == Parsing '/etc/asterisk/misdn.conf': Found
Got: 1 from get_ports
Init. Stack on port:1
No Connect port:1
init_stack: Success
#

Nothing else. Asterisk crashes. If I look at /var/log/messages :

# tail /var/log/messages
Nov 25 00:22:39 toto kernel: Debug: sleeping function called from
invalid context at arch/i386/lib/usercopy.c:634
Nov 25 00:22:39 toto kernel: in_atomic():0, irqs_disabled():1
Nov 25 00:22:39 toto kernel:  [] copy_from_user+0x18/0x80
Nov 25 00:22:39 toto kernel:  [] mISDN_write+0x318/0x7c5 [mISDN_core]
Nov 25 00:22:39 toto kernel:  [] mISDN_write+0x0/0x7c5 [mISDN_core]
Nov 25 00:22:39 toto kernel:  [] vfs_write+0xa2/0x15a
Nov 25 00:22:39 toto kernel:  [] sys_write+0x41/0x6a
Nov 25 00:22:39 toto kernel:  [] syscall_call+0x7/0xb
Nov 25 00:22:39 toto kernel: MISDN free_device: entitylist not empty
#

Any idea ?... I've been on it for 1 whole week... I'm exhausted :-(

Cheers,

YLB.
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Re: [Asterisk-Users] [Fwd: call status with FXO]

2005-11-24 Thread Gabriel Rojas

Adam Goryachev wrote:
> see the zapata.conf for callprogress=yes
> However, this is unreliable, and could provide incorrect results. For
> accurate information you will need to get a BRI or PRI and related
> interface card. These provide the information Out Of Band, and as such
> are accurate.

I've already tried it and the results were awful. Every call was
terminated in the middle of it. My big question is how a modem can detect
such states with 100% accuracy and asterisk can't. Is there a basic
problem or difference that cannot be solved?

> callerid=no will solve that, but then you won't get callerid :)
> Possibly, your line doesn't supply callerid anyway, or you don't care,
> so then that is a good solution.

I wasn't clear enough. It seems that if you something like

  exten => _99.,n,Dial(Zap/4/${EXTEN:2})

Asterisk considers that Zap/4 ( an FXO device ) is answering Dial() first
and then dialling the outside number. That first answering is what I mean.
I realize now that there is CID transfer between the calling extension and
Zap/4 but why does it takes so long? And yes, I do need CID. As I said,
asterisk 1.0.7 ( I think
it was that version I tried first ) with Linux 2.4.2x didn't have such
behavior: dialling the FXO iface would get an immedate dialtone or the
outside number dialed right away.
   My greatest surprise is there seems to be a very small amount of people
connecting telco lines to internal extensions via Asterisk. The whole
point of trying to get accurate info about call state is to be able to
do thing like outside call accounting and billing distribution on
simple PSTN lines via common Digium TDM400 cards. Any work on this or
trick to do the job? Thanks in advance

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[Asterisk-Users] Re: jittering with Iax2 and Meetme on Asterisk 1.2.0

2005-11-24 Thread Tony Mountifield

In article <[EMAIL PROTECTED]>,
Steven Langley <[EMAIL PROTECTED]> wrote:
> 
> I have been using Asterisk 1.0.9 fairly successfully. I have Iax2 softphones
> based on the IaxClient library that are dialing into Meetme conferences. I
> am using a Zaptel card as a timing source.
> 
> I am now trying to migrate to Asterisk 1.2.0, mainly because of the alleged
> improved jitterbuffer implementation. I have installed 1.2.0 (Zaptel and
> Asterisk) and am running it on a 100 mbit LAN. I am dialing in with the same
> softphone (as the other server with Asterisk 1.0.9), but experience
> consistently bad jitter, both when jitterbuffer=no and when
> jitterbuffer=yes. I have run zttest and am getting pretty much 100% accuracy
> from the card.
> 
> Does anyone have any ideas what the problem could be?

Have a look at http://bugs.digium.com/view.php?id=5697 for some ideas...

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk

2005-11-24 Thread Alex Ternero

Title: Linksys SPA-841 Disconnects from Asterisk








I don t have problems, after upgrade the
firmware to the latest version.

 

Alex

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow
Sent: Thursday, November 24, 2005
3:49 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Linksys
SPA-841 Disconnects from Asterisk



 

Hi
all, I wonder if anyone out there has experienced an issue I am having with my
Sipura / Linksys SPA-841 phones. 

They
work fine generally, but occasionally, incoming calls are missed.  It's
like the SIP registration is expiring.  Does anyone know how to force the
phone to re-register automatically?  

 

David
A. Morrow 
Technical
Systems Lead 
Autodata
Solutions Company 
[EMAIL PROTECTED]

http://www.autodata.net


* PLEASE NOTE THAT EFFECTIVE DEC 1,2005
MY TELEPHONE NUMBER WILL CHANGE * 

NEW
!!! Tel: (519) 963-3020 
Fax:
(519) 451-6615 


< Poor planning on your part does not necessarily
constitute an emergency on my part! > 

This
message has originated from Autodata Solutions. The attached material is the
Confidential and Proprietary Information of Autodata Solutions. This email and
any files transmitted with it are confidential and intended solely for the use
of the individual or entity to whom they are addressed. If you have received
this email in error please delete this message and notify the Autodata system
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Re: [Asterisk-Users] PRI problems again - What should I do ?

2005-11-24 Thread Krishna Sumanth Chava

Hi Julian,
 
I think the Dell poweredge2850 servers are not too compatible with the zaptel cards..
 
Thanks
krishna 
On 11/24/05, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
I know that's a real newbie question, but I have a problem.I keep getting frame rejects, and a D-channel bouncing up and down. BT
say that it is at my end. If I stop asterisk, stop the zaptel serviceand restart, things seem ok for a while.I posted a similar problem a couple of days ago, and one of theresponses suggested that the TE4xxP may be on it's way out.
Is there any way of testing this card to see if that may be the case ?I was thinking of buying a sangoma a102 as a fall-over - are there anyissues with the sangoma cards, or should I buy another te4xxp as a backup ?
I was also thinking of moving the * server to a dell 2850 (2x3.06processors, 2GB ram, 2x146gb hdd) - again, any gotchas ?Sorry for so many questions, but we are placing / receiving near on 3000calls a day now and my butt is getting sore from all the kicking I've
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Re: [Asterisk-Users] Loss of Registration for SIP Trunks

2005-11-24 Thread Scott Clements

Hi Jerry & List,

I have the following registrations  in sip_additional.conf

register=02820:@202.177.XXX.XXX/02820

[02820]
type=user
secret=
host=202.177.XXX.XXX
context=from-pstn

sip_additional.conf is (or should be) included from sip.conf

Any other suggestions? Unfortuantely I wasn't backing up my conf files before this happened.

Scott

 --Message: 15Date: Thu, 24 Nov 2005 00:17:06 -0600
From: Jerry Jones <[EMAIL PROTECTED]>Subject: Re: [Asterisk-Users] Loss of Registration for SIP TrunksTo: Asterisk Users Mailing List - Non-Commercial Discussion<
asterisk-users@lists.digium.com>Message-ID: <[EMAIL PROTECTED]
>Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowedOn Nov 24, 2005, at 12:06 AM, Scott Clements wrote:> HI List,>> You'll have to pardon the newbieness of this question, I was
> editing the sip.conf file on my asterisk server yesterday, and now> none of my asterisk trunks will connect. From my knowledge sip.conf> does not effect registration, but there have been no other changes
> at all. Below is my sip.conf, and some other CLI info. If anone has> some thoughts please let me know.>>> [general]>> port = 5060   ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)> disallow=all> allow=g729> allow=ulaw> allow=alaw> context=from-pstn> ;context = from-sip-external ; Send unknown SIP callers to this
> context> callerid = Unknown> ;dtmfmode=rfc2833> ;relaxdtmf=yes>> #include sip_nat.conf> #include sip_custom.conf> #include sip_additional.conf>>>
>> cee*CLI> sip show registry>
HostUsername  
Refresh StateThis shows other servers to which asterisk has registered. I see noregister statements in your sip.conf above.>>> cee*CLI> sip show peers>
Name/usernameHostDyn
Nat ACL Mask> Port Status>
sip-out-test/02  202.177.222.24  255.255.255.255> 5060 Unmonitored>
127/127  (Unspecified)D  255.255.255.255> 0Unmonitored>
126/126  (Unspecified)D  255.255.255.255> 0UnmonitoredThese are sip clients registered to your asterisk server. I see nousers listed in your 
sip.conf above, though I guess they are in yourinclude files. I also looks like user sip-out-test has a hardcoded IPand is not set to dynamic so cannot really tell if it is registeredor not from this info. Users 127 and 126 are not registered. None
have a qualify to verify connectivity.perhaps restoring to your previous config and editing more slowlywill show where things broke:)> I have tried removing the trunks, confirmed the username and
> passwords for the trunks are ok. I am totally stumped as to what> would cause it.>> If anyone can help it'd be great :)>> SCott> ___
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RE: RE : RE : [Asterisk-Users] What does it mean?

Je ne donne pas de réponse !
Il me semble t'avoir suggèrer asterisk comme système
de messagerie vocale au lieu d'SEMS, avoir fourni
quelques fichiers de configuration, ce n'étaient pas
des devinettes.

Conbien de fois on ma répondu "personne n'est obligé
de faire ton tavail, tu n'as qu'a payé pour ce que tu
demandes.

IL me semble même me souvenir avoir lu un développeur
te faire la remarque "les utilisateurs de nos projets
vous ne profitez que de notre travail !".


Pour répondre à ton problème configure logger.conf .

Harry

  
--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:

> Cela veut simplement dire que tu te plains de ne pas
> avoir de réponses, mais
> qu'en fait tu n'en donnes pas non plus, sauf sous
> forme de devinette.
> Auquel cas, il est plus simple de ne pas répondre,
> 
> merci
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De
> la part de harry gaillac
> Envoyé : jeudi 24 novembre 2005 17:54
> À : Asterisk Users Mailing List - Non-Commercial
> Discussion
> Objet : RE: RE : [Asterisk-Users] What does it mean?
> 
> 
> Je ne connais pas la signification de "sybillines".
> Harry
> --- Olivier Taylor <[EMAIL PROTECTED]> a
> écrit
> :
> 
> > Tes réponses sont aussi sybillines que tes
> questions
> > :)
> > 
> > Olivier
> > 
> > -Message d'origine-
> > De : [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED]
> De
> > la part de harry gaillac
> > Envoyé : jeudi 24 novembre 2005 16:45
> > À : Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > Objet : RE: [Asterisk-Users] What does it mean?
> > 
> > 
> > Hello,
> > 
> > Read the Makefile in apps.
> > Harry
> > --- Olivier Taylor <[EMAIL PROTECTED]> a
> > écrit
> > :
> > 
> > > Hello,
> > > 
> > > I have compiled asterisk cvs under freebsd, no
> > > problems.
> > > 
> > > When starting asterisk, I get :
> > > 
> > > [res_config_mysql.so] => (MySQL RealTime
> > > Configuration Driver)
> > > /libexec/ld-elf.so.1:
> > > /usr/lib/asterisk/modules/res_config_mysql.so:
> > > Undefined symbol "ast_config_load"
> > > 
> > > What's wrong?
> > > 
> > > Olivier
> > > 
> > > ___
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> > Easynews.com
> > > --
> > > 
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > >
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >   
> > >
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> >
>
___
> > 
> > Appel audio GRATUIT partout dans le monde avec le
> > nouveau Yahoo! Messenger
> > Téléchargez cette version sur
> > http://fr.messenger.yahoo.com
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> > --
> > 
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> > Asterisk-Users@lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
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> >   
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
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> Easynews.com
> > --
> > 
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> 
> 
>   
> 
>   
>   
>
___
> 
> Appel audio GRATUIT partout dans le monde avec le
> nouveau Yahoo! Messenger 
> Téléchargez cette version sur
> http://fr.messenger.yahoo.com
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> --
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>
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> 







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RE: RE : RE : [Asterisk-Users] What does it mean?

Merci pour ces précisions.
Harry
--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:

> SIBYLLIN, INE. adj. Qui appartient aux sibylles. Il
> n'est guère usité au
> sens propre que dans ces locutions : Les oracles,
> les livres, les vers
> sibyllins, Les oracles, les livres, les vers des
> sibylles. 
> Il signifie au figuré Qui est mystérieux obscur,
> dont le sens est difficile
> à saisir. Il m'a répondu en termes sibyllins. Des
> paroles sibyllines. Un
> langage sibyllin.
> 
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De
> la part de harry gaillac
> Envoyé : jeudi 24 novembre 2005 17:54
> À : Asterisk Users Mailing List - Non-Commercial
> Discussion
> Objet : RE: RE : [Asterisk-Users] What does it mean?
> 
> 
> Je ne connais pas la signification de "sybillines".
> Harry
> --- Olivier Taylor <[EMAIL PROTECTED]> a
> écrit
> :
> 
> > Tes réponses sont aussi sybillines que tes
> questions
> > :)
> > 
> > Olivier
> > 
> > -Message d'origine-
> > De : [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED]
> De
> > la part de harry gaillac
> > Envoyé : jeudi 24 novembre 2005 16:45
> > À : Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > Objet : RE: [Asterisk-Users] What does it mean?
> > 
> > 
> > Hello,
> > 
> > Read the Makefile in apps.
> > Harry
> > --- Olivier Taylor <[EMAIL PROTECTED]> a
> > écrit
> > :
> > 
> > > Hello,
> > > 
> > > I have compiled asterisk cvs under freebsd, no
> > > problems.
> > > 
> > > When starting asterisk, I get :
> > > 
> > > [res_config_mysql.so] => (MySQL RealTime
> > > Configuration Driver)
> > > /libexec/ld-elf.so.1:
> > > /usr/lib/asterisk/modules/res_config_mysql.so:
> > > Undefined symbol "ast_config_load"
> > > 
> > > What's wrong?
> > > 
> > > Olivier
> > > 
> > > ___
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> > Easynews.com
> > > --
> > > 
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > >
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >   
> > >
> >
>
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Re: [Asterisk-Users] PRI problems again - What should I do ?

2005-11-24 Thread tim panton

On 24 Nov 2005, at 10:26, Julian Lyndon-Smith wrote:I know that's a real newbie question, but I have a problem.I keep getting frame rejects, and a D-channel bouncing up and down. BT say that it is at my end. If I stop asterisk, stop the zaptel service and restart, things seem ok for a while.Pardon me for asking the obvious, but...Have you _double_ checked the timing params in /etc/zaptel.conf?your bt span should say something like:span=1,1,0,ccs,hdb3or perhaps span=1,1,0,ccs,hdb3,crc4(if they have enabled crc on the link)I found that changes to this file only seemed to take effect on a cold start.Tim.http://www.westhawk.co.uk/  ___
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Re: [Asterisk-Users] (AMUG) Asterisk Montreal User Group today's meeting

2005-11-24 Thread Fred Blaise

On Thu, 2005-11-24 at 16:00 -0500, Adrien Laurent wrote:
> Hi,
> 
> This is just a reminder to inform you that the asterisk usergroup in 
> montreal will hold a meeting today at 4h45.
So much stuff in Montreal, can't wait to move up there :)

> 
> For more information, please visit:
> http://amug.modulis.ca/
> 
> See you there,
> 
> Adrien
> 
> 
> --
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> [EMAIL PROTECTED]
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[Asterisk-Users] Newbie requesting help!

2005-11-24 Thread Joao Carlos Mavimbe

Dear all.

I am new using asterisk.
I planned to have in my company an asterisk pbx that as a start would
be serving one analog phone, four sip hardphone extensions and two Iax
softphone. The next plan is to integrate asterisk with an old PBX
Alcatel 4100 and 3 lines to the public phone company.
So I bought a very basic digium card (10B) with one FXS module as a start.
I have been strugling for a month to get it going without success.
Installation went fine, zaptel driver loads correctly. What I can hear
is a constant ring on the connected phone (is this normal?) and when I
pick it up, I hear a dialtone. Went into configuration of zapata.conf.
Everything seems to be correct as when I run asterisk in verbose mode,
it gives no errors (it connects). Lifing the headset, I can still
hear a tone but when it comes to dialing , nothing happens
(meaning I hear the DTMF with the dialtone). I have been
fighting with the extension.conf file and try to
understand the logic behind it. Read so many things on the net
(voip wiki) but seems to be very difficult since I did not find
any "dummies" explanation for it.Can somebody help me configuring my basic asterisk pbx possible with an explanation on what the most important lines means?
Can somebody help me build the configuration for further expansion of my pbx system?

Regards.-- Joao Carlos de Timóteo Mavimbe

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[Asterisk-Users] Bad quality

2005-11-24 Thread Pablo Allietti

hi all, i have asterisk configured and working but the quality is very
poor. i ear noise and braks in the voice when the people talk to me, and
the people that eared me have the same problem any recommendation?
any files you need to post?
-- 

.-

Pablo Allietti
LACNIC

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Re: [Asterisk-Users] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...

2005-11-24 Thread Francesco Peeters

On Thu, November 24, 2005 20:09, Francesco Peeters said:
> On Wed, November 23, 2005 20:29, Francesco Peeters said:

>
> Just made myself a crossed NT1 connection to the NT mode card (as
> described on the PBX4linux site) and connected my phone.
>
> The zaphfc driver shows that layer 1 is activated (G3) once the phone is
> connected, but that is where it stops, as anything above that should be
> handled in chan_zap.
>
> However when I leave the card in bri_cpe_ptmp in zapata.conf, the layer2+
> protocols are not correct (TE mode) and when I put in bri_net_ptmp, the
> chan_zap somehow doesn't complete loading or exits in an unexpected
> manner, resulting in a situation where Asterisk stops loading it's
> configs, and thus runs without a dialplan and other modules...
>
> Seems to me there's an issue in that area: chan_zap, maybe libpri, etc.
>

I keep replying to myself...  ;-)

Some extra info:
/proc/zaptel/*:
Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)" AMI/CCS

   1 ZTHFC1/0/1 Clear (In use)
   2 ZTHFC1/0/2 Clear (In use)
   3 ZTHFC1/0/3 HDLCFCS (In use)
Span 2: ZTHFC2 "HFC-S PCI A ISDN card 1 [NT] layer 1 ACTIVATED (G3)" AMI/CCS

   4 ZTHFC2/0/1 Clear (In use)
   5 ZTHFC2/0/2 Clear (In use)
   6 ZTHFC2/0/3 HDLCFCS (In use)

/proc/pci:
  Bus  0, device   7, function  2:
USB Controller: Intel Corp. 82371AB/EB/MB PIIX4 USB (rev 1).
  IRQ 11.
  Master Capable.  Latency=32.
  I/O at 0xc000 [0xc01f].
  Bus  0, device   9, function  0:
Network controller: Cologne Chip Designs GmbH ISDN network controller
[HFC-P
CI] (rev 2).
  IRQ 11.
  Master Capable.  Latency=16.  Max Lat=16.
  I/O at 0xc400 [0xc407].
  Non-prefetchable 32 bit memory at 0xe3001000 [0xe30010ff].
  Bus  0, device  17, function  0:
Network controller: Cologne Chip Designs GmbH ISDN network controller
[HFC-P
CI] (#2) (rev 2).
  IRQ 11.
  Master Capable.  Latency=16.  Max Lat=16.
  I/O at 0xcc00 [0xcc07].
  Non-prefetchable 32 bit memory at 0xe3002000 [0xe30020ff].

I have tried all I can to assign different IRQ's to the HFC-PCI cards, but
they *always* take IRQ 11  :-(

-- 
Francesco Peeters

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RE: [Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue?? (Solved)

2005-11-24 Thread Aaron Clauson

 Hi,

I got the person to force the G729 codec on their Linksys WRT54GP2 and
forced it on Asterisk as well. The person then managed to get a single call
out but all subsequent call set ups failed with the same 488 error.

I went back over my SIP traces and noticed that the Cseq's were often out of
order or duplicated. This looked a lot more like the cause and was more
inline with a timing issue which would explain why it was only happening
over satellite. I did some more digging and came across the SIP timing
settings defined in the SIP RFC. I didn't get a chance too read exactly the
mecahnism but one of these settings does seem to be the interval between
resending INVITE requests.

The good news for me and anybody else reading this is with the same problem
is that changing the SIP T1 parameter does get the INVITE requests through.
It's on the SIP configuration page for the Linksys/Sipura devices. In this
case it was changed from the default 0.5s to 2s and then finally to 4s after
which outgoing call set up reliably worked.

In addition it does look like there is a bug in the Asterisk SIP channel
possibly to do with getting confused about receiving a bunch of INIVTE
requests with the same Cseq and stale nonces. It could be related to the
recent 403 problem for the Asterisk SIP channel and the Sipura REGISTER
requests with stale nonces. I will attempt to replicate the SIP dialogue and
produce a SIP trace and if successful file a bug report.

Aaron

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Aaron Clauson
> Sent: 24 November 2005 03:07
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Satellite, SIP Invite 488 Codec 
> Rejection,SIP Timing Issue??
> 
>  Hi,
> 
> Thanks for the tip I'll try it out. That would explain some 
> situations where
> one of the peeople concerned was mucking around with the 
> codec settings on
> the PAP2 and managed to get some calls out.
> 
> It's a bit baffling how the Linksys devices will get INVITES 
> through without
> G.729 being set across non-satellite links and yet can't get 
> the very same
> INVITE through across a satellite link. Fair enough if it was 
> the Linksys
> generating the 488 during the INVITE negotiation but how does 
> Asterisk even
> know the difference??
> 
> Aaron
> 
> > -Original Message-
> > From: Jason p [mailto:[EMAIL PROTECTED] 
> > Sent: 24 November 2005 02:25
> > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> > Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Satellite, SIP Invite 488 Codec 
> > Rejection, SIP Timing Issue??
> > 
> > I had the same problem when we were setting up these boxes 
> > after katrina. What i found is that they will only do one 
> > G729 session at a time. so that mesg that your showing is 
> > that its trying to register  two chans as 729. what i did to 
> > get around this was to turn off fource prefered codec on one 
> > line. This threw me for a loop also but trust me this is the 
> > fix, and yes you can only make one 729 call at a time.
> > 
> > 
> > Jason Price
> > 
> > 
> > On 11/23/05, Aaron Clauson <[EMAIL PROTECTED]> wrote: 
> > 
> > Hi,
> > 
> > I have a very strange Asterisk SIP call signalling 
> > problem that is proving
> > extremely difficult to track down. The problem is that 
> > any SIP INVITE
> > request that is coming into Asterisk over a satellite 
> > connection from a 
> > Linksys Router or PAP2 is getting a "Not Acceptable 
> > Here (codec error)" from
> > Asterisk. I've done all the normal checks on the 
> > allowed codecs in sip.conf
> > but to no avail.
> > 
> > I've even gone as far as writing a basic SIP stack to 
> > authenticate and send 
> > the INVITE request to Asterisk with exactly the same 
> > SDP payload to let me
> > brute force different options in the SDP request to try 
> > an narrow it down
> > that way. The preplexing thing from that length 
> > exercise is that if exactly 
> > the same INVITE request comes in from my app across the 
> > same satellite
> > connection to Asterisk it gets 200 Ok'ed but coming 
> > from the Linksys PAP2 or
> > WRT54GP2 it gets 488 Codec Not Acceptable Here'ed.
> > 
> > The first time this happened we went through all the 
> > usual checks and got 
> > nowhere and the person drifted off and it was put down 
> > to something speicifc
> > to that set up/connection. But now it's cropped up 
> > again with a different
> > person who also just happens to be on a satellite 
> > connection but from a 
> > different provider, although it is possible both 
> > providers use the same
> > infrastructure. In both cases incoming calls to the 
> > Linksys devices worked
> > correctly it's just the outgoing calls from the devices 
> > to Asterisk that are 
> > getting the rejection. In the second case we can't put 
> > it down to someth

Re: [Asterisk-Users] Asterisk + WiFi Phones

2005-11-24 Thread Cory Andrews

The F3000 is also a clamshell, "flip" type phone.  I should be receiving 
an eval unit shortly and will post my findings after we work it over in 
the lab.


Cory Andrews
Senior Partner
+++
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
+++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
fax - 716.630.1548



Luki wrote:


UTStarCom has the F3000 coming in December, which will have according
to their spec

   * WEP (64 and 128 bit )/WPA/MD5 Auth
   * Handover/Roaming between different AP and SSID
   



So what else is different compared to the F1000? The 1000 also does
WEP 64/128 and WPA with the newest firmware. Not sure about MD5 auth,
but SIP nonce/MD5 response certainly is implemented.

Roaming kind of works, but could be improved. In one place I made it
from 4th floor -> elevator -> lobby while on the phone and without any
noticeable dropouts (ulaw codec). But the building was covered with
access points, on average NetStumbler saw 6 at the same time. So it
works, but not always.

Don't get me wrong, the phone does have issues and in my opinion is
not production quality, meaning it will freak out unexpectedly and
only a reboot helps, which hardly ever happens to any Sipura adapters
or phones. Hopefully the new 3.6 firmware performs better.

Luki
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Re: [Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk

2005-11-24 Thread Dave Walker


What version of Asterisk are you using?

I had a similar problem with another make.  Since using Asterisk 1.20 
this issue has gone.


Dave Morrow wrote:

Hi all, I wonder if anyone out there has experienced an issue I am 
having with my Sipura / Linksys SPA-841 phones.


They work fine generally, but occasionally, incoming calls are 
missed.  It's like the SIP registration is expiring.  Does anyone know 
how to force the phone to re-register automatically? 



David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED] 
_http://www.autodata.net_

* PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE *

NEW !!! Tel: (519) 963-3020
Fax: (519) 451-6615 

< Poor planning on your part does not necessarily constitute an 
emergency on my part! >


This message has originated from Autodata Solutions. The attached 
material is the Confidential and Proprietary Information of Autodata 
Solutions. This email and any files transmitted with it are 
confidential and intended solely for the use of the individual or 
entity to whom they are addressed. If you have received this email in 
error please delete this message and notify the Autodata system 
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[Asterisk-Users] (AMUG) Asterisk Montreal User Group today's meeting

2005-11-24 Thread Adrien Laurent


Hi,

This is just a reminder to inform you that the asterisk usergroup in 
montreal will hold a meeting today at 4h45.


For more information, please visit:
http://amug.modulis.ca/

See you there,

Adrien


--
Adrien Laurent - CIO
(514) 284-2020 x 202
[EMAIL PROTECTED]
www.modulis.ca
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RE: [Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk

2005-11-24 Thread Benjamin Lawetz

Title: Linksys SPA-841 Disconnects from Asterisk



Check in you console or your logs when this happens. I'm 
guessing it's a Stale Nonce
 
If this is the case, Sipura supposedly fixed the bug on 
it's most recent firmware (At least for the SPA-1001 and SPA-2100, but I'm 
guessing the SPA-841 also)
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dave 
MorrowSent: November 24, 2005 3:49 PMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] 
Linksys SPA-841 Disconnects from Asterisk

Hi all, I wonder if anyone out there has experienced 
an issue I am having with my Sipura / Linksys SPA-841 phones. 
They work fine generally, but occasionally, incoming 
calls are missed.  It's like the SIP registration is expiring.  Does 
anyone know how to force the phone to re-register automatically?  

David A. Morrow Technical Systems Lead Autodata 
Solutions Company [EMAIL PROTECTED] http://www.autodata.net 
* PLEASE NOTE THAT EFFECTIVE 
DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * 
NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615  
< Poor planning on your part does 
not necessarily constitute an emergency on my part! > 
This message has originated from Autodata Solutions. 
The attached material is the Confidential and Proprietary Information of 
Autodata Solutions. This email and any files transmitted with it are 
confidential and intended solely for the use of the individual or entity to whom 
they are addressed. If you have received this email in error please delete this 
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[Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk

2005-11-24 Thread Dave Morrow

Title: Linksys SPA-841 Disconnects from Asterisk






Hi all, I wonder if anyone out there has experienced an issue I am having with my Sipura / Linksys SPA-841 phones.


They work fine generally, but occasionally, incoming calls are missed.  It's like the SIP registration is expiring.  Does anyone know how to force the phone to re-register automatically?  


David A. Morrow

Technical Systems Lead

Autodata Solutions Company

[EMAIL PROTECTED]

http://www.autodata.net


* PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE *


NEW !!! Tel: (519) 963-3020

Fax: (519) 451-6615 


< Poor planning on your part does not necessarily constitute an emergency on my part! >


This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] 


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Re: [Asterisk-Users] Call parking on Polycom IP501


From my original post:

"using ParkAndAnnouce puts the parked call on hold, hangs up the parker 
and then immediately calls them back with an announcement of the stall 
number"


So, I would say, yes :-)

On Nov 24, 2005, at 11:09 AM, Alvaro Parres wrote:


 Hi... I have the polycom 301 with firmware 1.6.3

 When i Press Park, i get a dialog to enter a extension.

 A dial 700 ther

 and the call get parked, and i recive a call announceme where the 
calls was parked.


 is this normal ???


On 11/24/05, Alvaro Parres <[EMAIL PROTECTED]> wrote:





On 11/24/05, Adam Goryachev < [EMAIL PROTECTED]> 
wrote:


I just tried it on my IP600, and when I press the park button, it 
waits
for me to dial an extension number, then I press park again, and it 
just

hangs up the call.

Thanks,
Adam

On Tue, 2005-11-22 at 13:56 -0800, Anthony Rodgers wrote:
> Hi there,
>
> Instead of asking a question, I thought I'd post an answer. I got 
the

> Polycom IP501 'Park' softkey working with * by doing the following:
>
> features.conf:
>
> [general]
> parkext => 1000
> parkpos => 1001-1009
> context => parkedcalls
> parkingtime => 120
> transferdigittimeout => 3
 > courtesytone = beep
>
> Nothing unusual there. Here's the neat bit:
>
> extensions.conf:
>
> [internal] ; or whatever the relevant context is for you - it's 
usually

> wherever your Polycom lives
> include => parkedcalls
> exten =>
> callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/
> ${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1)
>
> By using SIP DEBUG, I discovered that the Polycom attempts to 
re-invite
> the call to an extension called callpark. I couldn't get Park() to 
work

> (it announces the stall number to the parked caller, instead of the
> parker, for some reason), but using ParkAndAnnouce puts the parked 
call
> on hold, hangs up the parker and then immediately calls them back 
with

> an announcement of the stall number.
>
> Hope this helps someone out..
>
> Regards,

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Re: [Asterisk-Users] Looking for Windows based Asterisk

2005-11-24 Thread Mike Dent

On 11/24/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Without putty, my windows would be meaningless.
>
> PaulH
>
Subtle Paul! but nice! :)
Mike
UK
> - Original Message -
> From: "C F" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Friday, November 25, 2005 6:21 AM
> Subject: Re: [Asterisk-Users] Looking for Windows based Asterisk
>
>
> > I use putty.exe it works wonders.
> > available here:
> > http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html
> > You need ssh running on linux for it to work.
> >
> > On 11/24/05, Stefan-Michael. Guenther (in-put GbR) <[EMAIL PROTECTED]>
> wrote:
> > > Hi,
> > >
> > > >Does anyone know of a Asterisk Manager Interface client application
> that can
> > > >run from a Windows XP machine to manage Asterisk installed on a Linux
> > > >Machine.
> > > >
> > > if you consider the IE to be a client application, you could use the
> Asterisk
> > > PBX Manager from Thirdlane (www.thirdlane.com).
> > >
> > > Bye,
> > >
> > > Stefan
> > >
> > > --
> > >
> > > 
> > > in-put GbR - Das Linux-Systemhaus
> > > Stefan-Michael Guenther
> > > Moltkestrasse 49 D-76133 Karlsruhe
> > > Tel./Fax : +49 (0)721 / 83044 - 98/93
> > > http://www.in-put.de
> > > 
> > >  Schulungen  Installationen
> > >  Beratung   Support
> > >   Voice-over-IP-Lösungen
> > > 
> > >
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> >
>
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Re: [Asterisk-Users] Call parking on Polycom IP501


This is with Bootrom 2.6.2.0032, SIP 1.5.2.0054.

On Nov 24, 2005, at 3:32 AM, Adam Goryachev wrote:


What firmware version did you use for the polycom phone ??

I just tried it on my IP600, and when I press the park button, it waits
for me to dial an extension number, then I press park again, and it 
just

hangs up the call.

Thanks,
Adam

On Tue, 2005-11-22 at 13:56 -0800, Anthony Rodgers wrote:
> Hi there,
>
> Instead of asking a question, I thought I'd post an answer. I got 
the 

> Polycom IP501 'Park' softkey working with * by doing the following:
>
> features.conf:
>
> [general]
> parkext => 1000
> parkpos => 1001-1009
> context => parkedcalls
> parkingtime => 120
> transferdigittimeout => 3
> courtesytone = beep
>
> Nothing unusual there. Here's the neat bit:
>
> extensions.conf:
>
> [internal] ; or whatever the relevant context is for you - it's 
usually 

> wherever your Polycom lives
> include => parkedcalls
> exten => 
> callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/
> ${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1)
>
> By using SIP DEBUG, I discovered that the Polycom attempts to 
re-invite 
> the call to an extension called callpark. I couldn't get Park() to 
work 

> (it announces the stall number to the parked caller, instead of the 
> parker, for some reason), but using ParkAndAnnouce puts the parked 
call 
> on hold, hangs up the parker and then immediately calls them back 
with 

> an announcement of the stall number.
>
> Hope this helps someone out..
>
> Regards,

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Re: [Asterisk-Users] Call parking on Polycom IP501


Hi Adam,

Same - the parkee gets the stall number announcement instead of the 
parker.


On Nov 24, 2005, at 2:49 AM, Adam Goryachev wrote:


On Wed, 2005-11-23 at 12:53 -0800, Anthony Rodgers wrote:
> Hi Dave,
>
> exten => callpark,1,Dial(SIP/1000) didn't work - invalid extension

What about:
exten => callpark,1,Dial(Local/[EMAIL PROTECTED])

Regards,
Adam


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Re: [Asterisk-Users] Re: Queue Callback - SOLVED

2005-11-24 Thread pdhales

Excellent work!

PaulH

- Original Message - 
From: "Tyler" <[EMAIL PROTECTED]>
To: 
Sent: Friday, November 25, 2005 4:05 AM
Subject: [Asterisk-Users] Re: Queue Callback - SOLVED


> Happy Thanksgiving everyone..  I added the following page to the Wiki
> documenting how I solved this problem without having to hack with ICD or
> any commercial offerings.
>
> http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback
>
> Hope it can help somebody out.
>
> tf.
>
> > -Forwarded Message-
> > > From: Tyler <[EMAIL PROTECTED]>
> > > To: asterisk-users@lists.digium.com
> > > Subject: Queue Callback
> > > Date: Tue, 15 Nov 2005 08:39:27 -0500
> > >
> > > Hello,
> > >
> > > Does anyone have any information on configuring app_icd (or know of
any
> > > way to do it with the dialplan) that would allow a user holding in a
> > > queue to hang up, and have the system call them back when their place
in
> > > line comes up next?
> > >
> > > I can (obviously) allow them to '0' out to voicemail or something, but
I
> > > can only find vague references to app_icd and 'OrderlyQ' for doing
what
> > > I want to do...
> > >
> > > Anyone?
> > >
> > > Bueller? ;-)
> > >
> > > Thanks
> > >
> > > tf.
> > >
> > >
>
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RE: [Asterisk-Users] Eicon Diva Server query

2005-11-24 Thread Erik Slooff


> Hi Erik,
> You'll have to excuse my ignorance here. But why is this?
> 
> I don't have isdn4linux and capi4linux installed but do have 
> isdn4k-utils-devel-3.2-13.p1.1
> isdn4k-utils-3.2-13.p1.1
> 
> installed.
> 
> Is this for the capi20.h needed for chan_capi to compile?
> 
> thanks David

Hi Dave,

On my SuSE system the only way to get capi20.h is to install these rpm
packages; I like to compile as less as possible by hand... I have these
versions installed:
i4l-base 2005.8.15-2
Capi4linux 2005.8.15-2

Where capi4linux depends on i4l-base.

Whatever you prefer.

Erik

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Re: [Asterisk-Users] Looking for Windows based Asterisk

2005-11-24 Thread pdhales

Without putty, my windows would be meaningless.

PaulH

- Original Message - 
From: "C F" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Friday, November 25, 2005 6:21 AM
Subject: Re: [Asterisk-Users] Looking for Windows based Asterisk


> I use putty.exe it works wonders.
> available here:
> http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html
> You need ssh running on linux for it to work.
>
> On 11/24/05, Stefan-Michael. Guenther (in-put GbR) <[EMAIL PROTECTED]>
wrote:
> > Hi,
> >
> > >Does anyone know of a Asterisk Manager Interface client application
that can
> > >run from a Windows XP machine to manage Asterisk installed on a Linux
> > >Machine.
> > >
> > if you consider the IE to be a client application, you could use the
Asterisk
> > PBX Manager from Thirdlane (www.thirdlane.com).
> >
> > Bye,
> >
> > Stefan
> >
> > --
> >
> > 
> > in-put GbR - Das Linux-Systemhaus
> > Stefan-Michael Guenther
> > Moltkestrasse 49 D-76133 Karlsruhe
> > Tel./Fax : +49 (0)721 / 83044 - 98/93
> > http://www.in-put.de
> > 
> >  Schulungen  Installationen
> >  Beratung   Support
> >   Voice-over-IP-Lösungen
> > 
> >
> > ___
> > --Bandwidth and Colocation sponsored by Easynews.com --
> >
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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>

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RE: [Asterisk-Users] Looking for Windows based Asterisk

2005-11-24 Thread Anders Svensson

http://ipswitchboard.thorben.dk/index.php?option=com_content&task=view&id=26
&Itemid=46


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: den 24 november 2005 20:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Looking for Windows based Asterisk

I use putty.exe it works wonders.
available here:
http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html
You need ssh running on linux for it to work.

On 11/24/05, Stefan-Michael. Guenther (in-put GbR) <[EMAIL PROTECTED]>
wrote:
> Hi,
>
> >Does anyone know of a Asterisk Manager Interface client application that
can
> >run from a Windows XP machine to manage Asterisk installed on a Linux
> >Machine.
> >
> if you consider the IE to be a client application, you could use the
Asterisk
> PBX Manager from Thirdlane (www.thirdlane.com).
>
> Bye,
>
> Stefan
>
> --
>
> 
> in-put GbR - Das Linux-Systemhaus
> Stefan-Michael Guenther
> Moltkestrasse 49 D-76133 Karlsruhe
> Tel./Fax : +49 (0)721 / 83044 - 98/93
> http://www.in-put.de
> 
>  Schulungen  Installationen
>  Beratung   Support
>   Voice-over-IP-Lösungen
> 
>
> ___
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Re: [Asterisk-Users] 1.6.3 Polycom Firmware?

2005-11-24 Thread C F

According to this not:
http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,5082,00.pdf
but they do mentions some new blf support, so go figure.

On 11/24/05, Kevin Hanson <[EMAIL PROTECTED]> wrote:
> Kevin Ragsdale wrote:
>
> >Has anyone tried the newest Polycom firmware?  The release notes
> >indicate they have added support for a new BLA draft.
> >
> >TIA,
> >
> >Kevin
> >
> >
> Does anyone know if this new firmware support watching more than 7
> buddies at a time?
>
> Cheers,
> Kevin
> --
> Optimacy Communications, LLC
> http://www.optimacycomm.com
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[Asterisk-Users] chan_bluetooth

2005-11-24 Thread Dan


Hi,

I have compiled chan_bluetooth on FC4 (kernel 2.6.14-1).
The phone (SonyEricsson W800i) is paired with the BT dongle (ID 
0db0:1967 Micro Star International Bluetooth Dongle).


I have configured vi /etc/asterisk/bluetooth.conf like that:

[general]
rfchannel_hs = 2
rfchannel_ag = 3
interface = 0

channel = 2
autoconnect = yes

[00:12:EE:C0:7A:81]
name= W800
type= AG
channel = 13
autoconnect = yes


..but when I start Asterisk, I get the folllowing errors:

Nov 24 20:55:13 NOTICE[25742]: chan_bluetooth.c:2227 try_connect: 
Initialised bluetooth link to device W800

[AG]   W800 < AT+BRSF=23
Nov 24 20:55:13 ERROR[25742]: chan_bluetooth.c:2628 handle_rd_data: 
Device W800: Expected '\n' got 13. state = BLT_STATE_WANT_N2:



What can I do to solve this issue?

Thank you and best regards,
Dan


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Re: [Asterisk-Users] Looking for Windows based Asterisk

2005-11-24 Thread C F

I use putty.exe it works wonders.
available here:
http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html
You need ssh running on linux for it to work.

On 11/24/05, Stefan-Michael. Guenther (in-put GbR) <[EMAIL PROTECTED]> wrote:
> Hi,
>
> >Does anyone know of a Asterisk Manager Interface client application that can
> >run from a Windows XP machine to manage Asterisk installed on a Linux
> >Machine.
> >
> if you consider the IE to be a client application, you could use the Asterisk
> PBX Manager from Thirdlane (www.thirdlane.com).
>
> Bye,
>
> Stefan
>
> --
>
> 
> in-put GbR - Das Linux-Systemhaus
> Stefan-Michael Guenther
> Moltkestrasse 49 D-76133 Karlsruhe
> Tel./Fax : +49 (0)721 / 83044 - 98/93
> http://www.in-put.de
> 
>  Schulungen  Installationen
>  Beratung   Support
>   Voice-over-IP-Lösungen
> 
>
> ___
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Re: [Asterisk-Users] 1.6.3 Polycom Firmware?

2005-11-24 Thread Kevin Hanson


Kevin Ragsdale wrote:


Has anyone tried the newest Polycom firmware?  The release notes
indicate they have added support for a new BLA draft.

TIA,

Kevin
 

Does anyone know if this new firmware support watching more than 7 
buddies at a time?


Cheers,
Kevin
--
Optimacy Communications, LLC
http://www.optimacycomm.com
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Re: [Asterisk-Users] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...

2005-11-24 Thread Francesco Peeters

On Wed, November 23, 2005 20:29, Francesco Peeters said:
> On Wed, November 23, 2005 11:17, Francesco Peeters said:

>
> Just a question: Does the card require a device connected to it to start
> up in NT mode? I have been testing so far, so I have not yet connected my
> phone to the card with a cross-cable because I did not want to lose normal
> telephone access...
>

Just made myself a crossed NT1 connection to the NT mode card (as
described on the PBX4linux site) and connected my phone.

The zaphfc driver shows that layer 1 is activated (G3) once the phone is
connected, but that is where it stops, as anything above that should be
handled in chan_zap.

However when I leave the card in bri_cpe_ptmp in zapata.conf, the layer2+
protocols are not correct (TE mode) and when I put in bri_net_ptmp, the
chan_zap somehow doesn't complete loading or exits in an unexpected
manner, resulting in a situation where Asterisk stops loading it's
configs, and thus runs without a dialplan and other modules...

Seems to me there's an issue in that area: chan_zap, maybe libpri, etc.

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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Re: [Asterisk-Users] Call parking on Polycom IP501

2005-11-24 Thread Alvaro Parres

Hi... I have the polycom 301 with firmware 1.6.3

When i Press Park, i get a dialog to enter a extension.

A dial 700 ther

and the call get parked, and i recive a call announceme where the calls was parked.

is this normal ???
On 11/24/05, Alvaro Parres <[EMAIL PROTECTED]> wrote:
i have the 1.6.3 firmware and also when i press park i need to dial another extension..

On 11/24/05, Adam Goryachev <
[EMAIL PROTECTED]> wrote:
What firmware version did you use for the polycom phone ??I just tried it on my IP600, and when I press the park button, it waitsfor me to dial an extension number, then I press park again, and it justhangs up the call.
Thanks,AdamOn Tue, 2005-11-22 at 13:56 -0800, Anthony Rodgers wrote:> Hi there,>> Instead of asking a question, I thought I'd post an answer. I got the> Polycom IP501 'Park' softkey working with * by doing the following:
>> features.conf:>> [general]> parkext => 1000> parkpos => 1001-1009> context => parkedcalls> parkingtime => 120> transferdigittimeout => 3

> courtesytone = beep>> Nothing unusual there. Here's the neat bit:>> extensions.conf:>> [internal] ; or whatever the relevant context is for you - it's usually> wherever your Polycom lives
> include => parkedcalls> exten =>> callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/> ${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1)>> By using SIP DEBUG, I discovered that the Polycom attempts to re-invite
> the call to an extension called callpark. I couldn't get Park() to work> (it announces the stall number to the parked caller, instead of the> parker, for some reason), but using ParkAndAnnouce puts the parked call
> on hold, hangs up the parker and then immediately calls them back with> an announcement of the stall number.>> Hope this helps someone out..>> Regards,___
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Re: [Asterisk-Users] Call parking on Polycom IP501

2005-11-24 Thread Alvaro Parres

i have the 1.6.3 firmware and also when i press park i need to dial another extension..

On 11/24/05, Adam Goryachev <[EMAIL PROTECTED]> wrote:
What firmware version did you use for the polycom phone ??I just tried it on my IP600, and when I press the park button, it waitsfor me to dial an extension number, then I press park again, and it justhangs up the call.
Thanks,AdamOn Tue, 2005-11-22 at 13:56 -0800, Anthony Rodgers wrote:> Hi there,>> Instead of asking a question, I thought I'd post an answer. I got the> Polycom IP501 'Park' softkey working with * by doing the following:
>> features.conf:>> [general]> parkext => 1000> parkpos => 1001-1009> context => parkedcalls> parkingtime => 120> transferdigittimeout => 3
> courtesytone = beep>> Nothing unusual there. Here's the neat bit:>> extensions.conf:>> [internal] ; or whatever the relevant context is for you - it's usually> wherever your Polycom lives
> include => parkedcalls> exten =>> callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/> ${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1)>> By using SIP DEBUG, I discovered that the Polycom attempts to re-invite
> the call to an extension called callpark. I couldn't get Park() to work> (it announces the stall number to the parked caller, instead of the> parker, for some reason), but using ParkAndAnnouce puts the parked call
> on hold, hangs up the parker and then immediately calls them back with> an announcement of the stall number.>> Hope this helps someone out..>> Regards,___
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Re: [Asterisk-Users] Asterisk + WiFi Phones

2005-11-24 Thread Luki

> UTStarCom has the F3000 coming in December, which will have according
> to their spec
>
> * WEP (64 and 128 bit )/WPA/MD5 Auth
> * Handover/Roaming between different AP and SSID

So what else is different compared to the F1000? The 1000 also does
WEP 64/128 and WPA with the newest firmware. Not sure about MD5 auth,
but SIP nonce/MD5 response certainly is implemented.

Roaming kind of works, but could be improved. In one place I made it
from 4th floor -> elevator -> lobby while on the phone and without any
noticeable dropouts (ulaw codec). But the building was covered with
access points, on average NetStumbler saw 6 at the same time. So it
works, but not always.

Don't get me wrong, the phone does have issues and in my opinion is
not production quality, meaning it will freak out unexpectedly and
only a reboot helps, which hardly ever happens to any Sipura adapters
or phones. Hopefully the new 3.6 firmware performs better.

Luki
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[Asterisk-Users] SIP softphone with subscription/hint support?

2005-11-24 Thread Philipp von Klitzing

Hi there,

for testing purposes I am searching for a freely available softphone that 
supports SIP subscriptions and display the status of a few of these via 
e.g. a simulated LED. I know about

* EyeBeam (not free)
* SNOM softphone (needs Win XP and has old firmware)

Are there other softphones with this feature set around (that aren't 
fixed to one specific VoIP operator)?

Philipp


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Re: [Asterisk-Users] Modem Connections to PPP Server

2005-11-24 Thread Martin Joseph


On Nov 23, 2005, at 1:10 PM, Denis Vella wrote:

Hi,
 
I'm trying to use modems with Asterisk+VoIP Gateways in an attempt at providing an Internet service. 
 
Home_PC-->Modem-->PSTN-->VoIP_Gateway_FXO-->Ethernet-->Asterisk-->Ethernet-->VoIP_Gateway_FXS-->Modem-->PPP_Server-->Internet
 
I've been trying to use G711u and G711a codecs on the VoIP Gateways but, so far, no joy.   Has anyone got this to work?
Any pointers to setting this up?

Why would you want to do that?  If you have PSTN coming in why not use a regular modem bank? Oh wait,  let me guess you want to share the PSTN for VOIP?  This seems crazy to me?

Marty

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RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install

2005-11-24 Thread gw

I do believe there is a system reset is there not? Thought I saw it in
the manual.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Thursday, November 24, 2005 8:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk
install

Did you get it?  I would like to take a whack at it if not.



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, November 23, 2005 10:30 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk

> install
> 
> Does anyone know of a brute force that will work on a serial interface

> like hyperterminal?
> 
> --Jim
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
> Sent: Wednesday, November 23, 2005 8:22 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk

> install
> 
> Is the password limited to four digits like the Adtran 600 (I think)?
> 
> Start plugging in numbers.  Only 10,000 possible combinations.
> 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> > Sent: Wednesday, November 23, 2005 9:59 AM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for
Asterisk
> > install
> >
> > Thanks Jerry,
> >
> > I have called Carrier Access and they can reset the password but for
a
> > considerable fee.   We have serial access but after it boots it
> > immediately
> > asks for a username and password.  We have the username but the
> password
> > is
> > not what it is suppose to be.   There's a reset switch on the
> faceplate
> > but
> > I think the LOCAL SET is OFF and that is why it doesn't respond.
> Their
> > manual says the Reset switch is not under the control of LOCAL SET,
> yet it
> > doesn't seem to work.  Well, we might not know the proper boot
> sequence.
> > It
> > contains flash memory and there is a timing that important to that
> reset
> > procedure.  Anyone's help is much appreciated.
> >
> > --Jim
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Jerry
> Jones
> > Sent: Wednesday, November 23, 2005 7:40 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Help need to reset Adit 600 for
Asterisk
> > install
> >
> > Not sure but are you connecting via serial or ehternet? Seems to be
> the
> > serial had a way to do this easily on bootup. Otherwise I would be 
> > interested for future reference. Carrier Access does have a good
> support
> > team, just need to know your serial number.
> >
> > On Nov 23, 2005, at 12:13 AM, <[EMAIL PROTECTED]> <[EMAIL PROTECTED]>
> wrote:
> >
> > > Looking for a way to hard reset a ADIT 600 just purchased used.
> > > But it
> > > seems to have a master password already set.  We've tried the
front
> > > reset but maybe we don't have the right sequence of boot order.
Any
> > > help would be much appreciated?  - Jim
> > >
> > >
> > >
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>http

RE: [Asterisk-Users] CallerID not passing through to Polycom 500

2005-11-24 Thread gw

 
Check your logs, make sure you are waiting long enough before sending
the call to the polycom.

Uf asterisk sees the CID, it should send it and it should show up on the
polycom.

Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
MacKay
Sent: Thursday, November 24, 2005 11:19 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] CallerID not passing through to Polycom 500

I have a basic system working, except for callerid. The Polycom 500 just
shows call from "Business Line" on the screen. "Business Line" is the
name of the context that line is in. How do I get it to show the
callerID on the screen instead? Yes, I have CallerID on that line and it
works on a standard analog phone.
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[Asterisk-Users] Re: sip URL peering

2005-11-24 Thread Wolfgang S. Rupprecht

Klaus Darilion <[EMAIL PROTECTED]> writes:
> It's not that easy. If you want to have open SIP URIs (just like email
> is open for everybody) you will receive SPIT calls. E.g. the SPEER
> group tries to define rules for VoIP peering which allows
> authentication to enable open SIP URIs. (I won't open acces to my SIP
> URI if I can not verify the senders URI).

Keeping spam in mind seems like a really good idea.  I'm also a big
fan of keeping a cryptographic "paper trail" so that one can figure
out who spammed.

On the other hand, is SPAM / SPIT a big enough problem at this point
to warrant scuttling any interconnectivity?  It seems a bit premature
to worry about a problem that may not develop for 5 years and allow
that fear to stop direct sip dialing.

As an amusing aside, I inadvertently added a "captcha" to my phone
line when I had the local number go into an IVR that asks the caller
to press 1 for person XXX and 2 for person YYY and 3 of they are a
telemarketer.  I don't think anyone other than my friends has ever
pressed 3, but the predictive dialers used by the phone-spammers
doesn't seem to pass the turing test and isn't able to press 1 or 2.
;-) I see lots of timeout-hangups in the IVR with caller-id's like
"CAR PROMO" or "VOIP CALL".

If spam/spit is ever a problem, I'm simply routing previously unseen
calls to a turing test of the same type and anyone that has previously
called (and/or been called) gets to bypass the turing test.

-wolfgang
-- 
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
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[Asterisk-Users] Send fax using PRI connection to TE405P

2005-11-24 Thread Andre Courchesne - Consultant


Hi,

 Anyone has experiences with sending faxes using Asterisk and a TE405P 
Digium card (or similar PRI) with a PRI connection?


 Any insights wanted, bood, bad and ugly.

 Thanks,

Andre
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RE : RE : [Asterisk-Users] What does it mean?

Cela veut simplement dire que tu te plains de ne pas avoir de réponses, mais
qu'en fait tu n'en donnes pas non plus, sauf sous forme de devinette.
Auquel cas, il est plus simple de ne pas répondre,

merci

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 17:54
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: RE : [Asterisk-Users] What does it mean?


Je ne connais pas la signification de "sybillines".
Harry
--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:

> Tes réponses sont aussi sybillines que tes questions
> :)
> 
> Olivier
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De
> la part de harry gaillac
> Envoyé : jeudi 24 novembre 2005 16:45
> À : Asterisk Users Mailing List - Non-Commercial
> Discussion
> Objet : RE: [Asterisk-Users] What does it mean?
> 
> 
> Hello,
> 
> Read the Makefile in apps.
> Harry
> --- Olivier Taylor <[EMAIL PROTECTED]> a
> écrit
> :
> 
> > Hello,
> > 
> > I have compiled asterisk cvs under freebsd, no
> > problems.
> > 
> > When starting asterisk, I get :
> > 
> > [res_config_mysql.so] => (MySQL RealTime
> > Configuration Driver)
> > /libexec/ld-elf.so.1:
> > /usr/lib/asterisk/modules/res_config_mysql.so:
> > Undefined symbol "ast_config_load"
> > 
> > What's wrong?
> > 
> > Olivier
> > 
> > ___
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> Easynews.com
> > --
> > 
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> 
> 
>   
> 
>   
>   
>
___
> 
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> nouveau Yahoo! Messenger
> Téléchargez cette version sur
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>   
>
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>   
>
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> 







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Re: [Asterisk-Users] TDM400 FXO port 1 only problem.

2005-11-24 Thread Kevin Hanson


Tom Vile wrote:


Everyone,

I have a TDM400 REV I Ver 1 board and am having an issue with 1 of the
4 FXO channels.  FXO 1 always has clicks, pops and echo but the others
are crystal clear all of the time.  The card is on its own IRQ zztest
shows 100% to 99.98%  and is getting 1000 int per second.  Its not
dropping interrupts either.

I ran FXOTUNE and it did nothing to fix the issue

It only is happening on FXO port 1

Anything else to try?

Thanks,
--
Tom Vile
 

I had the same problem w/ two boards.  Called Digium and they said they 
had a batch of bad boards go out where port 1 would exhibit the problems 
you describe.  They rma'd the boards for me and all is well.


Cheers,
Kevin
--
Optimacy Communications, LLC
http://www.optimacycomm.com
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Re: [Asterisk-Users] Lag in speech

2005-11-24 Thread Thor Atle Rustad

I found the mail from Pauline Middelink!

filename: hfc_pci.c.diff



--- /root/hfc_pci.c Wed Aug  7 15:31:24 2002
+++ /usr/src/linux/drivers/isdn/hisax/hfc_pci.c Thu Oct 31 10:18:05 2002
@@ -270,8 +270,16 @@
if (fifo_state)
cs->hw.hfcpci.fifo_en ^= fifo_state;
Write_hfc(cs, HFCPCI_FIFO_EN, cs->hw.hfcpci.fifo_en);
-   bzt->za[MAX_B_FRAMES].z1 = B_FIFO_SIZE + B_SUB_VAL - 1;
-   bzt->za[MAX_B_FRAMES].z2 = bzt->za[MAX_B_FRAMES].z1;
+   /* Notice the z2 is readonly, and could be active when we enter this
+* function. (I.e. changing.) When we now reset z1 to MAXSIZE, the
+* FIFO thinks there is data and runs it when re-enabled...
+* To prevent this from happening, we make z1 ONE higher than z2, so
+* when the FIFO gets re-enabled, it thinks it only has to send a
+* single byte, which hopefully nobody notices (1/8000 second?)
+* (Pauline Middelink - 2002) */
+   bzt->za[MAX_B_FRAMES].z1 = bzt->za[MAX_B_FRAMES].z2 + 1;
+   if (bzt->za[MAX_B_FRAMES].z1 >= B_FIFO_SIZE + B_SUB_VAL)
+   bzt->za[MAX_B_FRAMES].z1 -= B_FIFO_SIZE;
bzt->f1 = MAX_B_FRAMES;
bzt->f2 = bzt->f1;  /* init F pointers to remain constant */
if (fifo_state)
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RE : RE : [Asterisk-Users] What does it mean?

SIBYLLIN, INE. adj. Qui appartient aux sibylles. Il n'est guère usité au
sens propre que dans ces locutions : Les oracles, les livres, les vers
sibyllins, Les oracles, les livres, les vers des sibylles. 
Il signifie au figuré Qui est mystérieux obscur, dont le sens est difficile
à saisir. Il m'a répondu en termes sibyllins. Des paroles sibyllines. Un
langage sibyllin.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 17:54
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: RE : [Asterisk-Users] What does it mean?


Je ne connais pas la signification de "sybillines".
Harry
--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:

> Tes réponses sont aussi sybillines que tes questions
> :)
> 
> Olivier
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De
> la part de harry gaillac
> Envoyé : jeudi 24 novembre 2005 16:45
> À : Asterisk Users Mailing List - Non-Commercial
> Discussion
> Objet : RE: [Asterisk-Users] What does it mean?
> 
> 
> Hello,
> 
> Read the Makefile in apps.
> Harry
> --- Olivier Taylor <[EMAIL PROTECTED]> a
> écrit
> :
> 
> > Hello,
> > 
> > I have compiled asterisk cvs under freebsd, no
> > problems.
> > 
> > When starting asterisk, I get :
> > 
> > [res_config_mysql.so] => (MySQL RealTime
> > Configuration Driver)
> > /libexec/ld-elf.so.1:
> > /usr/lib/asterisk/modules/res_config_mysql.so:
> > Undefined symbol "ast_config_load"
> > 
> > What's wrong?
> > 
> > Olivier
> > 
> > ___
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> Easynews.com
> > --
> > 
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   
> >
>
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> > 
> 
> 
> 
>   
> 
>   
>   
>
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> nouveau Yahoo! Messenger
> Téléchargez cette version sur
> http://fr.messenger.yahoo.com
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> --
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>   
>
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Re: [Asterisk-Users] :: Success Case => Motorola 62802-51 as FXO device::

2005-11-24 Thread Nestor A. Diaz


Walter Willis wrote:


not work fine


Actually it is recognized as an x100p device:

Nov 21 19:54:34 asterix kernel: Zapata Telephony Interface Registered on 
major 196

Nov 21 19:54:34 asterix kernel: PCI: Found IRQ 5 for device 00:0d.0
Nov 21 19:54:34 asterix kernel: wcfxo: DAA mode is 'FCC'
Nov 21 19:54:34 asterix kernel: Found a Wildcard FXO: Wildcard X100P
Nov 21 19:54:34 asterix kernel: Registered tone zone 0 (United States / 
North America)


i have been able to call from the outside and the default greeting sound 
good, but it can not recognize tones, when program the extension to dial 
an inside line the sound is very bad, too much noise !!! i think the 
problem is with full duplex.


it will be nice to investigate if we can modify the sources to make this 
chipset (62802-52) work with asterisk in a nice way, i have been dealing 
with rxgain and txgain in order to tune the card, but i have failed, the 
sound is still bad.


62802 is one of the chipset that it is still available on the market, it 
is not designed to compete against digium analog card, is designed to 
introduce people on the voip field, for this it is important to be 
supported, think of  PC vs. Apple, the more people will use Asterisk the 
best the business will become.


Somebody have deal with zapata sources in order to make some changes and 
make that chipset works ? does anyone have tried newer intel modem 
chipset with asterisk ? they work ? the only chipset that works for me 
was the ambient md3200, have some echo problems but with echo 
chancelation and training things get better after a few seconds.


What are the requeriments for a modem chipset to be supported on asterisk ?

p.d. i am searching for ambiend md 3200 cards, anybody know where i can 
buy them ? at a reasonable price off course.


Thanks everyone.

--
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-600-5490 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:[EMAIL PROTECTED]
Email/MSN: [EMAIL PROTECTED]
http://www.tiendalinux.com/
Bogota, Colombia 


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[Asterisk-Users] Re: Asterisk not picking up calls.

2005-11-24 Thread David Yat Sin









This is usually a problem one of the pair not physically
disconnected, i.e loose plug/socket connection. Try
to replace your connectors and test your cable

 

David
Yat Sin

Sangoma
Technologies

(905) 474-1990
x119

(800)
388-2475 x119

Fax:
(905) 474 9223

MSN:
[EMAIL PROTECTED]

Email: [EMAIL PROTECTED]m

Website:
www.sangoma.com

 






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Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

2005-11-24 Thread Bharath

I found out that I have a faulty Belkin Router which was causing the
problem. I tried forwarding ports as well as DMZ'd the Sip device but
still could'nt not hear the voice. So i plugged the sip device directly
to the cable modem & it worked fine. So my guess is that though I
have set up the router to forwards port to the sip device there is
something happening at the router that is blocking the RTP ports
(1-2).
Thanks
On 11/24/05, Tom Rymes <[EMAIL PROTECTED]> wrote:
On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote:[snip]> Well, as the user stated on the original message, the asterisk> server is behind a NAT and the client is also behind a NAT….>> if you make it work just by opening ports, let me know..I have
> never been able to get it to work, that's why I don't use sip, just> plain iax2 for everything… J>> Manny>Manny,I have this working as I write this. (I just hung up the phone.) In
fact, I brought a Cisco 7940G to a completely unknown nat-ed networkthe other day, plugged it in and started making calls right away.Here's the setup I have for this specific configuration:1.) Asterisk server behind NAT. (Setup as DMZ on Linksys WRT54G, but
it's still NAT. I just don't have to forward ports this way)2.) externip, localnet, nat settings configured in the sip.conf file(sip_nat.conf for [EMAIL PROTECTED])3.) Cisco phone (or whatever SIP UA you choose) configured for NAT
(via the SIP.cnf file for Cisco)4.) Lather, rinse, repeat if necessaryHopefully that will work for you. I'd rather use IAX and avoid theseproblems altogether, but I have yet to find an IAX hardphone I am
willing to use. In fact, for softphone use, I do indeed use IAX viaLoudHush for the mac. (Great piece of software, BTW. No connectionhere, just a happy user...)TomTom Rymes
Cascade Link Systemswww.cascadelinksystems.com(603) 375-1414"Intelligent technology solutions for small businesses."___
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RE: RE : [Asterisk-Users] What does it mean?

2005-11-24 Thread Dave Cotton

On Thu, 2005-11-24 at 17:54 +0100, harry gaillac wrote:
> Je ne connais pas la signification de "sybillines".

http://www.village-justice.com/forum/viewtopic.php?t=1224&start=0&sid=964c2c9a1cd842eaca284be8899028a8


-- 
Dave Cotton <[EMAIL PROTECTED]>

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[Asterisk-Users] Re: Queue Callback - SOLVED

2005-11-24 Thread Tyler

Happy Thanksgiving everyone..  I added the following page to the Wiki
documenting how I solved this problem without having to hack with ICD or
any commercial offerings.

http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback

Hope it can help somebody out.

tf.

> -Forwarded Message-
> > From: Tyler <[EMAIL PROTECTED]>
> > To: asterisk-users@lists.digium.com
> > Subject: Queue Callback
> > Date: Tue, 15 Nov 2005 08:39:27 -0500
> > 
> > Hello,
> > 
> > Does anyone have any information on configuring app_icd (or know of any
> > way to do it with the dialplan) that would allow a user holding in a
> > queue to hang up, and have the system call them back when their place in
> > line comes up next?  
> > 
> > I can (obviously) allow them to '0' out to voicemail or something, but I
> > can only find vague references to app_icd and 'OrderlyQ' for doing what
> > I want to do...
> > 
> > Anyone?
> > 
> > Bueller? ;-)
> > 
> > Thanks
> > 
> > tf.
> > 
> > 

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RE: RE : [Asterisk-Users] What does it mean?

Je ne connais pas la signification de "sybillines".
Harry
--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:

> Tes réponses sont aussi sybillines que tes questions
> :)
> 
> Olivier
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De
> la part de harry gaillac
> Envoyé : jeudi 24 novembre 2005 16:45
> À : Asterisk Users Mailing List - Non-Commercial
> Discussion
> Objet : RE: [Asterisk-Users] What does it mean?
> 
> 
> Hello,
> 
> Read the Makefile in apps.
> Harry
> --- Olivier Taylor <[EMAIL PROTECTED]> a
> écrit
> :
> 
> > Hello,
> > 
> > I have compiled asterisk cvs under freebsd, no
> > problems.
> > 
> > When starting asterisk, I get :
> > 
> > [res_config_mysql.so] => (MySQL RealTime
> > Configuration Driver)
> > /libexec/ld-elf.so.1:
> > /usr/lib/asterisk/modules/res_config_mysql.so:
> > Undefined symbol "ast_config_load"
> > 
> > What's wrong?
> > 
> > Olivier
> > 
> > ___
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> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> 
> 
>   
> 
>   
>   
>
___
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>   
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>   
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[Asterisk-Users] Voicemail notifications alwats sent as [EMAIL PROTECTED]

2005-11-24 Thread Andre Courchesne - Consultant


Hi,

 I have a problem with e-mail notifications. For some reason Asterisk 
does not use the serveremail configuration when sending e-mails 
notifications. it always send it using [EMAIL PROTECTED]


My configuration:
pbxskip=yes ; Don't put [PBX]: in the subject line
[EMAIL PROTECTED]
fromstring=Voicemail System ; Real name of email sender
maxmessage=180  ; max length of vm message
minmessage=3; Minimum length of a voicemail 
message in seconds
maxsilence=5; Wait for 5 silent seconds and end the 
voicemail

silencethreshold=128; What do we consider to be silence
skipms=3000 ; How many miliseconds 
to skip forward/back when rew/ff in message playback
review=yes  ; Allow sender to 
review/rerecord their message before saving it

operator=yes; Allow caller to press 0

Exim log when sending an e-mail:
Nov 24 11:44:21 privalodc exim[8501]: 2005-11-24 11:44:21 
1EfKCr-0002D5-N1 ** [EMAIL PROTECTED] R=dnslookup T=remote_smtp: 
SMTP error from remote mail server after RCPT 
TO:<[EMAIL PROTECTED]>: host mail.privalodc.com 
[207.115.102.XX]: 504 <[EMAIL PROTECTED]>: Sender address rejected: need 
fully-qualified address


Any ideas?

Andre

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RE : [Asterisk-Users] What does it mean?

Tes réponses sont aussi sybillines que tes questions :)

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 16:45
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [Asterisk-Users] What does it mean?


Hello,

Read the Makefile in apps.
Harry
--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:

> Hello,
> 
> I have compiled asterisk cvs under freebsd, no
> problems.
> 
> When starting asterisk, I get :
> 
> [res_config_mysql.so] => (MySQL RealTime
> Configuration Driver)
> /libexec/ld-elf.so.1:
> /usr/lib/asterisk/modules/res_config_mysql.so:
> Undefined symbol "ast_config_load"
> 
> What's wrong?
> 
> Olivier
> 
> ___
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> Asterisk-Users mailing list
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>
http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   
>
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[Asterisk-Users] H323 to H323 calls problem

2005-11-24 Thread Javier Oviedo

This is my network scheme:


h323 endpoint1 ... endpint10 <=> gk1 <=> gk2 <=> asterisk

GK1 configuration: routed mode
GK2 configuration: direct mode

How to obtain that rtp channels not through asterisk for h323 to h323 calls

Thanks in advance!

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Re: [Asterisk-Users] Re: sip URL peering

2005-11-24 Thread Klaus Darilion


Wolfgang S. Rupprecht wrote:

Klaus Darilion <[EMAIL PROTECTED]> writes:


There is a new ietf WG to come which deals with peering issues. It's
called SPEER (formerly VOIPEER)

The list archive is at
http://darkwing.uoregon.edu/~llynch/voipeer/

minutes from last ietf meeting:
http://www3.ietf.org/proceedings/05nov/minutes/voipeer.html



It looks interesting, but these things always seem to be scuttled or
reduced to glacial progress by the telecom interests.

VOIP peering isn't something that should require years of meeting to
make happen.


It's not that easy. If you want to have open SIP URIs (just like email 
is open for everybody) you will receive SPIT calls. E.g. the SPEER group 
tries to define rules for VoIP peering which allows authentication to 
enable open SIP URIs. (I won't open acces to my SIP URI if I can not 
verify the senders URI).


regards
klaus
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[Asterisk-Users] CallerID not passing through to Polycom 500

2005-11-24 Thread Gary MacKay

I have a basic system working, except for callerid. The Polycom 500 just 
shows call from "Business Line" on the screen. "Business Line" is the 
name of the context that line is in. How do I get it to show the 
callerID on the screen instead? Yes, I have CallerID on that line and it 
works on a standard analog phone.

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Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

2005-11-24 Thread Tom Rymes


On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote:

[snip]
Well, as the user stated on the original message, the asterisk  
server is behind a NAT and the client is also behind a NAT….


if you make it work just by opening ports, let me know..I have  
never been able to get it to work, that’s why I don’t use sip, just  
plain iax2 for everything… J


Manny


Manny,

I have this working as I write this. (I just hung up the phone.) In  
fact, I brought a Cisco 7940G to a completely unknown nat-ed network  
the other day, plugged it in and started making calls right away.  
Here's the setup I have for this specific configuration:


1.) Asterisk server behind NAT. (Setup as DMZ on Linksys WRT54G, but  
it's still NAT. I just don't have to forward ports this way)
2.) externip, localnet, nat settings configured in the sip.conf file  
(sip_nat.conf for [EMAIL PROTECTED])
3.) Cisco phone (or whatever SIP UA you choose) configured for NAT  
(via the SIP.cnf file for Cisco)

4.) Lather, rinse, repeat if necessary

Hopefully that will work for you. I'd rather use IAX and avoid these  
problems altogether, but I have yet to find an IAX hardphone I am  
willing to use. In fact, for softphone use, I do indeed use IAX via  
LoudHush for the mac. (Great piece of software, BTW. No connection  
here, just a happy user...)


Tom


Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

"Intelligent technology solutions for small businesses."


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RE: [Asterisk-Users] What does it mean?

Hello,

Read the Makefile in apps.
Harry
--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:

> Hello,
> 
> I have compiled asterisk cvs under freebsd, no
> problems.
> 
> When starting asterisk, I get :
> 
> [res_config_mysql.so] => (MySQL RealTime
> Configuration Driver)
> /libexec/ld-elf.so.1:
> /usr/lib/asterisk/modules/res_config_mysql.so:
> Undefined symbol "ast_config_load"
> 
> What's wrong?
> 
> Olivier
> 
> ___
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> Asterisk-Users mailing list
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>
http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   
>
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> 







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[Asterisk-Users] What does it mean?