[Asterisk-Users] Flash Zap Channel

[Tom Rymes]

After having looked around on google, voip-info, etc, I am coming  
here for a bit of insight from anyone who might be willing to provide  
some


I am using * (it's [EMAIL PROTECTED], but this should apply to any * install) to  
handle my home phone line. I have a TDM11B card, and my Verizon line  
comes in on the FXO port and my cordless phone is plugged in to the  
FXS port. Everything works just dandy, except for call waiting on the  
Verizon line, since when I press flash on my cordless, it flashes the  
FXS port, not the Verizon line.


OK, no problem, after looking online, it seems that I need to use the  
Flash() command to flash the Verizon line by creating an entry in my  
dialplan (extensions_custom.conf since I use [EMAIL PROTECTED]) that flashes the  
Verizon line and then transfers the call back to me. So I found  
http://voip-info.org/tiki-index.php?page=Asterisk%20cmd% 
20Flash#comments on the Wiki, but that does not seem to work for me.  
(I do have call waiting enabled on my cordless extension, BTW). The  
calling party hears Allison say All Circuits are Busy Here's what I  
tried:


exten = 11,1,Flash()
exten = 11,2,Goto(ext-local,200,1)

I also tried:
exten = 11,1,Playback(transfer)
exten = 11,2,Flash()
exten = 11,3,transfer(200)
exten = 11,4,Hangup()

The idea being that I would transfer the call to extension 11, which  
would transfer it back to me. The problem, I think, is that * can't  
reach me on my cordless, b/c I already have two calls going (The  
orignal and me transferring to 11)


I'm sure that I am not the first to try this, and I'm sure that the  
way I am going about this is boneheaded, so can someone point out the  
best way to flash the incoming Verizon line whenever I have call  
waiting? (I'm considering cancelling call-waiting and having the  
Verizon line Call-forward-busy to a broadvoice number, but you would  
think I could dream up a better solution)


Tom

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Re: [Asterisk-Users] Flash and zap and # key

[Wilson Pickett]

 I'm not sure if this is supposed to happen, but when I press the # key it
 seems to have the effect of flashing the hook, or at least letting me
 transfer.  I am using Zap hardware.
Do you hear a transfer voice prompt? Asterisk will intercept the #
key if told to do so in the Dial application. In STABLE this is
hard-coded I think.

show application dial will give the details of how T and t can be used
as options.

If your dial commands are not using T and t as options, I'm not sure
what is happening.
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[Asterisk-Users] IAX DTMF Problem...

[Mark Edwards]

Hi.

Probably been asked before, but my IAX provider assures me its not their problem

I have a IAX connection to a peer providing a DID. I am dialing up
my number, seeing the DTMF tones come down the line, and the * IVR is
just ignoring them.

IAX debug output is:

Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: DTMF Subclass: 1 Timestamp: 02608ms SCall: 00016 DCall: 3 [
210.80.176.12:4569]Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK
 Timestamp: 02608ms SCall: 3 DCall: 00016 [210.80.176.12:4569]
for a press of 1

I am assuming this is the DTMF inband problem, but I appear unable to convince my provider.

Can I work around this on * or do I have to go back to SIP?

Mark


-- regards,Mark P. EdwardsTEL:+61 408 601 107SKYPE: mark.p.edwards
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[Asterisk-Users] Outbound answer on TDM400P

[Me]

How come an outgoing call using my TDM400P immediately
say the call is answered?  I'd like to be able to
detect when the call is actually picked up, is this
possible?

If this is normal with analog cards, does the same
thing happen with T1 cards?

-L



 
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Re: [Asterisk-Users] Linux Firewall Question

[Tzafrir Cohen]

On Thu, Jun 30, 2005 at 06:56:16PM -0500, OMS wrote:
 Hi,
 I am running Asterisks on Public IP with Fedora Core 3. 
 
 What is the recommendation for making Linux secure on the Public IP since 
 I am new to Linux. Which Firewall should I use?  I am not intending to 
 use Linux as router. 

I wrote a simple iptables script for Rapid. You can find it inside the
rapid-scripts deb package. Not as modular as shorewall, but then again,
fits the bill for a one-interface server.

It is part of the rapid-script debian package, e.g:

  http://updates.xorcom.com/rapid/pool/xorcom/rapid-scripts_1.0.1_all.deb

Extract etc/init.d/firewall from it. The command 'clear' will clear all
rules. Otherwise it will set the iptables rules.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Voicemail = SMS

[snacktime]

On 6/30/05, Mark Charlton [EMAIL PROTECTED] wrote:
 Hi
 
 I have been trying for a while to find a way to get an SMS send when I
 receive a voicemail into my asterisk system.  I don't want to send an
 SMS if the caller doesn't leave a message.  I have voicemail.conf set
 up to email and delete.

 Any and all suggestions will be greatly appreciated.

The manager action MailboxCount gives the number of old and new
messages in a mailbox.  You would have to call the manager via an agi
but it would give you the info you want.

Chris
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[Asterisk-Users] Caller ID problem..

[Sukardi Shahdan]

hi all,

i want to ask about caller id in asterisk.

if i make outgoing call, people who receive my call
can see the pilot number.

what i must do if i want that people receive my other
number (DID number)?

is it posibble?

thank to everybody..

regards,
shahdan

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[Asterisk-Users] Got this error after my installation when i do ztcfg -vv

[Ian Bert Tusil]

how can i solve the error on the last part?

need help. thnx...


Zaptel Configuration
==

SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: Individual Clear channel (Default) (Slaves: 03)
Channel 04: Individual Clear channel (Default) (Slaves: 04)
Channel 05: Individual Clear channel (Default) (Slaves: 05)
Channel 06: Individual Clear channel (Default) (Slaves: 06)
Channel 07: Individual Clear channel (Default) (Slaves: 07)
Channel 08: Individual Clear channel (Default) (Slaves: 08)
Channel 09: Individual Clear channel (Default) (Slaves: 09)
Channel 10: Individual Clear channel (Default) (Slaves: 10)
Channel 11: Individual Clear channel (Default) (Slaves: 11)
Channel 12: Individual Clear channel (Default) (Slaves: 12)
Channel 13: Individual Clear channel (Default) (Slaves: 13)
Channel 14: Individual Clear channel (Default) (Slaves: 14)
Channel 15: Individual Clear channel (Default) (Slaves: 15)
Channel 16: Individual Clear channel (Default) (Slaves: 16)
Channel 17: Individual Clear channel (Default) (Slaves: 17)
Channel 18: Individual Clear channel (Default) (Slaves: 18)
Channel 19: Individual Clear channel (Default) (Slaves: 19)
Channel 20: Individual Clear channel (Default) (Slaves: 20)
Channel 21: Individual Clear channel (Default) (Slaves: 21)
Channel 22: Individual Clear channel (Default) (Slaves: 22)
Channel 23: Individual Clear channel (Default) (Slaves: 23)
Channel 24: D-channel (Default) (Slaves: 24)

24 channels configured.

Notice: Configuration file is /etc/zaptel.conf
line 145: Unable to open master device '/dev/zap/ctl'
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Re: [Asterisk-Users] Caller ID problem..

[Sukardi Shahdan]

hello all,

what i mean is, i make a call from another did number
but people receive the pilot number.

i don't know how to do :( 

i try this but nothing happen.

exten = _01,1,SetCIDNum(0${CALLERIDNUM})
exten = _01,2,Dial(${TRUNK}/${EXTEN})

please help me..

regards,
shahdan

--- Sukardi Shahdan [EMAIL PROTECTED] wrote:

 hi all,
 
 i want to ask about caller id in asterisk.
 
 if i make outgoing call, people who receive my call
 can see the pilot number.
 
 what i must do if i want that people receive my
 other
 number (DID number)?
 
 is it posibble?
 
 thank to everybody..
 
 regards,
 shahdan
 
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 protection around 
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[Asterisk-Users] Asterisk and DHCP

[laine . marko]

Hi all!

I am working with asterisk and trying to get it work in DHCP network, where
asterisk gets a DHCP address as well as other computers (IP-phones). So far,
I've have got asterisk working with static IP address where phones are getting
their IP from DHCP server.
Is it possible at all to phones to find asterisk server if it gets random IP
address from the DHCP server also? I mean, is there some settings in asterisk
how I can bypass IP settings and force it to work with dynamic IP address?

Thanks for any help and guidance!




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RE: [Asterisk-Users] Caller ID problem..

[Florian Overkamp]

Hi, 

 -Original Message-
 what i mean is, i make a call from another did number
 but people receive the pilot number.
 
 i don't know how to do :( 
 
 i try this but nothing happen.
 
 exten = _01,1,SetCIDNum(0${CALLERIDNUM})
 exten = _01,2,Dial(${TRUNK}/${EXTEN})
 
 please help me..

It is very much dependant on what type of line (analog, T1, E1) you have and
what your telco allows you to do, so ask your telco.

In many of our setups, it is required that callerid's are sent in a specific
format, the national number without leading 0 to be exact. Similar
requirements may apply to your setup.

Talk to your telco.

Florian 


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RE: [Asterisk-Users] Asterisk and DHCP

[David Masure]

Hi,

I don't think it's goonna work that way !

The phones need to register with the asterisk server.  So when you
configure the phones, you specify the ip of the asterisk server... so...

Another way, if you want to work with dynamic ip address would be to set
up a DNS (name resolver) which handles dynamic IP and so, you could
configure your IP phone with the name of your asterisk server instead of
the IP address...

I hope I've helped you.

Best regards

David Masure



-Message d'origine-
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Envoyé : vendredi 1 juillet 2005 09:15
À : 'asterisk-users@lists.digium.com'
Objet : [Asterisk-Users] Asterisk and DHCP


Hi all!

I am working with asterisk and trying to get it work in DHCP network,
where
asterisk gets a DHCP address as well as other computers (IP-phones). So
far,
I've have got asterisk working with static IP address where phones are
getting
their IP from DHCP server.
Is it possible at all to phones to find asterisk server if it gets
random IP
address from the DHCP server also? I mean, is there some settings in
asterisk
how I can bypass IP settings and force it to work with dynamic IP
address?

Thanks for any help and guidance!




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[Asterisk-Users] ${BLINDTRANSFER} in *-1.0.X

[Ivan Meic (Vox Mundi)]

Hi everybody,

I'm not a programmer, so I really don't know this,
but is it possible to somehow backport ${BLINDTRANSFER}
variable functionality to 1.0.X versions of *.

I need this really badly ... thanks for your replies.

Ivan
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RE: [Asterisk-Users] Asterisk and DHCP

[laine . marko]

You helped me, David, Thanks!

Jussi

Quoting David Masure [EMAIL PROTECTED]:


 Hi,

 I don't think it's goonna work that way !

 The phones need to register with the asterisk server.  So when you
 configure the phones, you specify the ip of the asterisk server... so...

 Another way, if you want to work with dynamic ip address would be to set
 up a DNS (name resolver) which handles dynamic IP and so, you could
 configure your IP phone with the name of your asterisk server instead of
 the IP address...

 I hope I've helped you.

 Best regards

 David Masure



 -Message d'origine-
 De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 Envoyé : vendredi 1 juillet 2005 09:15
 À : 'asterisk-users@lists.digium.com'
 Objet : [Asterisk-Users] Asterisk and DHCP


 Hi all!

 I am working with asterisk and trying to get it work in DHCP network,
 where
 asterisk gets a DHCP address as well as other computers (IP-phones). So
 far,
 I've have got asterisk working with static IP address where phones are
 getting
 their IP from DHCP server.
 Is it possible at all to phones to find asterisk server if it gets
 random IP
 address from the DHCP server also? I mean, is there some settings in
 asterisk
 how I can bypass IP settings and force it to work with dynamic IP
 address?

 Thanks for any help and guidance!



 
 This mail sent through L-secure: http://www.l-secure.net/

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Re: [Asterisk-Users] Asterisk and DHCP

[Dave Cotton]

On Fri, 2005-07-01 at 10:15 +0300, [EMAIL PROTECTED] wrote:
 Hi all!
 
 I am working with asterisk and trying to get it work in DHCP network, where
 asterisk gets a DHCP address as well as other computers (IP-phones). So far,
 I've have got asterisk working with static IP address where phones are getting
 their IP from DHCP server.
 Is it possible at all to phones to find asterisk server if it gets random IP
 address from the DHCP server also? I mean, is there some settings in asterisk
 how I can bypass IP settings and force it to work with dynamic IP address?
 
 Thanks for any help and guidance!

IMHO It's not really a good idea to have important servers on a dynamic
internal IP, what does it achieve?

Look at the manual for dhcpd and you will see that you can have IP
addresses fixed by the MAC address of the NIC, best of both worlds
because you can use the DHCP server to send name server settings but the
phones can always find * by IP.  

Otherwise welcome to the world of a full DNS server that allows dynamic
updates.

 
-- 
Dave Cotton [EMAIL PROTECTED]

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[Asterisk-Users] Attended transfer works for caller, not for callee

[Younger Wang]

Hi,



I have been
trying to enable attended transfer for callee. When
the callee pressed *2, DTMF tone was heard by the
caller. But when the caller pressed *2, attended transfer started. Its
strange.



I used two
SIP phones. My Asterisk version is Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running
Linux on 2005-06-27 06:07:18.



In features.conf, I have:



[featuremap]


blindxfer = #1
; Blind transfer

disconnect = *0 ;
Disconnect

;automon =
*1
; One Touch Record

atxfer = *2
; Attended transfer



My extensions.conf is like this:



exten = _8XXX,1,Dial(SIP/${EXTEN},30,Ttm)



Another
problem is, when caller started the transfer, no dial tone is given. The log
said NOTICE[11245]: app.c:67 ast_app_dtget: Huh? no dial
for indications?.



Anybody has
the same problem as I do? BTW, can I have more precise control of transfer behavior?
If yes, will anybody show me the document?



Thank you
very much!



BR

Younger Wang










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[Asterisk-Users] Asterisk and Alcatel 4400

[Popp, Marc]

Hi there!

I wan't to connect my * to an Alcatel 4400. Does anybody have some
experiences with that? 

I have the problem that I can dial in to *, but not *-A4400 .  :-(

Thx
Marc
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[Asterisk-Users] Sometimes yes - sometimes no (dialplan)

[Ronald_Wiplinger]

I am confused about one of my installed server

The dial plan seems to be ok, but sometimes NOTHING happens if I try to 
dial an extension (from X-Lite), next time it is done.


X-Lite does not have a tone, nothing and does also have no time out. It 
seems it is not connected to the server. However, a sip show users / sip 
show peers   shows that the phone is connected.


What could be the reason?

I have installed Festiva, and was only able once to listen a text to 
speech, since then this extension number never gives me a tone. 
Sometimes it shows up in the CLI, but without a tone on the phone.

Other extensions have the same...


bye

Ronald

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[Asterisk-Users] IAX DTMF Challenges...

[Mark Edwards]

Probably been asked before, but my IAX provider assures me its not their problem

I have a IAX connection to a peer providing a DID. I am dialing up my number, seeing the DTMF tones come down the line, and the * IVR is just ignoring them.

IAX debug output is:

Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: DTMF Subclass: 1 Timestamp: 02608ms SCall: 00016 DCall: 3 [
ipaddress:4569]Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK  Timestamp: 02608ms SCall: 3 DCall: 00016 [
ipaddress:4569]
for a press of 1

I am assuming this is the DTMF inband problem, but I appear unable to convince my provider.

Can I work around this on * or do I have to go back to SIP?-- regards,Mark P. Edwards
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[Asterisk-Users] Groupcall problem

[Martin C.]

my extensions.conf

GROUPCALL = Zap/g2/12000221Zap/g2/12000222Zap/g2/12000223Zap/g2/12000224

exten = s,1,Dial(${GROUPCALL})
exten = s,2,Hangup
exten = s,102,Answer
...

The Problem is, the asterisk output and the Master.cvs tells me that the 
Group was called but not who in the group answerd the call. Ok, I can 
see which Zap channel is connected but not who belongs to the channel 
because the channel assignment must be non static.


Maybe there ist an other way to do a groupcall  ? I will be happy about 
any suggestions.


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[Asterisk-Users] E3 card

[Tamas J]

Hello!

Does anybody know what will be the around price for the announced E3
card from Digium? When is it planned to be ready?

What kind of harware can be used for a system with such card which makes
only IVR stuff?

Thanks in advance!

Kind regards,
Tamas
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Re: [Asterisk-Users] Asterisk and DHCP

[Tzafrir Cohen]

On Fri, Jul 01, 2005 at 10:15:20AM +0300, [EMAIL PROTECTED] wrote:
 Hi all!
 
 I am working with asterisk and trying to get it work in DHCP network, where
 asterisk gets a DHCP address as well as other computers (IP-phones). So far,
 I've have got asterisk working with static IP address where phones are getting
 their IP from DHCP server.
 Is it possible at all to phones to find asterisk server if it gets random IP
 address from the DHCP server also? I mean, is there some settings in asterisk
 how I can bypass IP settings and force it to work with dynamic IP address?

How exactly do you expect them to identify the asterisk server? Why
not use a host name? Distribute it by DNS or by any other means.

If you don't know exactly how, then what DHCP server are you using? I'm
currently using dnsmasq which works great for a simple dns/dhcp server
and so far was good enough for all the phones we needed to configure.

Setting a static IP is as simple as adding a line to /etc/hosts and a
line to /etc/ethers.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] Sip.conf problems

[David]

Hi,

I have been trying to configure my Asterisk to use a Sip provider for 
out and incoming calls.

I only have one user and password for connect to my sip provider.

My sip.conf is:

[general]
;disallow=gsm
;allow=ulaw
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0  ; Address to bind to
context = default   ; Default for incoming calls
callerid=No CallID
register = user:[EMAIL PROTECTED]

[sip_proxy]
type=friend
username=user
fromuser=user
secret=password
host=siprovider
dtmfmode=inband

The problem is:
If i put in the [sip_proxy] section type=friend, incoming calls doesn't 
works. If the type is set to another value (for example peer) incoming 
calls works fine, but outgoing calls doesn't works.


What can I do?

Thanks
David
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[Asterisk-Users] Ambient MD 3200 (X100P Clone)

[Sandy Thomson]

Has anyone had any success with this card?
Thank you.

Sandy.
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Re: [Asterisk-Users] Developing an application in Asterisk.

[Tzafrir Cohen]

On Thu, Jun 30, 2005 at 10:13:18AM -0500, Moises Silva wrote:
 Hi Bharat. I think that does not exists such a thing like an Asterisk
 Dev App howto :p , so for now the best way to learn i think is check
 out the apps/ directory on Asterisk Sources. Also check the app.h file
 in includes/
 
 in case you were wondering, i havent done any Asterisk App, just
 modified a couple of thinks to app_voicemail to suite my needs.
 
 May be you can get more help in asterisk-dev list

But he has asked there already, and was answered. As for IRC, I suppose
it's difficult not to stumble upon
http://asterisk.org/index.php?menu=support#irc . Though for a beginner
I'd recommend ChatZilla as an IRC client.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Sip.conf problems

[Mark Edwards]

Does the registration show up?

try sip show registry at the CLI

also try sip debug peer sip_proxy and post the result.

Might be able to see what's going on there...

mark

On 7/1/05, David [EMAIL PROTECTED] wrote:
Hi,I have been trying to configure my Asterisk to use a Sip provider forout and incoming calls.
I only have one user and password for connect to my sip provider.My sip.conf is:[general];disallow=gsm;allow=ulawport = 5060 ; Port to bind tobindaddr = 
0.0.0.0; Address to bind tocontext = default ; Default for incoming callscallerid=No CallIDregister = user:[EMAIL PROTECTED]
[sip_proxy]type=friendusername=userfromuser=usersecret=passwordhost=siproviderdtmfmode=inbandThe problem is:If i put in the [sip_proxy] section type=friend, incoming calls doesn't
works. If the type is set to another value (for example peer) incomingcalls works fine, but outgoing calls doesn't works.What can I do?ThanksDavid___
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To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users-- regards,
Mark P. EdwardsTEL:+61 408 601 107SKYPE: mark.p.edwards
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RE: [Asterisk-Users] voice mail problem

[Betül Gözlükoğlu]

 
  
  BudgeTone Software
  Version: 
  
  
   Program--1.0.5.11
  Bootloader--1.0.0.18 HTML--1.0.0.37
  VOC--1.0.0.6
  
 
 
  
  I set the dtmf to
  info and set via sip info on budgetone conf.page ,this time can not dial to
  internal extensions of telephony systems of the company
  
  
  
  
 












From: Doug Lytle [mailto:[EMAIL PROTECTED] 
Sent: Thursday, June 30, 2005 7:44
PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
voice mail problem





Betül Gözlükoğlu
wrote: 

Hi;

Have a BUDGETONE-100 and using it with
asteriskProblem occurs when I dial message centerMessage center
does not accept tones (password for example) that I dial,

Behaves as I do not dial any number and asks for the
password againChanged the DTMF Mode from in-audio to
RTP(RFC2833) it works but this time, dialing internal numbers

over telephony system is denied



Does anybody has any idea about correct configuration
on Asterisk or Budgetone?


What version of the firmware are you running?

Doug





Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup 
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[Asterisk-Users] astmanproxy

[Christian Lauinger]

Hello,

I want to recieve the output from astmanproxy in a php script.
Is that possible ?

I made a simple php script:

PRE
?php
$socket = fsockopen(127.0.0.1,1234, $errno, $errstr, $timeout);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: xxx\r\n);
fputs($socket, Secret: xxx\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: Show Channels\r\n\r\n);
fputs($socket, Action: Logoff\r\n\r\n);
while (!feof($socket)) {
$wrets[] = fread($socket, 8192);
}
fclose($socket);
var_dump($wrets);
?
/pre

Output in debugmode at the console is correct, but I cannot read the 
output in php.
If I use port 5038 I get the output, but I want to connect with multiple 
clients, so I should't use a direct connection to manager api, right ?


Why can't I read the output from astmanproxy ?
--

Regards


Christian Lauinger
ConOp Systems GbR
http://www.conop-systems.de
mailto:[EMAIL PROTECTED]
Gesellschafter: Madou Kono, Christian Lauinger, Andreas Roth

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Re: [Asterisk-Users] Computer to use

[Eric Wieling aka ManxPower]

Robert Goodyear wrote:

I'm sure you really only want to know about the absence of problems. 
 From watching this list for 6 months it seems the SuperMicro products 
are most lauded and have exhibited no hardware conflicts. Various votes 
on Dell products, so you're probably best to stay away, even though I've 
got five installs with TE110Ps in them that have never missed a beat -- 
Dimension boxes, not PowerEdge.


The SuperMicro Xeon board we tried failed miserably with both the T100P 
and TE110P.  It had the ServerWorks IDE Chipset, which I suspect was the 
problem.


--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
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Re: [Asterisk-Users] Outbound answer on TDM400P

[Eric Wieling aka ManxPower]

Me wrote:

How come an outgoing call using my TDM400P immediately
say the call is answered?  I'd like to be able to
detect when the call is actually picked up, is this
possible?

If this is normal with analog cards,


Yes


does the same thing happen with T1 cards?


No (unless you configure it for outofband)


--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
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Re: [Asterisk-Users] Ambient MD 3200 (X100P Clone)

[Tzafrir Cohen]

On Fri, Jul 01, 2005 at 11:36:31AM +0100, Sandy Thomson wrote:
 Has anyone had any success with this card?
 Thank you.

There are a number of cards based on that chipset. I've had success with
one here, FWIW.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] make error for zaptel

[Zoltan Szecsei]

Hi,
I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the 
box (in a hope to sort out the uname -r issue mentioned below).
I'm using the Asterisk Doc Proj vol 1 to guide me through the initial 
setup. I have no special HW and intend to use asterisk on an internal 
network  just to get some experience.


I have downloaded what I think I need and placed it in /usr/src (see 
listing below).
I run make clean ; make linux26 (what about the usual make with no 
parameters?) and I get a crash.


Note that uname -r returns a *different* version of what the linux is 
linked to (thanks to YOU??)


I have tried make clean ; make (no params) and it still crashes.

Can anyone offer me some suggestions? - or do I go first to the SuSE 
list to sort out the uname -r  usr/src/linux issue?


TIA,
Zoltan.

gl0:/usr/src # ls -la
total 499
drwxr-xr-x  17 root root680 2005-06-30 14:46 .
drwxr-xr-x  13 root root368 2005-06-30 09:21 ..
drwxr-xr-x   3 root root320 2005-05-19 06:53 astcc
-rw-r--r--   1 root root 130943 2005-06-25 19:09 astcc.tar
drwxr-xr-x  22 root root   2336 2005-06-23 04:16 asterisk-1.0.8
drwxr-xr-x   7 root root432 2005-06-23 04:21 asterisk-addons-1.0.8
drwxr-xr-x   3 root root184 2005-06-23 04:23 asterisk-sounds-1.0.8
drwxr-xr-x   2 root root440 2005-05-23 06:47 btp
-rw-r--r--   1 root root  32975 2005-06-25 19:08 btp.tar
drwxr-xr-x  23 root root776 2005-06-12 17:24 dicts
drwxr-xr-x   5 root root416 2005-04-01 18:50 gastman
-rw-r--r--   1 root root 332857 2005-06-25 19:08 gastman.tar
drwxr-xr-x  25 root root736 2005-06-12 17:29 kernel-modules
drwxr-xr-x   2 root root520 2005-06-23 04:11 libpri-1.0.8
lrwxrwxrwx   1 root root 19 2005-06-25 22:01 linux - 
linux-2.6.11.4-21.7

drwxr-xr-x   3 root root 72 2005-06-25 22:01 linux-2.6.11.4-20a
drwxr-xr-x   3 root root 72 2005-03-24 02:03 linux-2.6.11.4-20a-obj
drwxr-xr-x  19 root root720 2005-06-25 22:01 linux-2.6.11.4-21.7
drwxr-xr-x   3 root root 72 2005-06-02 16:54 linux-2.6.11.4-21.7-obj
lrwxrwxrwx   1 root root 23 2005-06-25 22:01 linux-obj - 
linux-2.6.11.4-21.7-obj

drwxr-xr-x   7 root root168 2005-06-12 17:43 packages
drwxr-xr-x   2 root root   2720 2005-07-01 12:43 zaptel-1.0.8
gl0:/usr/src # uname -r
2.6.11.4-20a-smp
gl0:/usr/src # cd zaptel-1.0.8/
gl0:/usr/src/zaptel-1.0.8 # make clean
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo
rm -f *.ko *.mod.c .*o.cmd
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f core
gl0:/usr/src/zaptel-1.0.8 # make linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o gendigits.o gendigits.c

cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c   -o makefw

./makefw tormenta2.rbt tor2fw  tor2fw.h
Loaded 69900 bytes from file
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo 
zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo 
tonezone.c

ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o -lm -L. libtonezone.a
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o torisatool.o torisatool.c

cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o ztmonitor.o ztmonitor.c

cc -o ztmonitor ztmonitor.o
cc -c ztspeed.c
cc -o ztspeed ztspeed.o
make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel-1.0.8 modules
make[1]: Entering directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp'
make[1]: *** No rule to make target `modules'.  Stop.
make[1]: Leaving directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp'
make: *** [linux26] Error 2
gl0:/usr/src/zaptel-1.0.8 #

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Re: [Asterisk-Users] voice mail problem

[Doug Lytle]

Betül Gözlükoğlu wrote:


*I set the dtmf to info and set via sip info on budgetone conf.page
,this time can not dial to internal extensions of telephony systems of
the company*

Install version 1.0.6.6, you can get it at: http://gs-firmware.gratissip.dk/

Make sure you are set to:

/Send DTMF:   / via RTP (RFC2833)
/Early Dial: /   No
/Silence Suppression: /   No
/SIP Registration: /   Yes

Doug

**

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Re: [Asterisk-Users] Ambient MD 3200 (X100P Clone)

[Sandy Thomson]

Tzafrir Cohen wrote:

On Fri, Jul 01, 2005 at 11:36:31AM +0100, Sandy Thomson wrote:
  

Has anyone had any success with this card?
Thank you.



There are a number of cards based on that chipset. I've had success with
one here, FWIW.
  


Thanks, I assume these 4 cards dont work then. The only unique thing I
have on these cards is the code:
1700201021110 V1.1

I googled for this and all I found was some spanish forum with people
complaining about how the card didn't work and something about
expensiveness :-)

The card gets detected  configured by the zaptel drivers fine, but the
code is 'red'. I have no idea what this means but i assume code red is
bad! :-)

Sandy.
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RE: [Asterisk-Users] make error for zaptel

[Terry Wade]

Ln -s /lib/modules/'uname -r'/build /usr/src/linux-2.6

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei
Sent: 01 July 2005 01:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] make error for zaptel

Hi,
I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the 
box (in a hope to sort out the uname -r issue mentioned below).
I'm using the Asterisk Doc Proj vol 1 to guide me through the initial 
setup. I have no special HW and intend to use asterisk on an internal 
network  just to get some experience.

I have downloaded what I think I need and placed it in /usr/src (see 
listing below).
I run make clean ; make linux26 (what about the usual make with no 
parameters?) and I get a crash.

Note that uname -r returns a *different* version of what the linux is 
linked to (thanks to YOU??)

I have tried make clean ; make (no params) and it still crashes.

Can anyone offer me some suggestions? - or do I go first to the SuSE 
list to sort out the uname -r  usr/src/linux issue?

TIA,
Zoltan.

gl0:/usr/src # ls -la
total 499
drwxr-xr-x  17 root root680 2005-06-30 14:46 .
drwxr-xr-x  13 root root368 2005-06-30 09:21 ..
drwxr-xr-x   3 root root320 2005-05-19 06:53 astcc
-rw-r--r--   1 root root 130943 2005-06-25 19:09 astcc.tar
drwxr-xr-x  22 root root   2336 2005-06-23 04:16 asterisk-1.0.8
drwxr-xr-x   7 root root432 2005-06-23 04:21 asterisk-addons-1.0.8
drwxr-xr-x   3 root root184 2005-06-23 04:23 asterisk-sounds-1.0.8
drwxr-xr-x   2 root root440 2005-05-23 06:47 btp
-rw-r--r--   1 root root  32975 2005-06-25 19:08 btp.tar
drwxr-xr-x  23 root root776 2005-06-12 17:24 dicts
drwxr-xr-x   5 root root416 2005-04-01 18:50 gastman
-rw-r--r--   1 root root 332857 2005-06-25 19:08 gastman.tar
drwxr-xr-x  25 root root736 2005-06-12 17:29 kernel-modules
drwxr-xr-x   2 root root520 2005-06-23 04:11 libpri-1.0.8
lrwxrwxrwx   1 root root 19 2005-06-25 22:01 linux - 
linux-2.6.11.4-21.7
drwxr-xr-x   3 root root 72 2005-06-25 22:01 linux-2.6.11.4-20a
drwxr-xr-x   3 root root 72 2005-03-24 02:03 linux-2.6.11.4-20a-obj
drwxr-xr-x  19 root root720 2005-06-25 22:01 linux-2.6.11.4-21.7
drwxr-xr-x   3 root root 72 2005-06-02 16:54 linux-2.6.11.4-21.7-obj
lrwxrwxrwx   1 root root 23 2005-06-25 22:01 linux-obj - 
linux-2.6.11.4-21.7-obj
drwxr-xr-x   7 root root168 2005-06-12 17:43 packages
drwxr-xr-x   2 root root   2720 2005-07-01 12:43 zaptel-1.0.8
gl0:/usr/src # uname -r
2.6.11.4-20a-smp
gl0:/usr/src # cd zaptel-1.0.8/
gl0:/usr/src/zaptel-1.0.8 # make clean
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo
rm -f *.ko *.mod.c .*o.cmd
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f core
gl0:/usr/src/zaptel-1.0.8 # make linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c   -o makefw
./makefw tormenta2.rbt tor2fw  tor2fw.h
Loaded 69900 bytes from file
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo 
zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo 
tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o -lm -L. libtonezone.a
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o torisatool.o torisatool.c
cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o ztmonitor.o ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -c ztspeed.c
cc -o ztspeed ztspeed.o
make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel-1.0.8 modules
make[1]: Entering directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp'
make[1]: *** No rule to make target `modules'.  Stop.
make[1]: Leaving directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp'
make: *** [linux26] Error 2
gl0:/usr/src/zaptel-1.0.8 #

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[Asterisk-Users] no voice

[Ritesh Jalan]

Hi All

We are unable to hear any voice where as in tcpdum it 
shows that RTP is flowing both ways

ERROR CONDITION- Executing 
Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- 
Called 2000 -- SIP/2000-0ead is 
ringing -- SIP/2000-0ead answered 
SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 
and SIP/2000-0eadHave searched web and archive w/o good 
results.Thks in advance for any 
help,sip.conf[general]port = 5060 bindaddr = 
0.0.0.0 allow=allcontext = bogon-calls externip = nn.nnn.nnn.nnn	: 
Behind router, but External static 
IPnat=yes[2000]type=friendusername=2000secret=2000host=dynamiccontext=from-sip 
mailbox=2000[2001]type=friendusername=2001secret=2001host=dynamiccontext=from-sip 
mailbox=2001;Also had some of these included, but don't 
understand;nat=yes	; have in [general] as seems to be 
req'd;reinvite=no ;canreinvite=no ;qualify=1000 ;disallow=all 
;allow=gsm ;allow=ulaw ;allow=alaw 
extensions.conf---[general]static=yes 
writeprotect=yes[bogon-calls]exten = _.,1,Congestion 
[from-sip];; Number 2000 - Dave Laptop #1;exten = 
2000,1,Dial(SIP/2000,20)exten = 2000,2,Voicemail(u2000)exten = 
2000,102,Voicemail(b2000)exten = 2000,103,Hangup;; Number 2001 - 
Dave Laptop #2;exten = 2001,1,Dial(SIP/2001,20)exten = 
2001,2,Voicemail(u2001)exten = 2001,102,Voicemail(b2001)exten = 
2001,103,Hangup
Thanks  Regards Ritesh Jalan Senior 
Engineer - Test  Audit Net4India Ltd. 703 Bikaji Cama Bhawan 11 
Bikaji Cama Place New Delhi - 110029 Ph: +91-11-26160129 ext. 131 
Cell : +91-9818616329Web site: http://www.net4india.com 
== 
This message may contain confidential and/or privileged information. If you 
are not the addressee or authorized to receive this for the addressee, you must 
not use, copy, disclose or take any action based on this message or any 
information herein. If you have received this message in error, please advise 
the sender immediately by reply e-mail and delete this message. Thank you for 
your cooperation. 
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Re: [Asterisk-Users] no voice

[Eric Wieling aka ManxPower]

Ritesh Jalan wrote:

Hi All

We are unable to hear any voice where as in tcpdum it shows that RTP is flowing 
both ways

ERROR CONDITION
---
-- Executing Dial(SIP/2001-f6c4, SIP/2000|20) in new stack
-- Called 2000
-- SIP/2000-0ead is ringing
-- SIP/2000-0ead answered SIP/2001-f6c4
-- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead

Have searched web and archive w/o good results.

Thks in advance for any help,

sip.conf

[general]
port = 5060 
bindaddr = 0.0.0.0 
allow=all
context = bogon-calls 
externip = nn.nnn.nnn.nnn	: Behind router, but External static IP

nat=yes


allow=all will do that.  Don't use allow=all.

--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
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[Asterisk-Users] Voicemail storage

[Alberto Risco]

Hi all,



Currently when someone leaves a voicemail the message is
stored in the /var/spool/asterisk/voicemail/default/(users ext)/INBOX ,
as it should. However Ive noticed that a copy is placed in the
/tmp directory. Once a message is heard and deleted, the copy in /tmp
remains. My question is why?, for how long? And is there a way to modify a
config file to send it somewhere else?. I would like to write a script to
remove files in the /tmp directory periodically but need to understand why
voicemail files are stored here first.



I am running asterisk 1.0.8 on a Dell 1850.



Thanks,



Alberto







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RE: [Asterisk-Users] wi-fi phone advice

[Huddleston, Robert]

Do you know where to get one of these?
 
 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 chawki hammoud
 Sent: Thursday, June 30, 2005 4:35 PM
 To: Asterisk-Users@lists.digium.com
 Subject: [Asterisk-Users] wi-fi phone advice
 
 Hi:
 
 I want to connect a wi-fi phone to my Asterisk box through a 
 wi-fi AP so I can make voip calls.
  
 please send me your recomendation about what wi-fi phone I 
 should be looking for. Anybody tried the
 HOP1502 Wi-Fi IP phone. Its listed price $39.
 
 Regards;
 Chawki
 
 
   
 
 Yahoo! Sports
 Rekindle the Rivalries. Sign up for Fantasy Football 
 http://football.fantasysports.yahoo.com
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Robert A. Huddleston.vcf
Description: Binary data
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RE : [Asterisk-Users] Ambient MD 3200 (X100P Clone)

[f6hqz-m]

Yes, I have  :-)
3 of this cards running well on my personnal *
What price for your ?

Best Regards,
Francois BERGERET,
France.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Sandy Thomson
Envoyé : vendredi 1 juillet 2005 12:37
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Ambient MD 3200 (X100P Clone)


Has anyone had any success with this card?
Thank you.

Sandy.
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Re: [Asterisk-Users] make error for zaptel

[Bob Goddard]

On Friday 01 Jul 2005 13:08, Terry Wade wrote:
 Ln -s /lib/modules/'uname -r'/build /usr/src/linux-2.6

Nope, I doubt that. The end user should read /usr/src/linux/README.suse
and see how to prepare the kernel for building thirparty modules.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan
 Szecsei Sent: 01 July 2005 01:51 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] make error for zaptel

 Hi,
 I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the
 box (in a hope to sort out the uname -r issue mentioned below).
 I'm using the Asterisk Doc Proj vol 1 to guide me through the initial
 setup. I have no special HW and intend to use asterisk on an internal
 network  just to get some experience.

 I have downloaded what I think I need and placed it in /usr/src (see
 listing below).
 I run make clean ; make linux26 (what about the usual make with no
 parameters?) and I get a crash.

 Note that uname -r returns a *different* version of what the linux is
 linked to (thanks to YOU??)

It looks like you updated the kernel but never rebooted.

 I have tried make clean ; make (no params) and it still crashes.

 Can anyone offer me some suggestions? - or do I go first to the SuSE
 list to sort out the uname -r  usr/src/linux issue?
[...]
 make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel-1.0.8 modules
 make[1]: Entering directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp'
 make[1]: *** No rule to make target `modules'.  Stop.
 make[1]: Leaving directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp'
 make: *** [linux26] Error 2
 gl0:/usr/src/zaptel-1.0.8 #
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Re: [Asterisk-Users] E3 card

[Kevin P. Fleming]

Tamas J wrote:


Does anybody know what will be the around price for the announced E3
card from Digium? When is it planned to be ready?


Pricing and release date have not been announced at this time.
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[Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card near NH??

[Tom Rymes]

I need a Digium or Sangoma T1 card that has at least 2 spans on it
fairly quickly. Does anyone know of a vendor for either of these in NH
or Northern MA?

Please let me know!

Tom



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Re: [Asterisk-Users] Sip.conf problems

[MF Hulber]

Try two different entries:

sip.conf:

   [general]
   ;disallow=gsm
   ;allow=ulaw
   port = 5060 ; Port to bind to
   bindaddr = 0.0.0.0  ; Address to bind to
   context = default   ; Default for incoming calls
   callerid=No CallID
   register = user:[EMAIL PROTECTED]/2025551212

   [2025551212]
   type=peer
   realm=sipprovider.com
   fromdomain=sipprovider.com
   username=user
   fromuser=user
   secret=password
   host=sipprovider.com
   dtmfmode=inband

   [sip_provider]
   type=peer
   context=sip_provider-inbound
   host=sipprovider.com


extensions.conf:

   [sip_provider-inbound]

  exten = 2025551212,n,Goto(default,s,1)

  exten = i,1,Goto(default,s,1)

  exten = t,1,Goto(default,s,1)

  exten = h,1,hangup


David wrote:


Hi,

I have been trying to configure my Asterisk to use a Sip provider for 
out and incoming calls.

I only have one user and password for connect to my sip provider.

My sip.conf is:

[general]
;disallow=gsm
;allow=ulaw
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0  ; Address to bind to
context = default   ; Default for incoming calls
callerid=No CallID
register = user:[EMAIL PROTECTED]

[sip_proxy]
type=friend
username=user
fromuser=user
secret=password
host=siprovider
dtmfmode=inband

The problem is:
If i put in the [sip_proxy] section type=friend, incoming calls 
doesn't works. If the type is set to another value (for example peer) 
incoming calls works fine, but outgoing calls doesn't works.


What can I do?

Thanks
David
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RE: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card near NH??

[Max W Blackmer Jr]

http://www.voipsupply.com
or call at
1-800-398-VOIP
they can rush deliver if you need it.

  Original Message 
 Subject: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card
 near NH??
 From: Tom Rymes [EMAIL PROTECTED]
 Date: Fri, July 01, 2005 8:31 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com

 I need a Digium or Sangoma T1 card that has at least 2 spans on it
 fairly quickly. Does anyone know of a vendor for either of these in NH
 or Northern MA?

 Please let me know!

 Tom



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Re: [Asterisk-Users] Ambient MD 3200 (X100P Clone)

[John Novack]

Sandy Thomson wrote:


Tzafrir Cohen wrote:

 


On Fri, Jul 01, 2005 at 11:36:31AM +0100, Sandy Thomson wrote:


   


Has anyone had any success with this card?
Thank you.
  

 


There are a number of cards based on that chipset. I've had success with
one here, FWIW.


   



Thanks, I assume these 4 cards dont work then. The only unique thing I
have on these cards is the code:
1700201021110 V1.1

I googled for this and all I found was some spanish forum with people
complaining about how the card didn't work and something about
expensiveness :-)

The card gets detected  configured by the zaptel drivers fine, but the
code is 'red'. I have no idea what this means but i assume code red is
bad! :-)

Sandy.
 


Red usually means that it doesn't see battery on the line input.
Do you have a station line connected, either from the local phone 
company, another PBX with a POTS port or ??


Conceptually this card replaces a Plain Old Telephone Set.

Be advised that even though Zaptel knows there is nothing connected, 
Asterisk doesn't, and will blind dial and not detect Dial Tone.
The clone cards seem to work OK in the US or elsewhere that there are 
600 ohm subscriber lines, but reportedly in the UK and elsewhere  echo 
can be a problem.


John Novack

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Re: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card near NH??

[[EMAIL PROTECTED]]

There's also http://www.atacomm.com.



On 7/1/05, Max W Blackmer Jr [EMAIL PROTECTED] wrote:
 http://www.voipsupply.com
 or call at
 1-800-398-VOIP
 they can rush deliver if you need it.
 
   Original Message 
  Subject: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card
  near NH??
  From: Tom Rymes [EMAIL PROTECTED]
  Date: Fri, July 01, 2005 8:31 am
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
 
  I need a Digium or Sangoma T1 card that has at least 2 spans on it
  fairly quickly. Does anyone know of a vendor for either of these in NH
  or Northern MA?
 
  Please let me know!
 
  Tom
 
 
 
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RE: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card near NH??

[mattf]

Hello,

Either Digium or Sangoma can overnight a card to you. As the car drives you
could go to Toronto and pickup a card from Sangoma if you needed if a few
hours before Overnight would deliver it.

There are also a lot of resellers that can overnight to you as well.

MATT---

-Original Message-
From: Tom Rymes [mailto:[EMAIL PROTECTED]
Sent: Friday, July 01, 2005 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card
near NH??


I need a Digium or Sangoma T1 card that has at least 2 spans on it
fairly quickly. Does anyone know of a vendor for either of these in NH
or Northern MA?

Please let me know!

Tom



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Re: [Asterisk-Users] Flash and zap and # key

[Seth Remington]

On Fri, 2005-07-01 at 08:01 +0200, Wilson Pickett wrote:
  I'm not sure if this is supposed to happen, but when I press the # key it
  seems to have the effect of flashing the hook, or at least letting me
  transfer.  I am using Zap hardware.
 Do you hear a transfer voice prompt? Asterisk will intercept the #
 key if told to do so in the Dial application. In STABLE this is
 hard-coded I think.
 
 show application dial will give the details of how T and t can be used
 as options.
 
 If your dial commands are not using T and t as options, I'm not sure
 what is happening.

If you are running CVS HEAD instead of stable you can set the transfer
key(s) in features.conf. There is also a double ## patch floating around
somewhere for STABLE I think.

-Seth

-- 
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559

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[Asterisk-Users] Problems loading asterisk .

[Bharat M. Sarvan]

Hello everybody,

 I have made a application of my
own. (I.e. Def ( )). I am able to compile the application successfully. And the
.so file is created as well. But when I load asterisk I get the following
error.





[Def.so]Jul 1 19:20:06 WARNING[15664]: loader.c:295
ast_load_resource: No key routine in module /usr/lib/asterisk/modules/Def.so

Jul 1 19:20:06 WARNING[15664]: loader.c:302
ast_load_resource: Key routine returned NULL in module
/usr/lib/asterisk/modules/Def.so

Jul 1 19:20:06 WARNING[15664]: loader.c:311
ast_load_resource: 2 error(s) loading module /usr/lib/asterisk/modules/Def.so,
aborted

Jul 1 19:20:06 WARNING[15664]: loader.c:440 load_modules:
Loading module Def.so failed!







So if anybody could help me out as to where must I be going
wrong, it would be very kind of you.









Regards,

Bharat M. Sarvan








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[Asterisk-Users] Call Transfer Problem

[Adam Robins]

I do not want to use the default key of '#' for call transfer, because
as we all know, it interferes with many IVRs that require # as a
termination character.  I modified features.conf and added:

[featuremap]
atxfer = **

The double-star now works great.  If I press it while on a call, I go
into transfer mode.  The problem is that the # still works as well!
Shouldn't the atzfer specification turn off the #?

Any insight would be appreciated.

Thanks,
Adam

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This transmission is sent in trust, for the sole purpose of delivery to the 
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[Asterisk-Users] how to PortaOne's Radius client for asterisk

[Kamran Ahmad]

hello

i am trying to follow 
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth

can any one tell how to install this 

2. Install Asterisk::AGI and Asterisk::Manager
(unfortunately it is not on CPAN yet!) 

thanks in advance
Kamran



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RE : [Asterisk-Users] Ambient MD 3200 (X100P Clone)

[f6hqz-m]

Hi men,

If you unplug a telephone line behind X100P/X101P, you must have an alarm
about this Zaptel device on the Asterisk console.
When you plug the line, you can see alarm stop on a new line (I don't
remember the exact message).
Very nice test.

Best Regards,
Francois BERGERET,
France.

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de John Novack
Envoyé : vendredi 1 juillet 2005 15:55
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] Ambient MD 3200 (X100P Clone)


Sandy Thomson wrote:

Tzafrir Cohen wrote:

  

On Fri, Jul 01, 2005 at 11:36:31AM +0100, Sandy Thomson wrote:
 



Has anyone had any success with this card?
Thank you.
   

  

There are a number of cards based on that chipset. I've had success 
with one here, FWIW.
 




Thanks, I assume these 4 cards dont work then. The only unique thing I 
have on these cards is the code: 1700201021110 V1.1

I googled for this and all I found was some spanish forum with people 
complaining about how the card didn't work and something about 
expensiveness :-)

The card gets detected  configured by the zaptel drivers fine, but the 
code is 'red'. I have no idea what this means but i assume code red is 
bad! :-)

Sandy.
  

Red usually means that it doesn't see battery on the line input. Do you have
a station line connected, either from the local phone 
company, another PBX with a POTS port or ??

Conceptually this card replaces a Plain Old Telephone Set.

Be advised that even though Zaptel knows there is nothing connected, 
Asterisk doesn't, and will blind dial and not detect Dial Tone. The clone
cards seem to work OK in the US or elsewhere that there are 
600 ohm subscriber lines, but reportedly in the UK and elsewhere  echo 
can be a problem.

John Novack

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Re: [Asterisk-Users] make error for zaptel

[Zoltan Szecsei]

Hi Bob,
Thanks - I'll run with the README idea of yours.
Your comment regarding re-boot however is not valid. I also thought that 
was the case and (as I said on the first line of my message) I 
specifically rebooted the box. Have to confess I am really flumuxed why 
the symbolinc link differs from the uname -r name.


Thanks  chat soon,
Zoltan.


Bob Goddard wrote:


On Friday 01 Jul 2005 13:08, Terry Wade wrote:
 


Ln -s /lib/modules/'uname -r'/build /usr/src/linux-2.6
   



Nope, I doubt that. The end user should read /usr/src/linux/README.suse
and see how to prepare the kernel for building thirparty modules.

 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan
Szecsei Sent: 01 July 2005 01:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] make error for zaptel

Hi,
I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the
box (in a hope to sort out the uname -r issue mentioned below).
I'm using the Asterisk Doc Proj vol 1 to guide me through the initial
setup. I have no special HW and intend to use asterisk on an internal
network  just to get some experience.

I have downloaded what I think I need and placed it in /usr/src (see
listing below).
I run make clean ; make linux26 (what about the usual make with no
parameters?) and I get a crash.

Note that uname -r returns a *different* version of what the linux is
linked to (thanks to YOU??)
   



It looks like you updated the kernel but never rebooted.

 


I have tried make clean ; make (no params) and it still crashes.

Can anyone offer me some suggestions? - or do I go first to the SuSE
list to sort out the uname -r  usr/src/linux issue?
   


[...]
 


make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel-1.0.8 modules
make[1]: Entering directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp'
make[1]: *** No rule to make target `modules'.  Stop.
make[1]: Leaving directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp'
make: *** [linux26] Error 2
gl0:/usr/src/zaptel-1.0.8 #
   


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[Asterisk-Users] Catch Autodial failure

[Chris Douglas]

Hello,
 I am 
using asterisk to autodial a phone number once an hour to verify an answering 
IVR scriptis working by listening for a beep played by the IVR, and that 
is working well. The one thing I have not been able to find is a way to 
trap if the call didn't connect at all, like if the entire IVR is down or the 
phone line is dead, since it doesnt enter the menu until the call is connected. 


Here is my dialout 
script that gets dumped into outgoing. Is there a way to tell it to run a 
script if it fails to connect?

Channel: SIP/[EMAIL PROTECTED]Callerid: 
1000MaxRetries: 1RetryTime: 300WaitTime: 15Context: 
IVR_TEST1Extension: sPriority: 1

Thank 
you
Chris
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Re: [Asterisk-Users] wi-fi phone advice

[Cory Andrews]

This unit is vaporware from what I can tell.

Cory Andrews
Purchasing / EVP
VOIPSupply.com
v – 716.630.1555 X22
e – [EMAIL PROTECTED]



Huddleston, Robert wrote:


Do you know where to get one of these?



 


-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
chawki hammoud

Sent: Thursday, June 30, 2005 4:35 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] wi-fi phone advice

Hi:

I want to connect a wi-fi phone to my Asterisk box through a 
wi-fi AP so I can make voip calls.


please send me your recomendation about what wi-fi phone I 
should be looking for. Anybody tried the

HOP1502 Wi-Fi IP phone. Its listed price $39.

Regards;
Chawki




Yahoo! Sports
Rekindle the Rivalries. Sign up for Fantasy Football 
http://football.fantasysports.yahoo.com

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[Asterisk-Users] voicemail and mysql

[Francesco Castellano]

Dear list members,
I am trying to use mysql for the mailbox definitions of the voicemail
system. All works fine, but my asterisk does not catch the options of
the mailbox. In particular, 'attach=no' has no effect. Here is my test
table:

mysql select * from users where email='[EMAIL PROTECTED]';
+-+-+--+--+---+---+++
| context | mailbox | password | fullname | email
| pager | options| stamp  |
+-+-+--+--+---+---+++
| default | 212 | 1978 | Francesco Castellano |
[EMAIL PROTECTED] |  | tz=italy|attach=no | 20050624171320 |
+-+-+--+--+---+---+++
1 row in set (0.00 sec)

Even if the only option is 'attach=no', it doesn't work.

And then I have a couple of question on voicemail: when asterisk reads
the mailbox definitions? At startup? Is it enough a reload to reread
the definitions, or is it necessary a restart?

Finally, is it possible to use at the same time some mailbox from mysql
and some from voicemail.conf? At cli prompt show voicemail users
doesn't work for mysql-defined mailboxes?

I've not found any documentation on these issues, and I have some
difficulties to get informations from the source code.

Any suggestions are welcomed!
Thanks, bye
Francesco

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Re: [Asterisk-Users] Ambient MD 3200 (X100P Clone)

[Richard Malcolm-Smith]

Sandy Thomson wrote:

Has anyone had any success with this card?
Thank you.


I am looking for a source for the clones in NZ - getting the real deal here isnt 
an option (killer shipping) and at the moment I am just having a play with 
asterisk and have given up on the internet linejacks I rescued from the skip at 
a previous employer.


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[Asterisk-Users] (en|dis)able CW within /etc/asterisk

[Tim Wesemann]

I know that CW can be turned on and off with database put CW which updates 
my /var/lib/asterisk/astdb.


However, I would also like to be able to configure this through the flat 
config files in my /etc/asterisk, is this possible? I'm using 
CVS-v1-0-06/24/05-14:34:05.


TIA for any help.

--
Tim Wesemann

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Re: [Asterisk-Users] Asterisk/SER/Call Manager

[Alistair Cunningham]

Dinesh,

This should be fine as long as you set canreinvite=no for both systems 
in sip.conf on Asterisk. I've done both Asterisk - SER and Asterisk 
- CCM, and they work well.


Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/


Dinesh wrote:

Hi all,

 

I have Asterisk talking to my call manager 4.0 with SIP trunk as 
mentioned in the wiki.  I also have SER talking to Asterisk.  I need the 
SER talking to my Call manager.  The reason why CCM cannot talk to SER 
is because SER is a on a public ip address, and CCM is on a private ip 
address. 

 

The asterisk how ever has 2 nics, which talks to both and external.  Is 
it possible to allow asterisk as a bridge between SER and call manager?  
Any thoughts on this would be great.


 


Regards,

Dinesh Birlasekaran
Network Engineer,
ComIT, Institute of Molecular and Cell Biology
61 Biopolis Drive, Singapore 138673
HP : 92962676 DID : 65869804 Fax : 67791117 Email : 
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]

WWW: www.imcb.a-star.edu.sg http://www.imcb.a-star.edu.sg

 




DISCLAIMER:
This email is confidential and may be privileged. If you are not the 
intended recipient, please delete it and notify us immediately. Please 
do not copy or use it for any purpose, or disclose its contents to any 
other person as it may be an offence under the Official Secrets Act. 
Thank you.





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Re: RE : [Asterisk-Users] Ambient MD 3200 (X100P Clone)

[Sandy Thomson]

[EMAIL PROTECTED] wrote:

Hi men,

If you unplug a telephone line behind X100P/X101P, you must have an alarm
about this Zaptel device on the Asterisk console.
When you plug the line, you can see alarm stop on a new line (I don't
remember the exact message).
Very nice test.

Best Regards,
Francois BERGERET,
France.


Thanks for all your replies guys.
After literally weeks of fiddling with this stuff, I have realised what
the problem is. The cable I was using, was wired to the two outer pins
rather than the two inner pins (or all 4 pins).
The other cable I had tried didn't work either. The main cable I was
testing with worked with a normal telephone which was really strange.

Anyway, the moral of the story is test your lines and cables before
blaming asterisk/zaptel! :-)

Sandy.
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Re: RE : [Asterisk-Users] Ambient MD 3200 (X100P Clone)

[VoIP Newbie]

My one from www.broad-tel.com works fine and is very cheap.

On 7/1/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 Yes, I have  :-)
 3 of this cards running well on my personnal *
 What price for your ?
 
 Best Regards,
 Francois BERGERET,
 France.
 
 
 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De la part de Sandy Thomson
 Envoyé : vendredi 1 juillet 2005 12:37
 À : asterisk-users@lists.digium.com
 Objet : [Asterisk-Users] Ambient MD 3200 (X100P Clone)
 
 
 Has anyone had any success with this card?
 Thank you.
 
 Sandy.
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RE: [Asterisk-Users] wi-fi phone advice

[Huddleston, Robert]

Well poo - if I can use that word I'm one of those poor family guys who 
loves to buy hardware on the cheap =)

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Cory Andrews
 Sent: Friday, July 01, 2005 10:26 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] wi-fi phone advice
 
 This unit is vaporware from what I can tell.
 
 Cory Andrews
 Purchasing / EVP
 VOIPSupply.com
 v - 716.630.1555 X22
 e - [EMAIL PROTECTED]
 
 
 
 Huddleston, Robert wrote:
 
 Do you know where to get one of these?
  
  
 
   
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf 
 Of chawki 
 hammoud
 Sent: Thursday, June 30, 2005 4:35 PM
 To: Asterisk-Users@lists.digium.com
 Subject: [Asterisk-Users] wi-fi phone advice
 
 Hi:
 
 I want to connect a wi-fi phone to my Asterisk box through 
 a wi-fi AP 
 so I can make voip calls.
  
 please send me your recomendation about what wi-fi phone I 
 should be 
 looking for. Anybody tried the
 HOP1502 Wi-Fi IP phone. Its listed price $39.
 
 Regards;
 Chawki
 
 
 
 
 Yahoo! Sports
 Rekindle the Rivalries. Sign up for Fantasy Football 
 http://football.fantasysports.yahoo.com
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 -
 --
 -
 
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Re: [Asterisk-Users] wi-fi phone advice

[Cory Andrews]

Robert - I suspect what they are doing is just trying to build buzz 
and simply not mentioning the fine print here. There is no way, given 
manufacturing and importation costs, that I can believe they can build a 
solvient business offering around a $39 WLAN phone selling just the 
hardware, that is just too low a price point.


If this ever does come to market, it will be tied to a service contract, 
essentially a locked unit that can only be used with their affiliated 
providers. That would be the only way for them to recoup the device cost 
is through service revenues. Unless you want to resell their service, I 
don't see much here to get excited about.


Cory Andrews
Purchasing / EVP
VOIPSupply.com
v – 716.630.1555 X22
e – [EMAIL PROTECTED]



Huddleston, Robert wrote:


Well poo - if I can use that word I'm one of those poor family guys who 
loves to buy hardware on the cheap =)

 


-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Cory Andrews

Sent: Friday, July 01, 2005 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] wi-fi phone advice

This unit is vaporware from what I can tell.

Cory Andrews
Purchasing / EVP
VOIPSupply.com
v - 716.630.1555 X22
e - [EMAIL PROTECTED]



Huddleston, Robert wrote:

   


Do you know where to get one of these?





 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf 
   

Of chawki 
   


hammoud
Sent: Thursday, June 30, 2005 4:35 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] wi-fi phone advice

Hi:

I want to connect a wi-fi phone to my Asterisk box through 
   

a wi-fi AP 
   


so I can make voip calls.

please send me your recomendation about what wi-fi phone I 
   

should be 
   


looking for. Anybody tried the
HOP1502 Wi-Fi IP phone. Its listed price $39.

Regards;
Chawki




Yahoo! Sports
Rekindle the Rivalries. Sign up for Fantasy Football 
http://football.fantasysports.yahoo.com

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-
 


--
   


-

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Re: [Asterisk-Users] wi-fi phone advice

[Richard Malcolm-Smith]

If it does materialize, im up for 3 or 4 of them at that price.

Huddleston, Robert wrote:

Well poo - if I can use that word I'm one of those poor family guys who 
loves to buy hardware on the cheap =)




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Re: [Asterisk-Users] make error for zaptel

[Bob Goddard]

On Friday 01 Jul 2005 15:14, Zoltan Szecsei wrote:
 Hi Bob,
 Thanks - I'll run with the README idea of yours.
 Your comment regarding re-boot however is not valid. I also thought that
 was the case and (as I said on the first line of my message) I
 specifically rebooted the box. Have to confess I am really flumuxed why
 the symbolinc link differs from the uname -r name.

I cannot see what the problem is with the output of 'uname -r'!

If you are saying that you are not running linux-2.6.11.4-20a, then
I would say you. Perhaps lilo or grub got corrupted. You should be
checking the layout of /boot at least.

 Bob Goddard wrote:
 On Friday 01 Jul 2005 13:08, Terry Wade wrote:
 Ln -s /lib/modules/'uname -r'/build /usr/src/linux-2.6
 
 Nope, I doubt that. The end user should read /usr/src/linux/README.suse
 and see how to prepare the kernel for building thirparty modules.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan
 Szecsei Sent: 01 July 2005 01:51 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] make error for zaptel
 
 Hi,
 I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the
 box (in a hope to sort out the uname -r issue mentioned below).
 I'm using the Asterisk Doc Proj vol 1 to guide me through the initial
 setup. I have no special HW and intend to use asterisk on an internal
 network  just to get some experience.
 
 I have downloaded what I think I need and placed it in /usr/src (see
 listing below).
 I run make clean ; make linux26 (what about the usual make with no
 parameters?) and I get a crash.
 
 Note that uname -r returns a *different* version of what the linux is
 linked to (thanks to YOU??)
 
 It looks like you updated the kernel but never rebooted.
 
 I have tried make clean ; make (no params) and it still crashes.
 
 Can anyone offer me some suggestions? - or do I go first to the SuSE
 list to sort out the uname -r  usr/src/linux issue?
 
 [...]
 
 make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel-1.0.8
  modules make[1]: Entering directory
  `/usr/src/linux-2.6.11.4-20a-obj/i386/smp' make[1]: *** No rule to make
  target `modules'.  Stop.
 make[1]: Leaving directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp'
 make: *** [linux26] Error 2
 gl0:/usr/src/zaptel-1.0.8 #
[ Oh for fsck sake, can't people delete old signatures ]
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Re: [Asterisk-Users] passing through MWI info from SBC

[John Novack]

Mike Myers wrote:


Hi..  I am about to replace my aging Nortel Venture
system with an Asterisk system and 6 Polycom IP 501
phones, and a couple sipura 841's for less used areas.

We have 3 phone lines here.  One is SBC, one Vonage,
and one Voipjet...  One hangup is that I can't figure
out how to pass through a voicemail waiting indication
from SBC.  This is important because my wife and her
family all exchange voicemails with each other on the
SBC voicemail system.  They can leave messages for
each other without having the phones ring, etc...  We
have a 2 yr old at home, and her sister has some small
kids too, so that's how they manage to send voicemails
when they are unsure if the kids are sleeping, etc... 
Anyway, preserving this capability of using the SBC VM

and being notified when a message is waiting is
critical for good WAF.  


The vonage line and voipjet line can be intergrated
into the Asterisk VM.  My Nortel venture phones light
the MWI if any line has VM on it, and the display
tells you which lines have VM waiting.  I would love
to be able to duplicate this function on the Polycom's
and hopefully the Sipura's as well.

I've looked for answers on this, but haven't found
one, hence the post.  My apologies if I have missed
something.  


Thanks much,
Mike
 


You haven't missed much.
With SBC you are out of luck, since Asterisk doesn't detect dialtone  ( 
it dials blind, sometimes too quickly for the CO to catch the first 
digit, resulting in wrong numbers )) or stutter dialtone either, and 
reportedly has had any indication of the DC status of a POTS line 
removed due to problems.


Only choice would to port the number to a VOIP provider and provide the 
VM in Asterisk.

Similar problem with Vonage VM.

John Novack

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[Asterisk-Users] asterisk newbie and phones which don't want to comunicate

[Sistemista WebSolvingJaa]

Hi list, i'm an asterisk newbie and i've to setup a net with an
asterisk server and several ip phones linked on the net.
i hope my questions are IT ans if you have some link for solving those
problems please mail me.
i've wrote the sip.conf in this way:
[2011]
type=friend
username=2011
secret=1945
regexten=1234
host=dynamic
;host=192.168.100.242
permit=192.168.100.0/24
context=sip-incoming
canreinvite=yes
dtmfmode=rfc2833
nat=1

[2012]
type=friend
username=2012
secret=1945
regexten=1234
host=dynamic
;host=192.168.100.221
permit=192.168.100.0/24
context=sip-incoming
canreinvite=yes
dtmfmode=rfc2833
nat=1

and the extension.conf if quitelly the same as the original. the
phones softwares are setted up correctly, but from a phone i can't
call another phone on the net. can somebody suggest me a possible
solution?

thanks a lot
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Re: [Asterisk-Users] make error for zaptel

[Zoltan Szecsei]

Bob Goddard wrote:


On Friday 01 Jul 2005 15:14, Zoltan Szecsei wrote:
 


Hi Bob,
Thanks - I'll run with the README idea of yours.
Your comment regarding re-boot however is not valid. I also thought that
was the case and (as I said on the first line of my message) I
specifically rebooted the box. Have to confess I am really flumuxed why
the symbolinc link differs from the uname -r name.
   



I cannot see what the problem is with the output of 'uname -r'!
 

I'm saying that I though that if uname -r returns:   2.6.11.4-20a-smp 
then I would expect that /usr/src/linux would link to 
linux-2.6.11.4-20a-smp and it does not, it links to linux-2.6.11.4-21.7


see:

gl0:/usr/src # ls -la
total 499
drwxr-xr-x  17 root root680 2005-06-30 14:46 .
drwxr-xr-x  13 root root368 2005-06-30 09:21 ..
drwxr-xr-x   3 root root320 2005-05-19 06:53 astcc
-rw-r--r--   1 root root 130943 2005-06-25 19:09 astcc.tar
drwxr-xr-x  22 root root   2336 2005-06-23 04:16 asterisk-1.0.8
drwxr-xr-x   7 root root432 2005-06-23 04:21 asterisk-addons-1.0.8
drwxr-xr-x   3 root root184 2005-06-23 04:23 asterisk-sounds-1.0.8
drwxr-xr-x   2 root root440 2005-05-23 06:47 btp
-rw-r--r--   1 root root  32975 2005-06-25 19:08 btp.tar
drwxr-xr-x  23 root root776 2005-06-12 17:24 dicts
drwxr-xr-x   5 root root416 2005-04-01 18:50 gastman
-rw-r--r--   1 root root 332857 2005-06-25 19:08 gastman.tar
drwxr-xr-x  25 root root736 2005-06-12 17:29 kernel-modules
drwxr-xr-x   2 root root520 2005-06-23 04:11 libpri-1.0.8
lrwxrwxrwx   1 root root 19 2005-06-25 22:01 linux - 
linux-2.6.11.4-21.7

drwxr-xr-x   3 root root 72 2005-06-25 22:01 linux-2.6.11.4-20a
drwxr-xr-x   3 root root 72 2005-03-24 02:03 linux-2.6.11.4-20a-obj
drwxr-xr-x  19 root root720 2005-06-25 22:01 linux-2.6.11.4-21.7
drwxr-xr-x   3 root root 72 2005-06-02 16:54 linux-2.6.11.4-21.7-obj
lrwxrwxrwx   1 root root 23 2005-06-25 22:01 linux-obj - 
linux-2.6.11.4-21.7-obj

drwxr-xr-x   7 root root168 2005-06-12 17:43 packages
drwxr-xr-x   2 root root   2720 2005-07-01 12:43 zaptel-1.0.8
gl0:/usr/src # uname -r
2.6.11.4-20a-smp


So I figured that this may be the reason why the zaptel make is failing.

Zoltan



If you are saying that you are not running linux-2.6.11.4-20a, then
I would say you. Perhaps lilo or grub got corrupted. You should be
checking the layout of /boot at least.

 


Bob Goddard wrote:
   


On Friday 01 Jul 2005 13:08, Terry Wade wrote:
 


Ln -s /lib/modules/'uname -r'/build /usr/src/linux-2.6
   


Nope, I doubt that. The end user should read /usr/src/linux/README.suse
and see how to prepare the kernel for building thirparty modules.

 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan
Szecsei Sent: 01 July 2005 01:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] make error for zaptel

Hi,
I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the
box (in a hope to sort out the uname -r issue mentioned below).
I'm using the Asterisk Doc Proj vol 1 to guide me through the initial
setup. I have no special HW and intend to use asterisk on an internal
network  just to get some experience.

I have downloaded what I think I need and placed it in /usr/src (see
listing below).
I run make clean ; make linux26 (what about the usual make with no
parameters?) and I get a crash.

Note that uname -r returns a *different* version of what the linux is
linked to (thanks to YOU??)
   


It looks like you updated the kernel but never rebooted.

 


I have tried make clean ; make (no params) and it still crashes.

Can anyone offer me some suggestions? - or do I go first to the SuSE
list to sort out the uname -r  usr/src/linux issue?
   


[...]

 


make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel-1.0.8
modules make[1]: Entering directory
`/usr/src/linux-2.6.11.4-20a-obj/i386/smp' make[1]: *** No rule to make
target `modules'.  Stop.
make[1]: Leaving directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp'
make: *** [linux26] Error 2
gl0:/usr/src/zaptel-1.0.8 #
   


[ Oh for fsck sake, can't people delete old signatures ]
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[Asterisk-Users] Problem with DTFM and complex international setup

[Rob Scott]

We have some guys working in the US who can't always dial back to our
company in Europe easily (lots of clients require authorization to make
international calls), so I set up the following:

   - ipkall.com number links to a FWD number
   - office Asterisk box registers with FWD

Then I programmed Asterisk to accept office extension number using DTFM
tones.
This works OK.

Then I programmed Asterisk so that it is possible, using a PIN code, to
dial out from Asterisk onto the local PSTN.

This also works occasionally.
Looking at the message from the Asterisk box it is clear that sometimes
numbers are missed or repeated in the dial string. This I suspect is
because Asterisk is listening to the DTMF tones but the signal is
dropped; sometimes the drop means that a whole digit is dropped and
sometimes is means that a digit is repeated.

Does anyone know how I can fix this to make it more reliable
(out-of-band DTMF?) or a better way to achieve a reliable setup?
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Re: [Asterisk-Users] Problems loading asterisk .

[Moises Silva]

may be reading include/asterisk/module.h you can find the answer. May be this?

/*! Returns the ASTERISK_GPL_KEY */
/*!
 * This returns the ASTERISK_GPL_KEY, signifiying that you agree to the terms of
 * the GPL stated in the ASTERISK_GPL_KEY.  Your module will not load if it does
 * not return the EXACT message, i.e.  char *key(void){return ASTERISK_GPL_KEY;}
 */
char *key(void);/*! Return the below mentioned key,
unmodified */


may be you missed to specify that function

best regards

On 6/1/05, Bharat M. Sarvan [EMAIL PROTECTED] wrote:
  
  
 
 Hello everybody, 
 
  I have made a application of my own. (I.e. Def (
 )). I am able to compile the application successfully. And the .so file is
 created as well. But when I load asterisk I get the following error. 
 
   
 
   
 
 [Def.so]Jul  1 19:20:06 WARNING[15664]: loader.c:295 ast_load_resource: No
 key routine in module /usr/lib/asterisk/modules/Def.so 
 
 Jul  1 19:20:06 WARNING[15664]: loader.c:302 ast_load_resource: Key routine
 returned NULL in module /usr/lib/asterisk/modules/Def.so 
 
 Jul  1 19:20:06 WARNING[15664]: loader.c:311 ast_load_resource: 2 error(s)
 loading module /usr/lib/asterisk/modules/Def.so, aborted 
 
 Jul  1 19:20:06 WARNING[15664]: loader.c:440 load_modules: Loading module
 Def.so failed! 
 
   
 
   
 
   
 
 So if anybody could help me out as to where must I be going wrong, it would
 be very kind of you. 
 
   
 
   
 
   
 
   
 
 Regards, 
 
 Bharat M. Sarvan 
 
   
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-- 
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC

[Juan Luis Moyano]

Darren Wiebe wrote:
Could you please post the output from the asterisk console when 
astcc.agi crashes?  I really would like to get this resolved.


Darren Wiebe
[EMAIL PROTECTED]

Darren here I post you the output from asterisk console and the mysql 
daemon log. After hanging the phone the field inuse stays '1' and I get 
no cdr record. I'm using the cvs astcc.agi with astcc.patch applied.



//ASTCC agi debug


-- Executing DeadAGI(Zap/2-1, astcc.agi|11|615) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
AGI Tx  agi_request: astcc.agi
AGI Tx  agi_channel: Zap/2-1
AGI Tx  agi_language: en
AGI Tx  agi_type: Zap
AGI Tx  agi_uniqueid: 1120221737.19
AGI Tx  agi_callerid: CMW 11
AGI Tx  agi_dnid: unknown
AGI Tx  agi_rdnis: unknown
AGI Tx  agi_context: highclient
AGI Tx  agi_extension: 77615
AGI Tx  agi_priority: 3
AGI Tx  agi_enhanced: 0.0
AGI Tx  agi_accountcode: CMW
AGI Tx 
AGI Rx  ANSWER
AGI Tx  200 result=0
AGI Rx  STREAM FILE astcc-tone 0123456789
AGI Tx  200 result=0 endpos=11200
AGI Rx  STREAM FILE astcc-youhave 0123456789
AGI Tx  200 result=0 endpos=6400
AGI Rx  SAY NUMBER 25 0123456789
-- Playing 'digits/20' (language 'en')
-- Playing 'digits/5' (language 'en')
AGI Tx  200 result=0
AGI Rx  STREAM FILE astcc-dollars 0123456789
AGI Tx  200 result=0 endpos=7200
AGI Rx  STREAM FILE astcc-remaining 0123456789
AGI Tx  200 result=0 endpos=3360
AGI Rx  STREAM FILE astcc-willcost 0123456789
AGI Tx  200 result=0 endpos=14240
AGI Rx  SAY NUMBER 50 0123456789
-- Playing 'digits/50' (language 'en')
AGI Tx  200 result=0
AGI Rx  STREAM FILE astcc-perminute 0123456789
AGI Tx  200 result=0 endpos=14240
AGI Rx  STREAM FILE astcc-pleasewait 0123456789
AGI Tx  200 result=0 endpos=15840
AGI Rx  EXEC DIAL 
IAX2/657XXX:[EMAIL PROTECTED]/615|30|HL(300:6:3) 
-- AGI Script Executing Application: (DIAL) Options: 
(IAX2/657XXX:[EMAIL PROTECTED]/615|30|HL(300:6:3))

-- Limit Data:
-- timelimit=300
-- play_warning=6
-- play_to_caller=yes
-- play_to_callee=no
-- warning_freq=3
-- start_sound=UNDEF
-- warning_sound=timeleft
-- end_sound=UNDEF
-- Called 657XXX:[EMAIL PROTECTED]/615
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/65.39.205.121:4569/1 is making progress passing it to Zap/2-1
-- IAX2/65.39.205.121:4569/1 answered Zap/2-1
-- Hungup 'IAX2/65.39.205.121:4569/1'
AGI Tx  200 result=-1
AGI Rx  GET VARIABLE ANSWEREDTIME
AGI Tx  200 result=1 (24)
AGI Rx  GET VARIABLE DIALSTATUS
AGI Tx  200 result=1 (ANSWER)
-- AGI Script astcc.agi completed, returning 0

//MYSQL

050701 12:54:42 120 Connect [EMAIL PROTECTED] on astcc
050701 12:54:44 120 Query   SELECT * FROM cards WHERE number='11'
120 Query   SELECT * FROM cards WHERE number='11'
120 Query   SELECT * FROM cards WHERE number='11'
120 Query   SELECT * FROM cards WHERE number='11'
120 Query   UPDATE cards SET used='0' WHERE 
number='11'
120 Query   UPDATE cards SET inuse='1' WHERE 
number='11'
050701 12:54:47 120 Query   SELECT * FROM routes WHERE '615' 
RLIKE pattern ORDER BY LENGTH(pattern) DESC

050701 12:54:53 120 Query   SELECT * FROM cards WHERE number='11'
120 Query   SELECT * FROM trunks WHERE name='FWD'
050701 12:55:18 120 Quit


--
Juan Luis Moyano
[EMAIL PROTECTED]

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[Asterisk-Users] Visual ring notification

[Chris Mason (Lists)]

I have put a pbx into a resort with Polyycom phones, everythign works 
great, except the kitchen staff cannot hear the phone ring. I know many 
legacy systems employ a big red flashing light, any ideas on doing 
something similar?


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


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Re: [Asterisk-Users] Voicemail = SMS

[Mark Charlton]

On 7/1/05, Wilson Pickett [EMAIL PROTECTED] wrote:
  I have been trying for a while to find a way to get an SMS send when I
  receive a voicemail into my asterisk system.  I don't want to send an
  SMS if the caller doesn't leave a message.  I have voicemail.conf set
  up to email and delete.
 
 I use a backward solution to this problem, but it works. Orange, my
 cell provider offers free SMS alerts for email sent to
 [EMAIL PROTECTED] I send my vmail messages to my regular email
 server which keeps them for online email retrieval. A procmail recipe
 on the server then makes up an email without the vmail attachment to
 my orange address with the callerid in the subject. Orange sends an
 SMS that tells me I have a vmail message from ${CALLERID}. Although it
 seems like a silly solution it does _exactly_ what you asked about.
 
I have been fighting with the Bayham Systems FastSMS AGI script, and I
re-wrote it as a stand alone Perl script.  I am now calling it with
the EXTERNNOTIFY option in the voicemail.conf file.  It gets passed
the context, extension and number of messages which I build into a
text, and since they all go to the same location its no problem.  I'm
planning on using the extension info to open the mailbox, and read the
text file for the latest message to pull out the caller for the text. 
I might also have an extension map in a text file so I can look up who
to notify about a VM.

This works after a fashion, and crucially is only triggered when
someone actually leaves a valid voice mail message.  It is limited in
the fact I can't pass any other system details than extn and context. 
Plus the voicemail count is wrong since the attach=yes|delete=yes has
already deleted the message when it counts them.

But it works.

Thanks for all the help and advice.
Mark
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RE: [Asterisk-Users] Visual ring notification

[Brian C. Fertig]

How good is your electrical engineering?

..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent: Friday, July 01, 2005 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Visual ring notification

I have put a pbx into a resort with Polyycom phones, everythign works 
great, except the kitchen staff cannot hear the phone ring. I know many 
legacy systems employ a big red flashing light, any ideas on doing 
something similar?

-- 
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 

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All information provided in this email is considered confidential
and proprietary of Planet Telecom, Inc. and Telecenter Inc.
Use of this information by anyone other than the recipient or 
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RE: [Asterisk-Users] Linux Firewall Question

[Michael Stahl]

You should be able to do a good job with IPTABLES which is included in
FC3.  You can limit source  destp IP and protocol, etc.

Type man iptables | more for more details...

OCG 

-Original Message-
From: Terry H. Gilsenan [mailto:[EMAIL PROTECTED] 
Sent: Thursday, June 30, 2005 8:11 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Linux Firewall Question

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of OMS
 Sent: Friday, 1 July 2005 9:56 AM
 To: Asterisk-Users@lists.digium.com
 Subject: [Asterisk-Users] Linux Firewall Question
 
 Hi,
 I am running Asterisks on Public IP with Fedora Core 3. 
  
 What is the recommendation for making Linux secure on the Public IP 
 since I am new to Linux. Which Firewall should I use?  I am not 
 intending to use Linux as router.
  
 Can any one provide some configuration documentation. 

I use shorewall, and I have found it powerful, and fairly easy to use.

http://www.shorewall.net/

T


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[Asterisk-Users] Re:passing through MWI info from SBC

[Mike Myers]

John Novack wrote:

Mike Myers wrote:

Hi..  I am about to replace my aging Nortel Venture
system with an Asterisk system and 6 Polycom IP 501
phones, and a couple sipura 841's for less used
areas.

We have 3 phone lines here.  One is SBC, one Vonage,
and one Voipjet...  One hangup is that I can't
figure
out how to pass through a voicemail waiting
indication
from SBC.  This is important because my wife and her
family all exchange voicemails with each other on
the
SBC voicemail system.  They can leave messages for
each other without having the phones ring, etc... 
We
have a 2 yr old at home, and her sister has some
small
kids too, so that's how they manage to send
voicemails
when they are unsure if the kids are sleeping,
etc... 
Anyway, preserving this capability of using the SBC
VM
and being notified when a message is waiting is
critical for good WAF.  

The vonage line and voipjet line can be intergrated
into the Asterisk VM.  My Nortel venture phones
light
the MWI if any line has VM on it, and the display
tells you which lines have VM waiting.  I would love
to be able to duplicate this function on the
Polycom's
and hopefully the Sipura's as well.

I've looked for answers on this, but haven't found
one, hence the post.  My apologies if I have missed
something.  

Thanks much,
Mike
  

You haven't missed much.
With SBC you are out of luck, since Asterisk doesn't
detect dialtone  ( 
it dials blind, sometimes too quickly for the CO to
catch the first 
digit, resulting in wrong numbers )) or stutter
dialtone either, and 
reportedly has had any indication of the DC status of
a POTS line 
removed due to problems.
 
Only choice would to port the number to a VOIP 
provider and provide the 
VM in Asterisk.
Similar problem with Vonage VM.

John Novack

Wow, this is a serious problem for me.  I don't need
to actually check the voicemail itself from Asterisk,
just to be able to tell that there is voicemail
waiting.  Are you saying there is no way in Asterisk
to do this?   Is that true for using Digium hardware
as well as FXO ports on a SIP ATA?  

Vonage VM doesn't matter to me, since I'll turn it off
and use Asterisk for that functionality, but
determining SBC's VM status is very important.  My
whole wife's family (multiple households) uses it.  In
the past, if one family tried to switch to a non SBC
provider, they always returned in less than a week
because of lack of VM interoperation. So my wife will
put the kibosh on the whole Asterisk project unless I
can light the MWI light when SBC VM is waiting.  Since
the cheapest analog phones can do this, I don't think
she's going to understand that these $200 Polycom
phones can't...  :-(

Is there no way around this?

Thanks,
Mike




 
Yahoo! Sports 
Rekindle the Rivalries. Sign up for Fantasy Football 
http://football.fantasysports.yahoo.com
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Re: [Asterisk-Users] Quality of provider: VocTel

[Andrew Kohlsmith]

On Wednesday 29 June 2005 23:36, Michael Stahl wrote:
 How have you found the quality (Choppy / smooth audio)?
 Any problems registering?  (I have been unable to register for hours)

I use them for some of my termination, they seem to work just fine (no 
quality/registration issues).

Actually once I did have a problem where I couldn't seem to get any calls out 
but power-cycling *MY* firewall fixed it, so it certainly wasn't an issue on 
their end.

-A.
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Re: [Asterisk-Users] Visual ring notification

[Jason Becker]

I have put a pbx into a resort with Polyycom phones, everythign works 
great, except the kitchen staff cannot hear the phone ring. I know many 
legacy systems employ a big red flashing light, any ideas on doing 
something similar?


 


FYI, the Uniden UIP200 has a big red flashing light.

Regards,

--
Jason Becker
Director  CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca

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RE: [Asterisk-Users] Visual ring notification

[Dean Collins]

Analog relay in the same ring group with a bell?

Cheers,
Dean


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Chris Mason (Lists)
 Sent: Friday, 1 July 2005 12:09 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Visual ring notification
 
 I have put a pbx into a resort with Polyycom phones, everythign works
 great, except the kitchen staff cannot hear the phone ring. I know
many
 legacy systems employ a big red flashing light, any ideas on doing
 something similar?
 
 --
 Chris Mason
 NetConcepts
 (264) 497-5670 Fax: (264) 497-8463
 Int:  (305) 704-7249 Fax: (815)301-9759
 Cell: 264-235-5670
 Yahoo IM: [EMAIL PROTECTED]
 
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Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)

[Robert Goodyear]

On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote:


I am confused about one of my installed server

The dial plan seems to be ok, but sometimes NOTHING happens if I try 
to dial an extension (from X-Lite), next time it is done.


X-Lite does not have a tone, nothing and does also have no time out. 
It seems it is not connected to the server. However, a sip show users 
/ sip show peers   shows that the phone is connected.




SIP clients generate their own dialtone, so if you've got no tone, that 
sounds suspicious of a problem with the client itself. I assume you've 
debugged the problem by registering a hard SIP client on that server?


--
Robert Goodyear
Brand Up LLC
http://www.brand-up.com

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Re: [Asterisk-Users] Computer to use

[Robert Goodyear]

On Jul 1, 2005, at 4:03 AM, Eric Wieling aka ManxPower wrote:


Robert Goodyear wrote:

I'm sure you really only want to know about the absence of problems.  
From watching this list for 6 months it seems the SuperMicro products 
are most lauded and have exhibited no hardware conflicts. Various 
votes on Dell products, so you're probably best to stay away, even 
though I've got five installs with TE110Ps in them that have never 
missed a beat -- Dimension boxes, not PowerEdge.


The SuperMicro Xeon board we tried failed miserably with both the 
T100P and TE110P.  It had the ServerWorks IDE Chipset, which I suspect 
was the problem.


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


Bummer! I thought I'd heard all good things about them... sorta like 
VoIP providers; as soon as everyone agrees things are OK, something 
goes awry!


-Rob.

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[Asterisk-Users] RE: [asterisk] VocTel service provider

[Michael Stahl]

I haven't heard much feedback yet - anyone here using VocTel?

The connection problem turned out to be my firewall, but I'm curious if
others experience any voice choppiness or high latency.  Some posters
have related the problem to specific VOIP providers, some seem to be ISP
related (local network latency).

Any feedback?

OCG
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[Asterisk-Users] Epia C3 Linux

[Wiley Siler]

Anyone know a good distro for an Epia Mobo with the C3
chip? 



I have been trying to get Debian and Gentoo installed (new
to me) and so far having little luck. 



Does anyone know a good install for this processor/mobo
combo?



Thanks

Wiley










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[Asterisk-Users] Re: passing through MWI info from SBC

[John Novack]

Mike Myers wrote:


John Novack wrote:

 


Mike Myers wrote:
   



 


Hi..  I am about to replace my aging Nortel Venture
system with an Asterisk system and 6 Polycom IP 501
phones, and a couple sipura 841's for less used
 


areas.
 


We have 3 phone lines here.  One is SBC, one Vonage,
and one Voipjet...  One hangup is that I can't
 


figure
 


out how to pass through a voicemail waiting
 


indication
 


from SBC.  This is important because my wife and her
   


family all exchange voicemails with each other on
 


the
 


SBC voicemail system.  They can leave messages for
each other without having the phones ring, etc... 
 


We
 


have a 2 yr old at home, and her sister has some
 


small
 


kids too, so that's how they manage to send
 


voicemails
 


when they are unsure if the kids are sleeping,
 

etc... 
 


Anyway, preserving this capability of using the SBC
 


VM
 


and being notified when a message is waiting is
critical for good WAF.  


The vonage line and voipjet line can be intergrated
into the Asterisk VM.  My Nortel venture phones
 


light
 


the MWI if any line has VM on it, and the display
tells you which lines have VM waiting.  I would love
to be able to duplicate this function on the
 


Polycom's
 


and hopefully the Sipura's as well.

I've looked for answers on this, but haven't found
one, hence the post.  My apologies if I have missed
something.  


Thanks much,
Mike
 




You haven't missed much.
With SBC you are out of luck, since Asterisk doesn't
detect dialtone  ( 
it dials blind, sometimes too quickly for the CO to
catch the first 
digit, resulting in wrong numbers )) or stutter
dialtone either, and 
reportedly has had any indication of the DC status of
   

a POTS line 
 


removed due to problems.
   



 

Only choice would to port the number to a VOIP 
provider and provide the 
VM in Asterisk.

Similar problem with Vonage VM.
   



 


John Novack
   



Wow, this is a serious problem for me.  I don't need
to actually check the voicemail itself from Asterisk,
just to be able to tell that there is voicemail
waiting.  Are you saying there is no way in Asterisk
to do this?   Is that true for using Digium hardware
as well as FXO ports on a SIP ATA?  


Vonage VM doesn't matter to me, since I'll turn it off
and use Asterisk for that functionality, but
determining SBC's VM status is very important.  My
whole wife's family (multiple households) uses it.  In
the past, if one family tried to switch to a non SBC
provider, they always returned in less than a week
because of lack of VM interoperation. So my wife will
put the kibosh on the whole Asterisk project unless I
can light the MWI light when SBC VM is waiting.  Since
the cheapest analog phones can do this, I don't think
she's going to understand that these $200 Polycom
phones can't...  :-(

Is there no way around this?

Thanks,
Mike
 


AFAIK, there is no way around this with a POTS line.
SBC's only indication of MW is stutter dial tone, correct?

Since Asterisk doesn't detect ANY sort of dialtone, either with the X100 
or TDM400, it seems you are out of luck.
AFAIK, the ATA's don't detect stutter dial tone either, though some may 
listen for dialtone before dialing.


Perhaps some others can offer a solution.
Curious, since a cheap $20 box sitting on your line can give you a 
visual indication.


What are you doing now with the Nortel? Does it know?

BTW - the Sipura 841 is an OK inexpensive phone, but speakerphone and 
display are unusable.
A somewhat better buy is the Grandstream. At least you can read the 
display. Speakerphone suffers from echo, and it has no built in echo 
canceller.
The more expensive brands are , well, too expensive and seem difficult 
to configure.


JMO

John Novack

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Re: [Asterisk-Users] Epia C3 Linux

[Oliver Rath]

Wiley Siler wrote:

 Anyone know a good distro for an Epia Mobo with the C3 chip?  

  

 I have been trying to get Debian and Gentoo installed (new to me) and
 so far having little luck. 

  

 Does anyone know a good install for this processor/mobo combo?

  

You have to compile without mmx and sse, best 586compatible, because
linux is recognizing C3 as PIII, what is definitly wrong.

Hth,

Oliver

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Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC

[Ade Agbero]

Finally, We have lift off, a shaky one though.

I deleted my Astcc.gi and replaced it with Darren's copy posted on his website and I have finally been able to get something recorded as BILLCOST.

I say "something" because the amount I expected and amount entered are different.

Please see below, I expect any call less than 60secs to generate a10 charge (see caller id 1234),butthe amount being generated doesn't follow the charges set in Routes (seecaller id 2, 3, 4): ANY IDEAS WHY???





Caller*ID
Called Number
Trunk
Disposition
Billable Seconds
Billed Cost

1234
19313256895
DANSAM
ANSWER
14
10

2
19313256895
DANSAM
ANSWER
4
1

3
19313256895
DANSAM
ANSWER
48
8

4
19313256895
DANSAM
ANSWER
21
4






Pattern
Comment
Trunks
Connect Fee
Inc. Seconds
Cost per additional minute

44.*

DANSAM
0
0
10


1.*

DANSAM
0
0
10



Darren Wiebe [EMAIL PROTECTED] wrote:
Could you please post the output from the asterisk console when astcc.agi crashes? I really would like to get this resolved.Darren Wiebe[EMAIL PROTECTED]Juan Luis Moyano wrote:Ade Agbero wrote: I tried using your working astcc.agi file instead of mine, but thatfailed to work too. Having the same issues here.. it seems astcc.agi is crashing. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Visual ring notification

[Andrew M Stemen]

I had the same problem in a welding lab. I did this on a 3Com NBX, but 
I'm sure that the same idea would apply to Asterisk: I went to 
RadioShack, and bought one of their visual ringers, for the hearing 
impaired (basically flashes a white strobe light, and sounds a really 
loud ringer), and attached it to an ATA adapter. Then I created a flat 
call group, and reassigned its number to what the single phone's 
extension used to be. That way, whenever anyone would dial that 
extension, the strobe would activate and the louder ringer would sound, 
in addition to the telephone.


Andrew M Stemen
[EMAIL PROTECTED]
http://www.andrewmstemen.com


Chris Mason (Lists) wrote:
I have put a pbx into a resort with Polyycom phones, everythign works 
great, except the kitchen staff cannot hear the phone ring. I know many 
legacy systems employ a big red flashing light, any ideas on doing 
something similar?



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RE: [Asterisk-Users] Epia C3 Linux

[Wiley Siler]

Oliver,

Thanks for the response!  Do you know where I can find an example of how
to do this?  I have never had to install a custom kernel before.

Thanks!
Wiley


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oliver
Rath
Sent: Friday, July 01, 2005 10:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Epia C3 Linux

Wiley Siler wrote:

 Anyone know a good distro for an Epia Mobo with the C3 chip?  

  

 I have been trying to get Debian and Gentoo installed (new to me) and
 so far having little luck. 

  

 Does anyone know a good install for this processor/mobo combo?

  

You have to compile without mmx and sse, best 586compatible, because
linux is recognizing C3 as PIII, what is definitly wrong.

Hth,

Oliver

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Re: [Asterisk-Users] Epia C3 Linux

[Oliver Rath]

Wiley Siler wrote:

Oliver,

Thanks for the response!  Do you know where I can find an example of how
to do this?  I have never had to install a custom kernel before.
  

For Gentoo there is a superb dokumentation on
http://www.gentoo.org/doc/en/index.xml to do this.

Regards,

Oliver

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RE: [Asterisk-Users] Got this error after my installation when i doztcfg -vv

[Mahmoud Badran]

hello u can see the readme.udev in the zaptel directory that's normally answers 
ur question
 
 
 



From: [EMAIL PROTECTED] on behalf of Ian Bert Tusil
Sent: Fri 7/1/2005 9:37 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Got this error after my installation when i doztcfg 
-vv



how can i solve the error on the last part?

need help. thnx...


Zaptel Configuration
==

SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: Individual Clear channel (Default) (Slaves: 03)
Channel 04: Individual Clear channel (Default) (Slaves: 04)
Channel 05: Individual Clear channel (Default) (Slaves: 05)
Channel 06: Individual Clear channel (Default) (Slaves: 06)
Channel 07: Individual Clear channel (Default) (Slaves: 07)
Channel 08: Individual Clear channel (Default) (Slaves: 08)
Channel 09: Individual Clear channel (Default) (Slaves: 09)
Channel 10: Individual Clear channel (Default) (Slaves: 10)
Channel 11: Individual Clear channel (Default) (Slaves: 11)
Channel 12: Individual Clear channel (Default) (Slaves: 12)
Channel 13: Individual Clear channel (Default) (Slaves: 13)
Channel 14: Individual Clear channel (Default) (Slaves: 14)
Channel 15: Individual Clear channel (Default) (Slaves: 15)
Channel 16: Individual Clear channel (Default) (Slaves: 16)
Channel 17: Individual Clear channel (Default) (Slaves: 17)
Channel 18: Individual Clear channel (Default) (Slaves: 18)
Channel 19: Individual Clear channel (Default) (Slaves: 19)
Channel 20: Individual Clear channel (Default) (Slaves: 20)
Channel 21: Individual Clear channel (Default) (Slaves: 21)
Channel 22: Individual Clear channel (Default) (Slaves: 22)
Channel 23: Individual Clear channel (Default) (Slaves: 23)
Channel 24: D-channel (Default) (Slaves: 24)

24 channels configured.

Notice: Configuration file is /etc/zaptel.conf
line 145: Unable to open master device '/dev/zap/ctl'
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Re: [Asterisk-Users] Voicemail = SMS

[Peter Bowyer]

On 01/07/05, Mark Charlton [EMAIL PROTECTED] wrote:
 
 I have been fighting with the Bayham Systems FastSMS AGI script, and I
 re-wrote it as a stand alone Perl script.  I am now calling it with
 the EXTERNNOTIFY option in the voicemail.conf file.  It gets passed
 the context, extension and number of messages which I build into a
 text, and since they all go to the same location its no problem.  I'm
 planning on using the extension info to open the mailbox, and read the
 text file for the latest message to pull out the caller for the text.
 I might also have an extension map in a text file so I can look up who
 to notify about a VM.

I also hacked Bayham Systems' script. I need to control the MWI on GSM
phones, which is turned on and off by custom SMS messages (which
Bayham helpfully provide macros for). And I needed to avoid sending
repeat notifications when a second or subsequent new message was left.

I ended up keeping a flag in a db file which stores per mobile number
what state the MWI is in for that phone. If the flag says the MWI is
already set when a new message comes in, then the script is a no-op,
otherwise it sends the 'MWI on' SMS and flips the flag. Ditto in
reverse.

Works fine and dandy here.


Peter



-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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[Asterisk-Users] how to send voicemail notifcation every 15 minutes until message is checked

[Pedro]

I have searched quite a few places and have not seen this discussed. 
Basically I was wondering how would you go about having an option for
a user to be notified every 15 minutes until their new voicemail
message is checked.  Since the notification e-mails we send get sent
to cell phones or actual pagers (via e-mail), there are times when a
person is out of range and misses a page or just simply is too busy to
check voicemail and then forgets.  They want to be reminded 15 minutes
later until that new message is checked.

Current version of asterisk that we are running is CVS-v1-0-11/12/04
(which has been running rock-solid I might add).  Any thoughts are
appreciated.
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Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC

[Juan Luis Moyano]

Ade Agbero wrote:

Finally, We have lift off, a shaky one though.
 
I deleted my Astcc.gi and replaced it with Darren's copy posted on his 
website and I have finally been able to get something recorded as BILLCOST.
 


I got it working too here with Darren's astcc.agi. And billing as 
expected so finally It's working. It would be nice if someone could 
update the cvs with Darren's astcc.agi, because the current one doesn't 
work, even patched.. it gets worse. Thanks for your attention Darren!


--
Juan Luis Moyano
[EMAIL PROTECTED]

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[Asterisk-Users] how does pattern routes works

[wassim darwish]

i tried to write to usa destination 1* it worked well
but when i tried to specify the number of digits i
wrote
1NXXNXX but it did'nt work.can anybody help me
please 
 please.   



 
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[Asterisk-Users] Unable to forward frame/voice

[Christian M. Watts]

Hi,

We've exhausted our internal capabilities as well as Sangoma tech support and
were hoping someone with some expertise could help us with a pointer. Briefly,
our issue is as follows.

Periodically (several times an hour), we get either of the following error
messages in our asterisk messages log. These correspond with dropped outbound
calls on a one-to-one basis when the second error happens. The first error
sometimes causes a dropped call and sometimes does not:

Jun 30 16:40:27 WARNING[5395] app_dial.c: Unable to forward frame
Jun 30 16:45:07 WARNING[5455] app_dial.c: Unable to forward voice


Our hardware is as follows:

Compaq DL380 Dual PIII 1Ghz, 1.2 GB RAM, Onboard SmartArray for SCSI RAID
Sangoma A102U dual-port T1 card
Digi Datafire T1 fax/modem board


Our software is as follows:

Linux 2.4.30
Asterisk, Zaptel and Libpri from CVS HEAD as of 6/28/05
Sangoma wanpipe 2.3.3-beta11 (latest as of this post)
Patton electronic's latest drivers and firmware for our Digi Datafire board
(still no 2.6 Linux support, which is why we're on 2.4)
Hylafax 4.2.1 driving the Digi Datafire


The path (for the problem calls) looks like this:

Digi Datafire - Sangoma Port B - Sangoma Port A - Telco

Basically, sending a fax over a PRI with asterisk doing TDM bridging in the
middle.


We have confirmed the following (based on similar posts to this list related to
the same problem with Digium boards as well as Sangoma tech support
assistance):

1. Sangoma Port A takes clocking from the telco
2. Sangoma Port B retransmits A's clocking and acts as master
3. Sangoma tech support says our configs are correct
4. Zaptel.conf is set up with Sangoma Port A as the primary clock source, and
Port B to not be used as a clock source
5. LBO, switch options, etc. are correct for the environment (since 98% of
outbound calls are fine, this seems fairly obvious)
6. ISDN Transfer Capability gets properly set to 3K1AUDIO for calls
7. No IRQ sharing on the system
8. IDE DMA mode is irrelevant, since there are no IDE disks in the system (other
than the CDROM)


We have tried the following:

1. Asterisk, libpri and zaptel versions from 6/1/2005, 6/15/2005 and 6/28/2005 -
no change in behavior
2. Wanpipe drivers 2.3.3-beta8 and 2.3.3-beta11 - no change in behavior
3. Wanpipe configured both with and without the D-Channel hardware HDLC - no
change in behavior
4. Firmware versions 24 (shipped version) and 25 (latest version) on the Sangoma
card - no change in behavior
5. callprogress and busydetect both 'yes' and 'no' in zapata.conf (currently
'no') - no change in behavior
6. Added SetTransferCapability(3K1AUDIO) to our dialplan, just to be sure - no
change in behavior


General environment:

1. We run TDM only, no VoIP protocols are in use. SIP, IAX2, MGCP are all noload
in modules.conf.
2. This problem occurs with as few as one simultaneous channel active and as
many as 15 simultaneous channels active with equal frequency (i.e.: not load
related). The load on the box is negligible in any case, plenty of RAM is free,
etc.
3. Restarting asterisk does seem to cause the problem not to re-present itself
for 30 minutes to 2 hours. When asterisk is restarted, the Sangoma and Zaptel
kernel modules are also unloaded and reloaded.


Again, any pointers or help would be greatly appreciated.

Thanks,
Christian
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Re: [Asterisk-Users] how does pattern routes works

[Christopher Stephens]

Pattern-matching extensions must be prefaced with an underscore thus:

_1NXXNXX

Enjoy!

On Fri, 1 Jul 2005 10:32:46 -0700 (PDT), wassim darwish
[EMAIL PROTECTED] said:
 i tried to write to usa destination 1* it worked well
 but when i tried to specify the number of digits i
 wrote
 1NXXNXX but it did'nt work.can anybody help me
 please 
  please.   
 
 
   
  
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 Rekindle the Rivalries. Sign up for Fantasy Football 
 http://football.fantasysports.yahoo.com
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Pattern-matching extensions must be prefaced with an underscore thus:

_1NXXNXX

Enjoy!
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Re: [Asterisk-Users] Can't build cdr_addon_mysql.

[Brian West]

You could have just done ln -s asterisk-1.0.9 asterisk and it would  
have fixed that.  It should by default do -I../asterisk


/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 30, 2005, at 1:13 AM, Chris Mason (Lists) wrote:


Marcel van Kaam, Fonetica wrote:


I had the same problem with installing addons. I checked out in  
the file
cdr_addons_mysql.c what the location of the asterisk.h must be and  
changed

the cdr_addons_mysql.c to the location of the asterisk.h file.

After this it worked. Also to be sure do: locate asterisk.h to  
check or you

have the file on your system.

Marcel


Yes, that worked. For the record, it had to be

#include ../asterisk-1.0.9/asterisk.h

--
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NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670
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RE: [Asterisk-Users] make error for zaptel

[Mahmoud Badran]

maybe zaptel verion incompatability try other newer or stable older versions 
not sure thats just a hint 



From: [EMAIL PROTECTED] on behalf of Zoltan Szecsei
Sent: Fri 7/1/2005 2:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] make error for zaptel



Hi,
I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the
box (in a hope to sort out the uname -r issue mentioned below).
I'm using the Asterisk Doc Proj vol 1 to guide me through the initial
setup. I have no special HW and intend to use asterisk on an internal
network  just to get some experience.

I have downloaded what I think I need and placed it in /usr/src (see
listing below).
I run make clean ; make linux26 (what about the usual make with no
parameters?) and I get a crash.

Note that uname -r returns a *different* version of what the linux is
linked to (thanks to YOU??)

I have tried make clean ; make (no params) and it still crashes.

Can anyone offer me some suggestions? - or do I go first to the SuSE
list to sort out the uname -r  usr/src/linux issue?

TIA,
Zoltan.

gl0:/usr/src # ls -la
total 499
drwxr-xr-x  17 root root680 2005-06-30 14:46 .
drwxr-xr-x  13 root root368 2005-06-30 09:21 ..
drwxr-xr-x   3 root root320 2005-05-19 06:53 astcc
-rw-r--r--   1 root root 130943 2005-06-25 19:09 astcc.tar
drwxr-xr-x  22 root root   2336 2005-06-23 04:16 asterisk-1.0.8
drwxr-xr-x   7 root root432 2005-06-23 04:21 asterisk-addons-1.0.8
drwxr-xr-x   3 root root184 2005-06-23 04:23 asterisk-sounds-1.0.8
drwxr-xr-x   2 root root440 2005-05-23 06:47 btp
-rw-r--r--   1 root root  32975 2005-06-25 19:08 btp.tar
drwxr-xr-x  23 root root776 2005-06-12 17:24 dicts
drwxr-xr-x   5 root root416 2005-04-01 18:50 gastman
-rw-r--r--   1 root root 332857 2005-06-25 19:08 gastman.tar
drwxr-xr-x  25 root root736 2005-06-12 17:29 kernel-modules
drwxr-xr-x   2 root root520 2005-06-23 04:11 libpri-1.0.8
lrwxrwxrwx   1 root root 19 2005-06-25 22:01 linux -
linux-2.6.11.4-21.7
drwxr-xr-x   3 root root 72 2005-06-25 22:01 linux-2.6.11.4-20a
drwxr-xr-x   3 root root 72 2005-03-24 02:03 linux-2.6.11.4-20a-obj
drwxr-xr-x  19 root root720 2005-06-25 22:01 linux-2.6.11.4-21.7
drwxr-xr-x   3 root root 72 2005-06-02 16:54 linux-2.6.11.4-21.7-obj
lrwxrwxrwx   1 root root 23 2005-06-25 22:01 linux-obj -
linux-2.6.11.4-21.7-obj
drwxr-xr-x   7 root root168 2005-06-12 17:43 packages
drwxr-xr-x   2 root root   2720 2005-07-01 12:43 zaptel-1.0.8
gl0:/usr/src # uname -r
2.6.11.4-20a-smp
gl0:/usr/src # cd zaptel-1.0.8/
gl0:/usr/src/zaptel-1.0.8 # make clean
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo
rm -f *.ko *.mod.c .*o.cmd
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f core
gl0:/usr/src/zaptel-1.0.8 # make linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c   -o makefw
./makefw tormenta2.rbt tor2fw  tor2fw.h
Loaded 69900 bytes from file
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo
zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo
tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o -lm -L. libtonezone.a
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o torisatool.o torisatool.c
cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o ztmonitor.o ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -c ztspeed.c
cc -o ztspeed ztspeed.o
make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel-1.0.8 modules
make[1]: Entering directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp'
make[1]: *** No rule to make target `modules'.  Stop.
make[1]: Leaving directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp'
make: *** [linux26] Error 2
gl0:/usr/src/zaptel-1.0.8 #

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RE: [Asterisk-Users] asterisk newbie and phones which don't want tocomunicate

[Mahmoud Badran]

hi do u have the sip phones extensions in the extension.conf and are they in 
the right context (sip-incoming)???
 
are the sip phone registering to asterisk?? try stop asterisk and reconnect as 
asterisk -vvvc to check see them registering...
 
 
 



From: [EMAIL PROTECTED] on behalf of Sistemista WebSolvingJaa
Sent: Fri 7/1/2005 6:43 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] asterisk newbie and phones which don't want 
tocomunicate



Hi list, i'm an asterisk newbie and i've to setup a net with an
asterisk server and several ip phones linked on the net.
i hope my questions are IT ans if you have some link for solving those
problems please mail me.
i've wrote the sip.conf in this way:
[2011]
type=friend
username=2011
secret=1945
regexten=1234
host=dynamic
;host=192.168.100.242
permit=192.168.100.0/24
context=sip-incoming
canreinvite=yes
dtmfmode=rfc2833
nat=1

[2012]
type=friend
username=2012
secret=1945
regexten=1234
host=dynamic
;host=192.168.100.221
permit=192.168.100.0/24
context=sip-incoming
canreinvite=yes
dtmfmode=rfc2833
nat=1

and the extension.conf if quitelly the same as the original. the
phones softwares are setted up correctly, but from a phone i can't
call another phone on the net. can somebody suggest me a possible
solution?

thanks a lot
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