Re: [asterisk-users] Best alternative for getting prompts recorded.

[Francesco Peeters]

Steve Edwards wrote:
> On Wed, 12 Mar 2008, [EMAIL PROTECTED] wrote:
>
>   
>> Thanks everyone for the reply.
>>
>> Till now we had simple IVR so we recorded it ourself.
>> Now I have a requirement where customer needs a customized message to be 
>> played to customer. I am basically looking for some Text to Speech software 
>> that would be cost effective (most probably a open source) and would convert 
>> Text to Speech.
>>
>> I tried Fetival, but the quality of the sound is not good. Can we improve 
>> the sound quality of Festival somehow.
>> 
>
> Cepstral with Allison is only $30.
>
> I did a demo IVR for a potential client and it was hard to tell the TTS 
> bits from the human bits. If I took the time to learn Cepstral's markup 
> language I probably could have fooled myself :)
>
> Thanks in advance,
Are there any tools like these for Dutch language Asterisk installs?...
-- 
Francesco Peeters
no sigs on this machine!  :-o

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Re: [asterisk-users] Need to find small footprint asterisk platform

[Francesco Peeters]

Tzafrir Cohen wrote:
> On Mon, Mar 30, 2009 at 06:20:20PM +0100, Chris Bagnall wrote:
>   
>>> One of the more common embedded platforms for Asterisk is the Soekris
>>> net5501 (or 4501 if you don't need as much processing power)
>>>   
>> Agreed. Though, given the Asus eeeBox (1.6Ghz Atom) can be had for 
>> almost the same money (Soekris stuff isn't cheap in the UK) and is 
>> about the same footprint, it might be worth considering that instead 
>> if you don't need ISDN or POTS connectivity.
>>
>> I've done a few Asterisk-based eeeBoxes over the last few weeks and 
>> been very impressed with them.
>> 
>
> In fact, with a netbook I suspect you'd be paying quite a sum for the
> display. Both in the price and in the heat consumption. 
>
>   
Who's talking about netbooks?  :-o
What screen?

--FP

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Re: [asterisk-users] Echo and static on PRI with errors

[Francesco Peeters]

John F. Ervin wrote:
> What do you do if you find things sharing interrupts (IRQ 11) in my
> case with my X100P card.  I believe there is some sort of internal
> audio card in my cheap slow PC.
>
Check the BIOS whether you can:
Change the IRQ assignments
Disable the extra hardware using the same IRQ

Or otherwise try changing the slot it is in... I had very good results
in the past swapping card around

Good luck!

--FP

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[asterisk-users] Voipbuster not ringing, other SIP peers are ringing...

[Francesco Peeters]

Does anybody else see the same behavior for VoipBuster connections?

When I trace one of the other SIP peers, I see it sends this message:
--
<--- SIP read from 82.101.62.99:5060 --->
SIP/2.0 180 Ringing
Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE
Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl
Contact: 
Content-Type: application/sdp
CSeq: 103 INVITE
From: "**" ;tag=as70e84199
Record-Route:
,
Server: Cirpack/v4.41b (gw_sip)
To: ;tag=00-08168-044b6f36-245cd72c7
Via: SIP/2.0/UDP
***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92
Content-Length: 182

v=0
o=cp10 125193221174 125193221174 IN IP4 82.101.62.66
s=SIP Call
c=IN IP4 194.109.8.2
t=0 0
m=audio 36984 RTP/AVP 8
b=AS:64
a=rtpmap:8 PCMA/8000/1
a=ptime:20
a=sendrecv

<->
--- (12 headers 10 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 194.109.8.2:36984
Found audio description format PCMA for ID 8
Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8
(alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 194.109.8.2:36984
-- SIP/*-089ca9b8 is ringing
-- SIP/*-089ca9b8 is making progress passing it to
IAX2/2104-2287
Scheduling destruction of SIP dialog
'740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 82.101.62.99:5060:
CANCEL sip:0031**...@sip.xs4all.nl SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport
From: "**" ;tag=as70e84199
To: 
Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

--


However when I dial exactly the same from VoipBuster, I see this instead:


--
<--- SIP read from 77.72.169.129:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport
From: "*" ;tag=as1374705a
To: ;tag=120113ac4a54a269af9e2c
Contact: sip:0031**...@77.72.169.129:5060
Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 162

v=0
o=* 1251932194 1251932194 IN IP4 194.221.62.33
s=SIP Call
c=IN IP4 194.221.62.33
t=0 0
m=audio 8958 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20

<->
--- (11 headers 8 lines) ---
Found RTP audio format 0
Peer audio RTP is at port 194.221.62.33:8958
Found audio description format PCMU for ID 0
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 194.221.62.33:8958
-- SIP/-089dc538 is making progress passing it to IAX2/2104-8077
  == Connect attempt from '127.0.0.1' unable to authenticate
Scheduling destruction of SIP dialog
'1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com' in 6400 ms
(Method: INVITE)
Reliably Transmitting (NAT) to 77.72.169.129:5060:
CANCEL sip:0031**...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport
From: "**" ;tag=as1374705a
To: 
Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
--

As you can see, there are different packets being sent, and in the 2nd
case, there is no "is ringing" message, which is rather irritating...

Any suggestions would be appreciated...

TIA
-- 
FP

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Re: [asterisk-users] Voipbuster not ringing, other SIP peers are ringing...

[Francesco Peeters]

Francesco Peeters wrote:
> Does anybody else see the same behavior for VoipBuster connections?
>
> When I trace one of the other SIP peers, I see it sends this message:
> --
> <--- SIP read from 82.101.62.99:5060 --->
> SIP/2.0 180 Ringing
> Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE
> Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl
> Contact: 
> Content-Type: application/sdp
> CSeq: 103 INVITE
> From: "**" ;tag=as70e84199
> Record-Route:
> ,
> Server: Cirpack/v4.41b (gw_sip)
> To: ;tag=00-08168-044b6f36-245cd72c7
> Via: SIP/2.0/UDP
> ***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92
> Content-Length: 182
>
> v=0
> o=cp10 125193221174 125193221174 IN IP4 82.101.62.66
> s=SIP Call
> c=IN IP4 194.109.8.2
> t=0 0
> m=audio 36984 RTP/AVP 8
> b=AS:64
> a=rtpmap:8 PCMA/8000/1
> a=ptime:20
> a=sendrecv
>
> <->
> --- (12 headers 10 lines) ---
> Found RTP audio format 8
> Peer audio RTP is at port 194.109.8.2:36984
> Found audio description format PCMA for ID 8
> Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8
> (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
> combined - 0x0 (nothing)
> Peer audio RTP is at port 194.109.8.2:36984
> -- SIP/*-089ca9b8 is ringing
> -- SIP/*-089ca9b8 is making progress passing it to
> IAX2/2104-2287
> Scheduling destruction of SIP dialog
> '740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl' in 6400 ms (Method: INVITE)
> Reliably Transmitting (NAT) to 82.101.62.99:5060:
> CANCEL sip:0031**...@sip.xs4all.nl SIP/2.0
> Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport
> From: "**" ;tag=as70e84199
> To: 
> Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl
> CSeq: 103 CANCEL
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
> --
>
>
> However when I dial exactly the same from VoipBuster, I see this instead:
>
>
> --
> <--- SIP read from 77.72.169.129:5060 --->
> SIP/2.0 183 Session progress
> Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport
> From: "*" ;tag=as1374705a
> To: ;tag=120113ac4a54a269af9e2c
> Contact: sip:0031**...@77.72.169.129:5060
> Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com
> CSeq: 103 INVITE
> Server: (Very nice Sip Registrar/Proxy Server)
> Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
> Content-Type: application/sdp
> Content-Length: 162
>
> v=0
> o=* 1251932194 1251932194 IN IP4 194.221.62.33
> s=SIP Call
> c=IN IP4 194.221.62.33
> t=0 0
> m=audio 8958 RTP/AVP 0
> a=rtpmap:0 PCMU/8000
> a=ptime:20
>
> <->
> --- (11 headers 8 lines) ---
> Found RTP audio format 0
> Peer audio RTP is at port 194.221.62.33:8958
> Found audio description format PCMU for ID 0
> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0
> (nothing), combined - 0x4 (ulaw)
> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
> combined - 0x0 (nothing)
> Peer audio RTP is at port 194.221.62.33:8958
> -- SIP/-089dc538 is making progress passing it to IAX2/2104-8077
>   == Connect attempt from '127.0.0.1' unable to authenticate
> Scheduling destruction of SIP dialog
> '1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com' in 6400 ms
> (Method: INVITE)
> Reliably Transmitting (NAT) to 77.72.169.129:5060:
> CANCEL sip:0031**...@sip.voipbuster.com SIP/2.0
> Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport
> From: "**" ;tag=as1374705a
> To: 
> Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com
> CSeq: 103 CANCEL
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
> --
>
> As you can see, there are different packets being sent, and in the 2nd
> case, there is no "is ringing" message, which is rather irritating...
>
> Any suggestions would be appreciated...
>
> TIA
>   
BTW: I am talking about the ringtone the caller should hear... The other
side is ringing, and calls are established just fine, but it is very
irritating to hear nothing until the call either fails or gets picked up...

-- 
FP

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Re: [asterisk-users] Voipbuster not ringing, other SIP peers are ringing...

[Francesco Peeters]

Francesco Peeters wrote:

  Francesco Peeters wrote:
  
  
Does anybody else see the same behavior for VoipBuster connections?

When I trace one of the other SIP peers, I see it sends this message:
--
<--- SIP read from 82.101.62.99:5060 --->
SIP/2.0 180 Ringing
Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE
Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl
Contact: 
Content-Type: application/sdp
CSeq: 103 INVITE
From: "**" ;tag=as70e84199
Record-Route:
,
Server: Cirpack/v4.41b (gw_sip)
To: ;tag=00-08168-044b6f36-245cd72c7
Via: SIP/2.0/UDP
***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92
Content-Length: 182

v=0
o=cp10 125193221174 125193221174 IN IP4 82.101.62.66
s=SIP Call
c=IN IP4 194.109.8.2
t=0 0
m=audio 36984 RTP/AVP 8
b=AS:64
a=rtpmap:8 PCMA/8000/1
a=ptime:20
a=sendrecv

<->
--- (12 headers 10 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 194.109.8.2:36984
Found audio description format PCMA for ID 8
Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8
(alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 194.109.8.2:36984
-- SIP/*-089ca9b8 is ringing
-- SIP/*-089ca9b8 is making progress passing it to
IAX2/2104-2287
Scheduling destruction of SIP dialog
'740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 82.101.62.99:5060:
CANCEL sip:0031**...@sip.xs4all.nl SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport
From: "**" ;tag=as70e84199
To: 
Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

--


However when I dial exactly the same from VoipBuster, I see this instead:


--
<--- SIP read from 77.72.169.129:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport
From: "*" ;tag=as1374705a
To: ;tag=120113ac4a54a269af9e2c
Contact: sip:0031**...@77.72.169.129:5060
Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 162

v=0
o=* 1251932194 1251932194 IN IP4 194.221.62.33
s=SIP Call
c=IN IP4 194.221.62.33
t=0 0
m=audio 8958 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20

<->
--- (11 headers 8 lines) ---
Found RTP audio format 0
Peer audio RTP is at port 194.221.62.33:8958
Found audio description format PCMU for ID 0
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 194.221.62.33:8958
-- SIP/-089dc538 is making progress passing it to IAX2/2104-8077
  == Connect attempt from '127.0.0.1' unable to authenticate
Scheduling destruction of SIP dialog
'1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com' in 6400 ms
(Method: INVITE)
Reliably Transmitting (NAT) to 77.72.169.129:5060:
CANCEL sip:0031**...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport
From: "**" ;tag=as1374705a
To: 
Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
--

As you can see, there are different packets being sent, and in the 2nd
case, there is no "is ringing" message, which is rather irritating...

Any suggestions would be appreciated...

TIA
  

  
  BTW: I am talking about the ringtone the caller should hear... The other
side is ringing, and calls are established just fine, but it is very
irritating to hear nothing until the call either fails or gets picked up...

  

NM! Found out this only happens on a single extension, and that one was
using IAX... Changed it to SIP as well and got ringing there too!

-- 
FP



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Re: [asterisk-users] my kernel is dazed and confused

[Francesco Peeters]

Dr. Michael J. Chudobiak wrote:
> Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason 
> a0 on CPU 0.
> Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely 
> on the PCI bus.
> Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue
>
>
> Would my Digium TDM410P cause an NMI, or is my computer failing?
>
> - Mike
>
>
>   
Googling for the error seems to indicate a possible kernel bug... Are
you using Ubuntu or Debian?...


-- 
Francesco

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Re: [asterisk-users] automon => *1 "one touch recording"

[Francesco Peeters]

Joseph wrote:
> On 12/08/09 11:11, Jared Smith wrote:
>   
>> On Mon, 2009-12-07 at 19:39 -0700, Joseph wrote:
>> 
>>> After pressing "*1" console is not showing anything indicating that the 
>>> call is being recorded:
>>>   
>> I find that I often have to adjust the "featuredigittimeout" setting in
>> features.conf, as users tend to take their time between the * and 1 keys
>> when turning on automon.
>>
>> --
>> Jared Smith
>> Digium, Inc.
>> 
>
> Well, ;transferdigittimeout => 3 (default is 3 seconds)
> but this does not work or does not take any effect, this feature worked 
> perfectly in Asterisk 1.2
>
> I just tried it, I set:
> transferdigittimeout => 4 
>
> it doesn't work.
>
> I'm using cordless phone and I'm 100% sure that it take me less then 1.5 
> seconds to press "*1" with one finger.
> However, when I tried pressing "*1" using two fingers it worked.
>
> So, it seems to me "transferdigittimeout" setting doesn't work or doesn't 
> take any effect.
>   
>   
Hmmm... That would possibly also explain why I always succeed in doing
*2 xfers, and my wife always fails... I always have 2 fingers on those
buttons, and she is the single-finger-typing-kind'o'gal...

Weird though that unattended (##) xfers DO work for her as well...

--FP

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Re: [asterisk-users] iphone client app

[Francesco Peeters]

Alex Samad wrote:
> On Tue, Dec 15, 2009 at 08:59:34PM +0100, Benny Amorsen wrote:
>   
>> Gavin Spurgeon  writes:
>>
>> 
>>> iSip (£2.39)
>>> http://itunes.apple.com/gb/app/isip-push-service-formerly-sipphone/id298202722?mt=8
>>>   
>> I have been very impressed by the audio quality from iSip, at least from
>> the "other end" so to speak. It shares the basic flaw of not being able
>> to run in the background with every other iPhone app. They try to
>> 
>
> can't you use backgrounder ?
>
>   
He probably could, but that is assuming he's jailbroken his phone... Not
everybody sees a need to do that, though backgrounder by itself would be
a very good reason to do it...

Best,

--FP

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Re: [asterisk-users] Asterisk recieves "11" when pressing "1" from SIPphone

[Francesco Peeters]

jonas kellens wrote:
> [Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid
> extension '11', but no rule 'i' in context ...[snip]...
>
> When testing IVR and pressing "1" from my Grandstream SIP-phone, the
> above message is printed on the Asterisk CLI.
>
> How come Asterisk receives my "1" as "11" ??
>
> Settings in my SIP-phone are :
> Send DTFM : via RTP(rfc2833) & via SIP INFO
>
> [Dec 31 10:45:40] WARNING[17928]: pbx.c:2518 __ast_pbx_run: Invalid
> extension '33', but no rule 'i' in context ...[snip]...
>
> Same problem when pressing "3"...
>
> Thank you.
>
> Jonas.
It may be me, but it looks like Asterisk correctly interprets the
information, as the phone is configured to send both via RTP (once) and
SIP INFO (twice).
Your config tells the phone to send the digits twice, so Asterisk sees
them twice... 1 twice makes 11, 3 twice makes 33!

Try changing the phone's config to only use either RTP *or* SIP INFO...

Good luck!

--FP


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Re: [asterisk-users] Please remove me from the mailing list.

[Francesco Peeters]

Steve Totaro wrote:
> read your posting and it will tell you haw to remove yourself.
>
> On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean  > wrote:
>
> Can I be taken off the mailing list please.
>
> Thanks.
> rick
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
And a proper mail client will also parse the headers and provide
unsubscribe information/buttons based on that...
--FP

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Re: [asterisk-users] Please remove me from the mailing list.

[Francesco Peeters]

Dan Journo wrote:
> I've never seen that in Outlook. What client do you use?
>
>   
Lately I have been using Thunderbird with an RFC2369 header plugin.

--FP

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Re: [asterisk-users] Please remove me from the mailing list.

[Francesco Peeters]

Rick Green wrote:
> On Thu, 7 Jan 2010, David Gibbons wrote:
>
>   
>> Yes, gmail DOES default to top posting, because bottom posting is silly 
>> (in general, but especially for a client that hides quoted text (like 
>> gmail)). Top posting is modern. And better. And doesn't make me scroll 
>> through 10 thousand messages and awful rsa keys to get to the message... 
>> FLAME AWAY!!!
>> 
>   This is not intended as a flame...  I just got a gmail account a month 
> ago, and haven't used it but for a single google group and calendar 
> notifications.  This morning, after seeing the above message, I actually 
> hit reply on several messages, and this is what I found:
>
> 1) In every case, gmail presented me with the entire text of the message 
> in the compose window.  There was NO indication of 'hidden' full-quote. 
> Yes, the cursor is initially placed at the top of the window.
>
> 2) The 'Daily Agenda' mails I get from Google Calendar arrive in some kind 
> of rich formatting, but right at the top of the composer window is a small 
> unobtrusive link labelled ' makes deleting the unnecessary text trivial.
>
> 3) Plain text email arriving from a friend's android/gmail device are 
> displayed in plain text already.
>
> 4) I searched thru the settings dialog, and I found nothing where I had 
> explicitly told it to include the text in a reply, or to show or hide that 
> text.  I DID specify that 'plain text' was to be my default outgoing 
> format.
>
>IMHO, top-posting isn't the problem, but just an obvious symptom of the 
> real problem, which is failure to edit/strip the quotes to the bare 
> minimum.  When a thread gets hijacked by top-posters, who bang out their 
> thoughts without even scrolling down to see all the garbage below, another 
> problem also becomes apparent, and that is the failure of many MUAs to 
> honor 'sigdashes', which is the convention of preceeding your sigfile with 
> a line that is 'dash dash space '.  A compliant MUA will strip that 
> line and everything after it when quoting for a reply or forward.  Note 
> for the list admin:  Please preceed your message-footer with a sigdashes 
> line!
>
>   
And to add on to this: aside from whether you think it is silly or not,
there are:
1) RFC's
2) List rules

And when both of those tell you to bottom-post, then who are you to
decide otherwise, just because you think it is silly?
Well, maybe I think it is silly that I cannot hit you in the face
everytime you say "I", would you allow me to hit you, or would you
protest and demand I keep to the rules that tell me I can't do that?

Civility demands I keep to the rules and do not hit you in the face.
The same civility demands you keep to the rules as well and do not
top-post! Is that *really* so hard?

Just because Microsoft and others decide to place the cursor at the
wrong position doesn't mean you have to be a mindless herd-animal and
follow that incorrect behavior!

Please people, stop these totally pointless discussions and get back
on-topic!...

PS: I did not have to cut anything, thanks to Rick using the
dash-dash-space convention, and Thunderbird honoring that convention.
PPS: Top or Bottom posting does NOT change anything about the fact you
should  stuff that is no longer relevant

Just my €0.02!
-- 
Francesco

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Re: [asterisk-users] You won't help me anymore?

[Francesco Peeters]

ABBAS SHAKEEL wrote:
> why don't you post your question
>
> On Sun, Jan 10, 2010 at 4:42 PM, hadi motamedi  > wrote:
>
>
>
> On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra  > wrote:
>
> Sunday, January 10, 2010, 11:24:22 AM, hadi wrote:
>
> > You are not willing to help me anymore ?
>
> Why do you think this?
>
> --
> Best regards,
>  Gergomailto:csi...@gmail.com
> 
>
>
>
>  
> Thank you for your reply . I am facing with callerId problem on my
> sip inbound calls , so I strongly need your technical help . Can
> you please help me ?
>
>  
>
>
Yes, post your question clear and consicely, include all relevant
information and snip all unneccessary history.

Note that: no reply != not wanting to help...
It *is* obviously possible people just do not KNOW the answer!... (Oh
what shock and horror!!!)

-- 
Francesco

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OT: Date off??? (was Re: [Asterisk-Users] News From Astricon)

[Francesco Peeters]

On Thu, September 23, 2004 8:41, Steve Totaro said:
> link doesnt work
>
>
You may want to take a look at your system settings, as I think it
unlikely that this e-mail has been in transit for approx 8 months...  ;-)

-- 
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Re: [Asterisk-Users] ENUM NL dead ?

[Francesco Peeters]

On Tue, June 7, 2005 10:14, Michiel van Baak said:
> Hi Florian,
>
> On 09:05, Tue 07 Jun 05, Florian Overkamp wrote:
>> Hi Michiel,
>>
>> > -Original Message-
>> > I been searching on the wiki and google for ENUM in NL.
>> > All I could find were some docs from the Dutch Financial
>> > Department about taskforces and plans. But it all links to
>> > dead pages and no-longer-connected phone numbers.
>> > Is there anyone who knows some more about Dutch ENUM stuff ?
>> > If it's all dead, anyone interested in setting it up
>> > together with me ?
>>
>> We have been trying to get it alive but have been unable to do so just
>> by
>> ourselves. Some cooperation with the dutch government (dgtp) and domain
>> registry (sidn) will be required. Care to join forces ?
>>
>> Florian
>
> Count me in.
> I really believe we can use this.
> We should contact eachother about what we need to do now.
>
> If others here are interested, reply to the list or to me :)
>
> Michiel

I currently do not have time (or money) to spare, but you *do* have my
support!

Hopefully I'll have some more time in the future, so if you need help, you
can always contact me to see how my free-time situation is at that time

-- 
Francesco Peeters (also from NL)

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Re: [Asterisk-Users] Portable USB headset for VoIP

[Francesco Peeters]

On Tue, June 14, 2005 13:04, Paul Mahler said:
> I've bought bunches of these:
> http://www.tigernetcom.com/products_USB_100.html
>
>
> they work great. Very handy.
>
> Paul
>
> [EMAIL PROTECTED]
>
>
> I'm trying to find a voip-suitable USB headset (I.E. headphones +
> microphone) which I can use with my laptop while I'm traveling and using
> Firefly or another softphone.
>
> I'm currently using a Logitech headset which works well (except the echo
> it
> generates toward the other caller when I turn up the gains too high), but
> it
> just doesn't carry well - in fact, I can't carry it in my laptop case any
> more
> just becuase it doesn't fit and it was getting very beat up.
> I'd like to find something which folds up and is designed for travel.  It
> has
> to be USB sicne I don't have a MIC in (just line) on my laptop.
>
> Any ideas?
>
> -forrest

I actually use a Bluetooth Headset and Bluetooth USB Dongle when
traveling... The one I use can be paired with both my cell-phone and my
computer (though not simultaneously, but I've never had the need for that
yet!)

Whichever device I want to use it with initiates a connect, and once it is
connected, it can be used with that device exclusively until it is turned
off or disconnected manually from the initiating device...

Just my € 0,02 worth!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
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RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?

[Francesco Peeters]

On Tue, June 14, 2005 21:30, Bill McLaughlin said:
>
>

>
>
> There's also the fact that a lot of companies charge LESS for home access
> than for a business, under the assumption that the business will utilize
> it
> more, and/or can afford the higher price.
>

Also home/private access (talking DSL here, not E1/T1!) usually is
overbooked 25:1, whereas business connection are usually between 10:1 and
1:1 rates...

Assuming the 10:1 ratio vs. 25:1, it would mean a company would have to
pay 2,5 times what a customer pays for the ISP to make the same... at 4:1
(another common ratio) that would be 6.25 times as much...

At EUR 75 for a home connection that'd be USD 525 for the business at 4:1
ratio...

I can imagine the home fiber connection will probably be multiplexed to
share a single uplink with x other homes as well, so you'd probably be
looking at similar issues as I mentioned regarding the DSL ...

-- 
Francesco Peeters

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Re: [Asterisk-Users] CellPhone BlueTooth adapater with Wireless Profile ??

[Francesco Peeters]

On Wed, June 15, 2005 21:27, TC said:
> All
> Any body know of a generic bluetooth adpater for the
> universal 2.5mm headset jack on a cell phone that supports
> the wireless profile *NOT* the headset profile
> I know jabra has the A210
> http://www.jabra.com/JabraCMS/NA/EN/MainMenu/Products/Accessories/JabraA210/
> JabraA210
> but it only support the headset profile ..
>
> I am trying to shoe horn my current braindead cell into DocknTalk
> http://www.phonelabs.com/prd05.asp, and the BlueTooth interface requires
> a WirelessProile
>

The 2.5 mm jack is a sound only device which is wired to the
speaker/microphone part of the phone. Therefore you will NOT be able to
make it pass data.

I therefore doubt there'll be *any* BT device with a 2.5mm jack that'll do
*anything* but the headset profile...

Just my € 0,02 worth...

-- 
Francesco Peeters

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Re: [Asterisk-Users] This mailing list is being spam filtered on my site.

[Francesco Peeters]

On Thu, June 16, 2005 3:26, Gary Guthary said:
> Sorry if this not the right place to post this  BUT...
>
> Since May 31st, ALL of these user list messages have been filtered by
> "spamassassin" running on my Linux box. - Claim to be listed in "Bayes" as
> spam. - Have no clue why this is happening.
>
> Luckily, "spamassassin" sent the messages to the "probably-spam" folder on
> the Linux box & I was able to retrieve them.
>
> If anybody else is having this problem **AND** us using "procmail" (along
> with "spamassassin") on a *NIX box, put the following three lines in the
> top
> of your ".procmailrc" file:
>
> :0
> ^To:[EMAIL PROTECTED]
> ${DEFAULT}
>
> Note: - That's a zero in the first line.
>
> This will allow delivery until somebody can figure out why this "Bayesian"
> filtering is happening and can get it stopped.
>
> If anybody wants to contact me "off-list" to discuss, email:
> [EMAIL PROTECTED]
>
> Gary Guthary
>
>

Check whether Bayes filter is set for auto-learn. It has somehow aquired
enough keywords from this list to mark the emails from here as SPAM. I do
not know which filter you use, but the SpamAssassin built in Bayesan
allows for 'HAM' (ie NON-SPAM) mails to be learnt... Try collecting a
weeks worth of list mails and then have the filter scan them (look for
sa-learn) as 'HAM'...

Good luck!

-- 
Francesco Peeters

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Re: [Asterisk-Users] This mailing list is being spam filtered on my site.

[Francesco Peeters]

On Thu, June 16, 2005 12:34, Andrew Kohlsmith said:
> On Thursday 16 June 2005 02:01, Francesco Peeters wrote:
>> Check whether Bayes filter is set for auto-learn. It has somehow aquired
>> enough keywords from this list to mark the emails from here as SPAM. I
>> do
>> not know which filter you use, but the SpamAssassin built in Bayesan
>> allows for 'HAM' (ie NON-SPAM) mails to be learnt... Try collecting a
>> weeks worth of list mails and then have the filter scan them (look for
>> sa-learn) as 'HAM'...
>
> I too am seeing this and I've been using SA for YEARS.  I've been trying
> to
> train it but some of the messages to this list just do not want to be
> classified as non-spam.  I"m trying to get them to come out clean without
> resorting to a whitelist.
>
> -A.

I too have SA running on my FC3/Postfix server, and it only picks out the
occasional post, so I'm not complaining (yet!)

First thing I did though was make sure it did not autolearn, and set up a
HAM and SPAM alias to send identified e-mails to for SA to learn from...

I have 31 mails in the HAM box and 1100+ in the SPAM box... (I also have
SPF and Grey-listing on, which catches a good amount of spam, as do the
sorbs and monkey lists)

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
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[Asterisk-Users] Grandstream 100 pricing question

[Francesco Peeters]

Hi All,

I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the
pricing for these in Europe, so I'd like to hear from people here whether
that is a reasonable price for them?

TIA & BRgds

-- 
Francesco Peeters

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Re: [Asterisk-Users] Grandstream 100 pricing question

[Francesco Peeters]

On Tue, June 21, 2005 23:07, Kristof Hardy said:
> Francesco Peeters wrote:
>> I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the
>> pricing for these in Europe, so I'd like to hear from people here
>> whether
>> that is a reasonable price for them?
>
> Prices I know are around 99 EUR, incl VAT. But if you ask me, depending
> on how many you need, you should take a look into the GXP-2000. (+- 125
> EUR incl VAT)
>
> The difference in quality (and features) between these is big enough to
> justify the difference in price.
>
> Cheers..

Thanks for the info... I am considering getting these to experiment with,
so I can do some testing *before* I actually get in to the real thing. The
cheaper the test period, the better, so 2 of these (which later can be
reused in less used area's) look pretty interesting...

The shop I saw these also sells - pretty cheap -  little devices (forgot
the name, they look like a translucent blue ice-hockey puck) that do SIP
conversion for analog telephones or PBX extensions. (I am thinking
migration period here: first connect one of those to each of the two PBXs
as an extension, so you can use it to 'dial' in to the * server.

Then my migration plan - after initial testing - would then look like this:
1) Install * on both sites
2) IAX2 link
3) 'SIP-puck' on both PBX's and connect these to the * servers
(at this point all users can talk to eachother over landlines *and* SIP)
4) Start migrating inidividual users to SIP & *
5) retire old PBXs

-- 
Francesco Peeters

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Re: [Asterisk-Users] Grandstream 100 pricing question

[Francesco Peeters]

On Wed, June 22, 2005 10:02, Ming-Wei Shih said:
> On Wed, Jun 22, 2005 at 08:35:58AM +0200, Francesco Peeters wrote:
>> The shop I saw these also sells - pretty cheap -  little devices (forgot
>> the name, they look like a translucent blue ice-hockey puck) that do SIP
>> conversion for analog telephones or PBX extensions. (I am thinking
>> migration period here: first connect one of those to each of the two
>> PBXs
>> as an extension, so you can use it to 'dial' in to the * server.
>
> Is this a webshop in Europe? Care to share the URL?
>
> Regards
>
> Ming-Wei

Sorry, it's a real shop in Zoetermeer, the Netherlands, I visit sometimes...

They do have a website, but I doubt whether they have all their stuff on
there (they often have these small lots of special kit in the store) and
whether they have a webstore...

http://www.telec.com



-- 
Francesco Peeters

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[Asterisk-Users] Is this server sufficient?

[Francesco Peeters]

I've tried to find some details on the wiki, but was unable to get a
satisfactory result, so I am asking here:

I have a Linux (FC3) box with these specs:

vendor_id   : AuthenticAMD
cpu family  : 6
model   : 3
model name  : AMD Duron(tm) Processor
stepping: 1
cpu MHz : 797.388
cache size  : 64 KB

MEM: currently 256, looking to upgrade to 512/768 (depending on available
sticks)

HDD: 80 GB

It is currently doing File/Printer serving.

Ideally I'd want it to do Asterisk (2 ISDN BRI & 8 phones), File/Printer
server on a home network (3 clients) and some light SMTP (< 100 emails a
day)

Is this machine sufficient for the task? (Ignoring the fact it needs
either a multi-BRI card or 2 single BRI cards to be able to connect to the
PSTN )

TIA!

-- 
Francesco Peeters

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RE: [Asterisk-Users] Is this server sufficient?

[Francesco Peeters]

On Wed, June 22, 2005 13:39, Dean Collins said:
> As an asterisk server it is more than fine but asterisk prefers to be a
> standalone machine.
>
> You would have a lot less issues if you had 2 machines, one handling
> file serving, SMTP and one dedicate machine for asterisk.
>
> Voice isn't very tolerant of interrupts.
>
>
> Cheers,
> Dean
>
>

I am aware of that, but the server is doing nada 99.9% of the time right
now, so I'd rather give up the other functionality and have it to * rather
than the other way round!  ;-)

I thought I'd give it a try with * and see whether we have issues when the
rare SMTP/SMB access occurs (and deal with it then!)

I just wanted to be sure that the machine is sufficient to do * and then
some...  ;-)

I think I'll use the upcoming vacation period to go play with it then!  :-D

Cheers!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
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RE: [Asterisk-Users] Grandstream 100 pricing question

[Francesco Peeters]

On Wed, June 22, 2005 17:17, Francesco Peeters said:
> On Wed, June 22, 2005 15:48, [EMAIL PROTECTED] said:
>> [EMAIL PROTECTED] wrote:
>>> Thanks for the info... I am considering getting these to
>>> experiment with,
>>> so I can do some testing *before* I actually get in to the
>>> real thing. The
>>> cheaper the test period, the better, so 2 of these (which later can
>> be
>>> reused in less used area's) look pretty interesting...
>>
>> I have several Grandstream BT101's that are slightly used and I'm
>> looking to sell, if you're interested.
>>
>> Barton
>>
>>
>
> Where are you located? What price? Shipping?
> (Always interested in deals!)  ;-)
>
> --
> Francesco Peeters
> 
> GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
> If your program doesn't recognize my signature, please visit
> http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
>

Ehrm... Perhaps you'd better respond off-list though!  ;-)

--FP
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RE: [Asterisk-Users] Grandstream 100 pricing question

[Francesco Peeters]

On Wed, June 22, 2005 15:48, [EMAIL PROTECTED] said:
> [EMAIL PROTECTED] wrote:
>> Thanks for the info... I am considering getting these to
>> experiment with,
>> so I can do some testing *before* I actually get in to the
>> real thing. The
>> cheaper the test period, the better, so 2 of these (which later can
> be
>> reused in less used area's) look pretty interesting...
>
> I have several Grandstream BT101's that are slightly used and I'm
> looking to sell, if you're interested.
>
> Barton
>
>

Where are you located? What price? Shipping?
(Always interested in deals!)  ;-)

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
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Re: [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!

[Francesco Peeters]

On Mon, June 27, 2005 0:15, harry gaillac said:
> I agree you.
>
> Does asterisk (Digium) project provide a good
> documentation ?
>
> Does Asterisk Handbook has been released ?
>
> When developpers improve Asterisk where are you
> looking for help, mailing list, wiki, asteriskdocs,
> ...-:(
>
> It's the job to all Asterisk developpers/users to
> provide docs on Asterisk.org.
>
> In fact Digium host both a open project and a
> commercial site.
> www.digium.com. 86299   IN  A
> 69.16.138.164
>
> www.asterisk.org.   86254   IN  A
> 69.16.138.164
>
> Asterisk/Digium don't provide docs so you have to pay
> for help or waste time to google.
>
> Open source community as often criticize enterprises
> like Microsoft, Cisco, 
> However these ones pay R&D.
>
> May be enterprises like Digium an others want to earn
> money with works of open source community
>
> I tell to these enterprises you want to earn money do
> like Microsoft
>
> Harry from France
>
>
>> Je soupconne que votre application fonctionne tres
>> bien, mais que vous RENDEZ VONLONTAIREMENT
>> L'INSTALLATION DIFFICILLE. Pour que tout le monde
> soit
>> oblige de vous contacter et de vous payer: C'EST UNE
>> PRATIQUE TRES COURANTE DANS LE MONDE GPL, mais C'EST
>> MALHONNETE!
>>
>
>> > Jeam Marie K
>
>
Me thinks you haven't grasped the idea of OSS...

If you have a problem with it, do something about it! (Dig in the project
and write some good documentation!)

I am however surprized you say there is no documentation... Maybe you mean
no documentation in French? (It's only in France that it is obligatory to
supply French versions of everything you publish!) a quick google for
asterisk gives me a plethora of web-sites with documentation, how-to's,
etc.!

And as Areski said: if you have an issue, first contact the person(s)
involved. Only if they do NOT reply, solve or otherwise take satisfactory
action, you should consider going to the community like this! In the time
I have been on this list, I have seen many helpful (unpaid) posts by him,
and I think you have - unjustfully as far as I can tell - damaged his
reputation by this manner of behavior!

My EUR 0,02...

-- 
Francesco Peeters

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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Francesco Peeters]

On Mon, June 27, 2005 13:04, Andrew Kohlsmith said:
> On Monday 27 June 2005 02:45, Marcel van Kaam, Fonetica wrote:
>> I think by now everybody knows that LiveVoip went down, bankrupt etc
>> So please stop nagging about it and move on to some topics that really
>> matter.
>>
>> If you want to discuss LiveVoip, get all together in a restaurant, eat,
>> drink and nag and wine about it as much as you want. But do it there and
>> not here.
>
> I've never understood this -- people are having a decent discussion.
> There's
> no flaming, there's no bashing.  Sure it's offtopic but it'll die within a
> few more days...  Why snuff it?  I am positive we're all not
> geographically
> close to discuss this in a restaurant, and setting up an entirely new list
> is
> silly.
>
> So I ask you -- what should people do?
>
> -A.

*Shrugs*  Seen it, been there... This happens on all lists at some point
in time... Several lists I am on have already created an OT or TALK list
besides the main list...

Whenever that happens, I just subscribe to the 2nd list and customize my
procmail recipes to toss it in the same folder...  :-/

And then live goes on...

-- 
Francesco Peeters

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Re: [Asterisk-Users] Dialogic D/300pci-E1 card

[Francesco Peeters]

On Mon, June 27, 2005 21:54, Eric Wieling aka ManxPower said:
> Florin Mandache wrote:
>
>> It is any way to use this card with Asterisk, and if yes, HOW to ?
> According to this: http://asterisk.org/index.php?menu=hardware  no.
>
>
>

According to that page this one is:
D/300JCT-1E1E1 + 30 voice

Don't know whether it is similar/the same tho...

-- 
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Re: [Asterisk-Users] Dialogic D/300pci-E1 card

[Francesco Peeters]

On Mon, June 27, 2005 21:04, Florin Mandache said:
> It is any way to use this card with Asterisk, and if yes, HOW to ?
>
> Thanks.
>

According to this
<http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware> there
is a Dialogic driver available from Digium... Best to ask them directly
whether these support you card as well...

As an aside, please start a NEW email when you have a NEW item/question...
Unlike what M$ makes you believe, replying to an existing email does NOT
start a new thread, even when you change the subject...

Your question is now somewhere in the middle of the LiveVOIP thread, and
people who have marked this thread for immediate delete will NEVER see
your question, nor any replies to it (My reply will be part of that thread
as well, BTW)

Have fun!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
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[Asterisk-Users] Just let whiners whine... Please?

[Francesco Peeters]

There'll be whiners on all lists, and no list is ever going to be pleasing
everybody... (And everyone should decide for themselves whether or not
they belong to that group, as I will not be mentioning names! I'm sure
there are those who'll think *I* am a whiner...)

So at some point just let it be and continue the thread. Keeping on and on
about it only increases noise to post ratio's...

Just my EUR 0,02!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
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[Asterisk-Users] Is the Siemens SX353 (DECT) Base Station compatible with *?

[Francesco Peeters]

Hi,

I am just new to Asterisk, and would like to experiment a little before
implementing and investing...

We have a Siemens SX353 DECT Base Station/desk telephone with USB
connection for PC, I know there are (beta) I4L drivers for this device
available.

Does anybody know whether this device would be usable as a temporary ISDN
device for * until we are ready to push it and invest in some proper BRI
interface cards?

TIA!

-- 
Francesco Peeters

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Re: [Asterisk-Users] Why should I answer a Newbie question, there thick!

[Francesco Peeters]

On Wed, March 2, 2005 10:38, BCS Support said:
> It would be nice just for once to actually use a mailing list with people
> who are a little more sympathetic to the fact that your not a rocket
> scentist or molecular biologist and that you might actually need some
> help,
> without being made to feel like your completely useless and should be
> cleaning toilets for a living.

> Yes I have spent hours researching on Google, but what may take me 3 days
> to workout, wading through pages of out of date information, can normally
> be answered by some with a little experience in seconds.


Ah, but the issue here is that your questions seemed to indicate you
haven't even read the basic information on the site iself (i.e. the
manual), as even I (just started actively looking in to Asterisk 4 or 5
days ago) was able to find the answers to the questions you asked...

I do not recall the exact questions, but I do remember agreeing with the
conclusion that you apparently hadn't done any actual research, based on
the questions you asked...
Had I had some more time on my hands at that time, I would have replied,
but as I am usually very busy, and din't have the answers ready from the
top of my head at that time, I didn't...

I do agree that the reaction was a bit ott, but in the basis correct...
Sorry!

I do wish you good luck implementing *, because I am confident this has a
lot of potential...

God bless!

(PS: Can we please play nice now? I left my flame-retardant gear at the
firestation when I quit as a volunteer!)
-- 
Francesco
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Re: [Asterisk-Users] newbie questions

[Francesco Peeters]

On Mon, March 7, 2005 22:50, Brian Nehring said:
> I've read through a good amount of documentation on voip-info.org, but
> hadn't found a solution, so I thought this list might help. I'm not

> just give it username/password and point it at a SIP proxy. However,
> as far as I can tell it isn't able to register, or it's not listening
> to Asterisk... hard to tell really.
>

> On Mon, 7 Mar 2005 14:32:57 -0700, Wiley Siler <[EMAIL PROTECTED]>
> wrote:
>> Ignore the error if it isn't messing anything up.
>>
>> Check out the Wiki here
>> http://www.voip-info.org/tiki-index.php?page=Asterisk
>>
>> A search of X-lite here also yields proper setup info for the softphone
>> to Asterisk connection.
>>
>> The archive of this list can be search via google by entering...
>> site:lists.digium.com 
>>


>From your reply above, it is not clear to me whether you even read the
reply, or tried what was suggested?

Searching the Wiki for 'X-lite conf' gives a link to the X-lite page,
which links to the xten page, which has a link to the X0lite and Asterisk
configuration PDF file...

Took me 30 seconds...

If you *did* follow the PDF (which I cannot tell from either your initial
post or your reply), then maybe the X-lite specific config data and logs
would be helpful?...

-- 
FP
Also an *-n00b, just very skilled at Googling...
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Re: [Asterisk-Users] Skype - Bandwidth

[Francesco Peeters]

On Mon, March 14, 2005 22:11, Paulo Sequeira said:
> César Davi Ávila do Nascimento wrote:
>> Talk about skype is forbidden, but to be impolite is allowed...
>> Great list!
>>
>> Regards
>>
> César, I'm sorry your experience here has been very bad, but I'd like to
> make some clarifications in order to avoid you further frustrations with
> this or even some other lists you may want to be in.
>
> The harsh answers you've got are the result of violating several
> politeness rules that usually apply to mailing lists (not only this
> one), and it will be very good for you to learn them, (or a last be
> aware that they may exist).
>

> So, if you feel people have been impolite with you is because you've
> been (inadvertently) very impolite to the list in the first place. I
> don't have at hand any links to give about places you could learn more
> of theese and other etiquette rules, but you should care about it.
>
> Hope this helps you.
>

A few other things to keep in mind when posting to maillists:

1)  is you friend! SNIP away anything not related! (especially sigs
and CERTAINLY when replying to DIGESTS!!!)

2)
A: Because you lose the logic sequence of the thread
Q: Why is top-posting bad?

Seriously: Don't top-post... Post in between or at the bottom, after
you've snipped anything unrelated... The fact that your mailclient puts
your cursor at the top of the quoted text does not mean you *have* to type
there... Cursor keys and mice were invented for just that!  ;-)

3) When asking questions like these, tell people what you know already and
what you have tried... Let them see you tried doing your homework yourself
before asking your big brother!

4) Get a thick skin! English is not everybody's native language, and this
can make replies seem harsher than they were meant.
If you *did* have it coming, you'll be better prepared anyway!  

5) Get a flame-retardant suit! Flames are a matter of fact on maillists,
even the most polite ones have'em!

But most of all: Read the posts, get the info you need, but also take in
the extra info that helps you understand the context of the post!!!

Have fun!

-- 
Francesco
  * n00b, but maillist veteran!  ;-)
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Please do not use 'reply' for new threads? (was: Re: [Asterisk-Users] Egytpian call progress frequencies and cadences (second request))

[Francesco Peeters]

On Wed, March 30, 2005 13:04, Ezabi said:
> Hi,

> Ezabi

Ezabi, (a.o.)

I am assuming that you aren't using a threaded email reader, as you would
be aware of what replying to a message in order to start a new thread -
which I am assuming you did, judging from the results - would do to the
threading if you were...
(I am not holding that against you , just drawing conclusions!)

Please (to all) start a *new* message when you want to start a new
thread... Replying to an existing message and changing the subject will
*not* start a new thread. Threading (in proper clients at least) is based
on special information in the message headers, which does not get altered
by changing the subject.

The result is that the 'new' thread gets weaved in to the existing one,
with unwanted results for both threads...

TIA!

PS: Please, no replies on whether this should be done on or off list. I
happen to think it belongs on-list for the education of all. A
'discussion' on this subject will only server to pollute both threads even
further, but will only end, as previously, in agreement to disagree... 
;-)

BRgds

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
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Re: [Asterisk-Users] job offer - in german only

[Francesco Peeters]

On Wed, March 30, 2005 16:49, Florian Buzin said:

> Interesse?
> Ihre aussagefähigen Bewerbungsunterlagen senden Sie bitte per E-mail
> an [EMAIL PROTECTED]:
> 
> 76137 Karlsruhe

Leider ein bißchen zu weit weg für mich!

-- 
Francesco
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Re: [Asterisk-Users] Online forums vs email list...

[Francesco Peeters]

On Thu, March 31, 2005 20:07, Chuck Bunn said:
> Hi,
>
> I am curious just what the advantages of an email forum over an online
> one.

off-line reading for one! I can sync my email and read it anywhere
anytime, with or without Internet access!

> Thanks for the search tip, but it is still an annoying way to
> search. Some advantages to an online forum might be the ability to see
> the number of views on a thread, number of answers on a thread without
> counting the stupid emails, moving a thread to a sticky position to use
> as a FAQ, etc.

Use a threaded email client... Helps a lot!

> Do not get me wrong I am a big command line fan with
> Linux but sometimes GUI's are easier, I am not one of those purist who
> only use a MAC, Windows or Linux - I use what I think is the best fit
> for a job...
>

me too... for this type of thing maillists are IMHO the best fit, but to
everybody his/her own!  ;-)

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
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Re: [Asterisk-Users] Are there online forums instead of this

[Francesco Peeters]

On Thu, March 31, 2005 21:03, [EMAIL PROTECTED] said:
>
> There are many on-line forums related to asterisk, including
> www.asteriskforum.com ... the problems is that as long as Digium supports
> this type of "dinosaur era" list, the other communties will not grow.
>
> The responsbility is Digium and if they truly care about building this
> community, they should endorse one of the forums, or set up their own, and
> turn this list off.
>
> There is no need to set up another one, as their are already many people
> who have done that.  You can set up 100 of these, but if Digium does not
> support it by using it, then there is is a problem
>
> I don't believe that Digium is as narrow minded as suggested by the poster
> below.
>
>
>>
>> This subject has come up about every two months for the past year
>> or more, and the exact same answers still apply. If you want a forum,
>> go set it up; it ain't going to happen at digium.
>>
>> Others have already set up forums; go find them and use those.
>>

This is a discussion I see popping up every other month or so on virtually
every mailinglist I am on... Some people prefer web forums, but in the end
- at least on the lists I am on - a vast (often mostly silent )
majority prefer mailinglists...

- It's much easier to just leech (silent majority anyone? ;-) ) on
mailinglists, and absorb information.
- Mail can be read anywhere, with or without web access
- Mail reading is usually faster and easier than forum topics
- Mail takes less bandwidth (data plans, etc.)
etc.

So besides disadvantages, there are also advantages to mailinglists, which
for many take precedence over the 'advantages' of webforums.

I do not claim/pretend to speak for everybody on this list, but I *do*
think that others that promote web forums should not do so either...

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
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Re: [Asterisk-Users] Are there online forums instead of this email

[Francesco Peeters]

On Fri, April 1, 2005 3:01, Bruno Hertz said:
> Martijn van Oosterhout <[EMAIL PROTECTED]> writes:
>
>>
>> Ok, basic use case. I today go to a forum and read all the messages.
>> Next day I come along, how do I get a list of all the messages I havn't
>> read in thread order in such a way that if I decide to go somewhere
>> in the meantime, it knows what I've read and what I havn't.
>>
>> I also monitor several other projects all on mailing lists. With one
>> mail box I can monitor six projects in one interface. I don't touch the
>> mouse the whole time. I can whizz through a message every few seconds
>> because every one is in the same font, same colour, same spacing (HTML
>> all disabled). No forum is ever going to compete with that sorry.
>
> This really is a killer argument, and I wholeheartedly agree with that.
> One point comes to mind though, which has been troubling people here
> for some time and where web forums, as much as i dislike them, could
> actually be of use, i.e. partitioning.

Multiple lists can do the same

> As of now, all kinds of stuff is thrown into this list, mostly * related
> but not always, from whatever cards over sipura products and manager api
> to softphone setup and whatever. Now, even if mailing lists were set up
> for particular topics, I think experience tells us that quite a few users
> would come here anyway, and people would have a tough time educating them.
> That's I think the main reason no serious effort is taken in that
> direction.

I do not see the difference with forums... People can still post in the
wrong forum there as well. It is up to the owners and community to point
them to the right forum or list.

I am on a few lists that have been split up, and that works fine as long
as people adhere to one basic rule:
When a post arrives that belongs in a different list, point that out on
the list it arrived on, with a cross-post to the list it belongs on,
quoting the entire message. After that, the original list should ignore
it, and the correct list can pick it up. It's up to the oroginal poster to
ensure he's on the correct list to receive the replies...
(Sounds awfully similar to moving a thread to a different forum to me!)

> On the other hand imagine a forum with subtopics like sipura, softphones,
> zap or whatever. Wouldn't that maybe help to put some load off at least
> the casual reader and poster seeking or giving advice for topics he/she
> specialized in, and maybe even the more active participants? Just a
> thought, and not a bad one imho.
>

Nah, like I said, IMHO it's not different from multiple maillists, as long
as the same rules are applied consistenly...   ;-)

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
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Re: [Asterisk-Users] Re: Are there online forums instead of this email forum??

[Francesco Peeters]

On Fri, April 1, 2005 11:31, Tony Mountifield said:
> In article <[EMAIL PROTECTED]>,
> Bruno Hertz <[EMAIL PROTECTED]> wrote:
>> Andrew Kohlsmith <[EMAIL PROTECTED]> writes:
> I totally agree. I run a local INN server and all the mailing lists I
> subscribe to get turned locally into newsgroup postings in moderated
> groups. When I make a posting, it gets mailed out through a filter to the
> moderator address, which is just the list posting address. Makes handling
> threads a breeze.
>
> I still use trn to read and post too, as I have yet to find anything that
> is as fast to use.
>
> Cheers
> Tony
> --

I very much like - and heavily use - Squirrelmail for OoO access to my
mail... Light, fast, threads, searching, extensions based (Procmail
management!!!) and usable from virtually everywhere I like, when I don't
have my laptop along! (Or when I cannot access my private mail server
directly)



-- 
Francesco Peeters

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Re: [Asterisk-Users] Are there online forums instead of this email

[Francesco Peeters]

On Fri, April 1, 2005 15:10, Bruno Hertz said:
> "Francesco Peeters" <[EMAIL PROTECTED]> writes:
>
>>> On the other hand imagine a forum with subtopics like sipura,
>>> softphones,
>>> zap or whatever. Wouldn't that maybe help to put some load off at least
>>> the casual reader and poster seeking or giving advice for topics he/she
>>> specialized in, and maybe even the more active participants? Just a
>>> thought, and not a bad one imho.
>>>
>>
>> Nah, like I said, IMHO it's not different from multiple maillists, as
>> long
>> as the same rules are applied consistenly...   ;-)
>
> Well, it's easy to say nah if you don't want to think about it. Again, I
> favor mailing lists too, and all would be OK for me if ppl here weren't
> already complaining about volume and stuff.
>


I think you took my Nah a itsy bit out of context there...  ;-)

Your points about Subscription and Topic choice are valid, and the ideal
would be a forum that would behave like a maillist... I.e. post and read
either on web or mail, and it'll get where it should be... It is difficult
though.

Totally OT:
I have been looking at this as a plugin for my own (non tech)
WebBBS/Forum, but the problem is that not all clients adhered to the
'references' SMTP-header behavior at that time...

I haven't looked at that for a while now, so that may have changed...

(Besides that I unfortunately do not have time to write such an extension
at this point in time... It'd be an interesting challenge tho...)

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RE: [Asterisk-Users] Are there online forums instead of this email

[Francesco Peeters]

On Fri, April 1, 2005 18:07, Tim Bass said:
>
> The purpose of all support groups, forums and lists are to support users
> who
> need help, not to insult them. When people make mistakes and post in the
> wrong place, we don't insult them or call them names, and if a moderator
> did
> not have time to move the post in a wrong forum to another one, that is
> not
> such a serious issue, vice having a bit of profanity laced discussions
> with
> women and students in the community.
>
Women and students are used to more (and can take more) than you seem to
give them credit for!

Do you also want to prevent women and students from reading newspapers
(lots of  profanity, violence, etc. in them!) or watching TV?

Let everybody decide for themselves what they will put up with, and do not
try to enforce your idea of what is right or (un)suitable on those that do
not want it...

Another little thing:

Because it disturbs the logical flow of a discussion
Why is top posting in an email bad?



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RE: [Asterisk-Users] Are there online forums instead of this email

[Francesco Peeters]

On Fri, April 1, 2005 22:04, Tim Bass said:
>
> Dear All,
>
> Thanks for your help :).   As I mentioned earlier, I don't have any issues
> with email, except that I thought most advanced communities had moved past

>
>

NOW SHUT UP TROLL!!!

If all you can do is insult people then go troll some other list!

People like you are one of the few things that can really make me angry!

A message for you d00d: newer is not always better! Email is a point in
case: It's been there for a real long time, and will probably survive you
on the 'net!

Sometimes the simplest tools are the best ones!

If people don't agree with you, live with it, don't harass or insult them!
You are going against each and every of your sacred forum community rules
on this maillist... How would people out there respond if they knew one of
their great and holy moderators (or whatever you're out there) is
misbehaving like a spoiled little brad on a maillist here, just because he
can't have things his way!

GROW UP!

-- 
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Re: [Asterisk-Users] Re: Are there online forums instead of this email

[Francesco Peeters]

On Fri, April 1, 2005 23:37, Tom Ivar Helbekkmo said:
> Francesco Peeters wrote (to Tim Bass):
>
>> NOW SHUT UP TROLL!!!
>
> Hear! Hear! And the rest of us should now just ignore Tim Bass. Please!
>
> -tih
> --
> Don't ascribe to stupidity what can be adequately explained by ignorance.

I just created a procmail recipe to put any post he is in into the
bitbucket...

As long as everybody else ignores him, I won't be seeing anything from him
anymore...

-- 
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Re: [Asterisk-Users] Are there online forums instead of this email

[Francesco Peeters]

On Fri, April 1, 2005 23:50, John Novack said:
>
>
> Francesco Peeters wrote:
>
>>On Fri, April 1, 2005 22:04, Tim Bass said:
>>
>>
>>>Dear All,
>>>
>>>Thanks for your help :).   As I mentioned earlier, I don't have any
>>> issues
>>>with email, except that I thought most advanced communities had moved
>>> past
>>>
>>>
>>
>>
>>
>>>
>>>
>>
>>NOW SHUT UP TROLL!!!
>>
>>If all you can do is insult people then go troll some other list!
>>
>>People like you are one of the few things that can really make me angry!
>>
>>A message for you d00d: newer is not always better! Email is a point in
>>case: It's been there for a real long time, and will probably survive you
>>on the 'net!
>>
>>Sometimes the simplest tools are the best ones!
>>
>>If people don't agree with you, live with it, don't harass or insult
>> them!
>>You are going against each and every of your sacred forum community rules
>>on this maillist... How would people out there respond if they knew one
>> of
>>their great and holy moderators (or whatever you're out there) is
>>misbehaving like a spoiled little brad on a maillist here, just because
>> he
>>can't have things his way!
>>
>>GROW UP!
>>
> A clear case of the pot calling the kettle black
>
> One more rude social misfit to put in the "discard" filter
>
> JN
>

I beg to differ... (But then again, I'd be biased!)
I have politely replied several times, stating my arguments, but there is
a point one goes too far! TB crossed that line a bit back already, and I
still contained my anger, but acting as if all of us mail list
users/protagonists are redneck zero-IQ morons just had to trigger a
response... (And he must have known it! He has been doing nothing but
Trolling since the first moment his forum propaganda - which, for the
record, I'll admit he did NOT start, but gladly continued - was shot down)
I happened to be the first to respond, but others have already expressed
agreement on and off list...

And even in my response I refrained from profanities, something TB
apparently could not, judging from the other post I just saw appear on
list!

Or were you referring to yourself? As calling someone you do not know a
'rude social misfit' can only automatically qualify you for the same
label... Will you put yourself in the discard filter too?

I'll gladly shut up about this, and would have not said anything about
this anymore, if not for your - totally off the mark - post...

*I* don't go around joining maillists and then almost immediately start
stressing how old-fashioned and dinosaurian they are (Why join one anyway
if you detest them and prefer forums, of which there are already several
as well?) and subsequently start insulting the pre-existing members!

What I *did* do was try to learn and make useful contributions, which this
thread absolutely does NOT qualify for!

This is my last post - unless anybody else wishes to insult me as well,
which I'd prefer off-list, please, as not to further pollute the list - on
this topic.

*gets off of soap box*

PS: To the rest of the list: Sorry for continuing this thread!

-- 
Francesco Peeters

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Re: [Asterisk-Users] Asterisk Discussion Form

[Francesco Peeters]

On Sun, April 3, 2005 5:42, [EMAIL PROTECTED] said:
> List:
>
> With recent discussions in regards to a forum, I have set-up a
> multi-faceted Asterisk and Open Source Discussion Board. The link is
> www.voipnewbie.com/forum It is open and ready for use.
>
> Enjoy!
>
> VoIPNewbie
>

Thank you for the effort... I will be sure to take a look there!

-- 
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Re: [Asterisk-Users] IAX2 + Sonicwall

[Francesco Peeters]

On Fri, March 10, 2006 14:49, Dr. Michael J. Chudobiak said:
>>> I've found that inbound IAX2 calls don't work reliably (i.e., I get a
>>> busy tone) unless I enable "Use Consistent NAT" in the Sonicwall. This
>>> feature is poorly documented by Sonicwall, so I thought I'd pass it
>>> along.
>>
>> I've used the iaxcomm softphone and a snom 200 behind serveral different
>> sonicwalls over the past year or so without any problem. The sonicwall
>> should not be a problem for iax calls at all.
>
> I think the problem occurs when an Asterisk server inside the firewall
> tries to register multiple DIDs with one IAX2 provider outside the
> firewall. The Asterisk server worked fine when it was connected outside
> the firewall.
>
> The Sonicwall TZ170s do handle SIP transformations very nicely, though,
> if your Asterisk server is outside the firewall.
>

If the persistent NAT is not enabled, the SonicWALL is allowed to change
the NATted (source) portaddress. I can imagine that changing the port on
an IAX2 connection can cause problems on inbound sessions. When Persistent
NAT is on, the SonicWALL is told to use the same portnumber as the
original request from the LAN based machine.

This can cause problems if you have multiple machines connecting to the
same remote resource as there is a 1 in (approc) 64k chance per connected
machine that it uses the same port number as another machine that already
has a session up.

The chance of it causing a screw up are small enough to be able to have it
turned on, as I have. I am not sure what the default is for new machines,
but I know older machines that were upgraded to newer firmware will be off
by default...

HTH!

(PS: It's not the only thing poorly documented by SNWL... They
unfortunately have a history of poor documentation! It *does* keep their
support agents working though, so I guess that's something!  )

-- 
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Re: [Asterisk-Users] IAX2 + Sonicwall

[Francesco Peeters]

On Fri, March 10, 2006 18:56, Rich Adamson said:
>> >>Hi all,
>> >>
>> >>I currently have an Asterisk test server behind a TZ170 Sonicwall
>> >>firewall / NAT box, with several DIDs.
>> >>
>> >>I've found that inbound IAX2 calls don't work reliably (i.e., I get a
>> >>busy tone) unless I enable "Use Consistent NAT" in the Sonicwall. This
>> >>feature is poorly documented by Sonicwall, so I thought I'd pass it
>> along.
>> >>
>> >>Has anyone else run into this, or figured out the rationale for it?
>> >>
>> >>
>> >
>> >I've used the iaxcomm softphone and a snom 200 behind serveral
>> different
>> >sonicwalls over the past year or so without any problem. The sonicwall
>> >should not be a problem for iax calls at all.
>> >
>> >
>> OK apart of my beleive that sonicwall is a piece of crap (personal), try
>> to do a port forwarding for the IAX port (4569)
>
> I don't have a sonicwall here to test with. The ones that I was referring
> to are production units in Schools and Banks in the Midwest, and I was
> using iaxcomm (inside) to access asterisk (outside on a registered IP).
> In general terms, the customers that have them are very satisfied with
> them, but most don't have to deal with their tech support. The state of
> South Dakota has standardized on the sonicwall stuff and really have not
> had any significant issues (they have had some minor ones though).
>
> We've also configured some of those boxes to port forward (to the inside)
> for various functions, but none with udp 4569. Never had any issues other
> then making sure another rule isn't higher in the chain doesn't block
> first.
>
> The majority of those sonicwall boxes have been the larger units intended
> for small business use. No experience here with the small entry-level
> boxes.
>
> Our company does not resell any hardware or software, and we have no
> association with sonicwall whatsoever.
>
>

SonicWALLs are targeted at SMB and above. They don't really have
'entry-level'  boxes. I currently have a TZ170 and SonicPoint combo, and
don't have any serious issues with it. There are some minor issues, but
there's no product I know of that doesn't have some issues...

On the whole I am pretty satisfied with them and would recommend them to
others. (BTW: I do not sell SonicWALLs nor do I work for the company)

Other people like Zyxel's, which I think are crap... To each their own!  ;-)

-- 
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[Asterisk-Users] Com-On-Air (PCI/PCMCIA) chan drivers?

[Francesco Peeters]

Does anybody know whether somebody ever implemented (linux) drivers or a
chan component for the Com-On-Air cards, or is the only way to make it
work the use of a Windows box as Dect to SIP gateway for *? (Or does it
work at all?)

I can still get the cards, which would be great to retire the old DECT
based PBX, but it would be of little use if it's not going to work, now is
it? ;-)

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Re: [Asterisk-Users] Install just to play with experiment

[Francesco Peeters]

On Fri, August 12, 2005 0:25, Doug Lytle said:
> Michael Jones wrote:
>
>>
>>Is there a quick configuration that can be put into place to simply
>>experiment with the system (like create a couple of extensions wit Xlite
>> and
>>make a "internal phone system" in my office that doesn't really go out
>>anywhere - or maybe connects to a voip provider later)?
>>
>>
>>
> If you compile from source you can do a make samples .
>
> Doug
>

And otherwise, you can try Asterisk @ Home, but be warned that it installs
a complete new system, so do not try it on an existing machine you do not
wish to be wiped clean first!

-- 
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RE: [Asterisk-Users] voipbuster

[Francesco Peeters]

On Thu, December 1, 2005 17:09, Don Fanning said:
> I ended up buying a second 1 euro account because of this.  But it does
> work fine.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
> Vargas
> Sent: Thursday, December 01, 2005 7:21 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] voipbuster
>
> I was testing voipbuster. With a new account, with no credit, I can make
> calls perfectly but of 1 minute.
>
> But I tried the username and passwrord of an account with credit, and
> the registration is refused. With the voipbuster propietary software it
> works ok (I sniffed the packets and I think it is not using standard iax
> or sip ports). Are the acconts with credit blocked for avoiding it's use
> with ohter software than voipbuster's?
>
> I tryed to send a mail to voipbuster's support but I never received an
> answer (then do not support other thing than their software).

Mine works just fine. It's a pain though if you have to get a new account,
as minimum amount is now EUR 5... OTOH, for free calls, it might be worth
it...

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[Asterisk-Users] Transfer problem...

[Francesco Peeters]

I am having issues with transferring calls.

I can transfer outgoing calls, but not incoming calls.

* 1.2 / BRIstuff 0.3 PRE1 / 1 HFC cards
 Connecting calls between the 2 cards (1 NT mode, 1 TE mode)

The caller is always able to xfer the call, the callee only sometimes
(have not yet been able to identify the exact circumstances)

Any suggestions welcome!

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Re: [Asterisk-Users] voipbuster

[Francesco Peeters]

On Fri, December 2, 2005 9:26, Alejandro Vargas said:
> 2005/12/1, Tony Hoyle <[EMAIL PROTECTED]>:
>> The following works in iax.conf for me:
>>
>> [voipbuster]
>> host=iax.voipbuster.com
>> type=peer
>> username=
>> secret=
>> qualify=yes
>> context=inbound
>
> Context=inbound? I'm using from-pstn.
>
> The problem is this: I configure asterisk (through amp) with the
> username and password of one just created account (without credit) and
> I'm able to make calls of one minute. All OK. Then I change the
> username and password for the one of one account that has credit and
> is working ok with the propietary software. The only change I make is
> username and password. Then, the registration is refused.
>
> I double checked and the username and password works fine with the
> propietary software. Also the asterisk configuration works with
> another username/password without credit. I sent a mail to voipbuster
> support with the user that has credit but they never answered.
>

My (Working!) VB AMP settings are:

Trunk Name: voipbuster

Peer Details:
host=iax.voipbuster.com
secret=
type=peer
username=VBUSERname

USER Context: VBUSERname

USER Details:
context=from-pstn
secret=picard
type=user

REGISTRATION String: VBUSERname:[EMAIL PROTECTED]

HTH

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Re: [Asterisk-Users] voipbuster

[Francesco Peeters]

On Fri, December 2, 2005 9:29, Alejandro Vargas said:
> 2005/12/1, Francesco Peeters <[EMAIL PROTECTED]>:
>> Mine works just fine. It's a pain though if you have to get a new
>> account,
>> as minimum amount is now EUR 5... OTOH, for free calls, it might be
>> worth
>> it...
>
> The account that I'm testing is one of 5 EUR. I'm trying to test it
> with asterisk because if it works, I will place credit on my account.
> But if I can't make it work with an account with credit, I must
> suspect they are blocking the accounts with credit for avoiding people
> to use it with other software than their propietary client.
>
>

Mine is a EUR 5 account, and is working fine...

I set up below outgoing routes for VB free calls:
0030.
00311.
00312.
00313.
00314.
00315.
00317.
0034.
00352.
00353.
00358.
0041.
0043.
0045.
0046.
0047.
0049.


(The 0031x are set up in this manner to avoid Cellphone (0031-6.) and
Premium (0031-8. & 0031-9.) numbers.)

HTH

-- 
Francesco Peeters

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Re: [Asterisk-Users] voipbuster

[Francesco Peeters]

On Fri, December 2, 2005 15:08, Patrick said:
> On Fri, 2005-12-02 at 10:24 +0100, Francesco Peeters wrote:
> [snip]
>> (The 0031x are set up in this manner to avoid Cellphone (0031-6.) and
>> Premium (0031-8. & 0031-9.) numbers.)
>
> Afaik 0031-8. are freephone numbers, not premium.
>
> Regards,
> Patrick

You're right from a user's point of view, of course, but because I have
had issues with dialing 0800 numbers before, I am keeping them out of the
VoipBuster range...  :-)

>From a provider's POV even 0800 numbers are premium numbers, because the
owner of that number pays extra to use them.

-- 
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Re: [Asterisk-Users] [EMAIL PROTECTED] isdn

[Francesco Peeters]

On Fri, December 2, 2005 14:00, Alejandro Vargas said:
> 2005/11/30, Francesco Peeters <[EMAIL PROTECTED]>:
>> Yep, tried APIC, NOAPIC, ACPI=OFF, etc. (capitals only for clarity!) but
>> to no avail! As soon as both share the same IRQ, the zaphfc driver stops
>> passing data to asterisk...
>
> It is supposed to when you are using APIC, you should obtain many
> interrupts and the devices will obtain new interrupts with values
> grater than 16 which is the maxymun in a normal PC without APIC. Chech
> this lspc, you will see:
>
> smbus in irq17
> sound card in irq18
> usb controllers in irq 20, 21 and 23
> 4 ethernet cards, in irqs 16, 18 and 18 (it seems one of them are
> sharing irq with other)
>
> Your problem could be the isdn cards are not apic compatible.
>
>

It's most likely the MoBo that is at fault... It *is* one of the earlier
MoBo's with APIC capabilities, and IIRC the damn' thing has shared IRQ
lines for the two bottom PCI slots!  :-o

Many people use older (surplus) hardware for their * server, and it is in
such cases that one can run in to things like above madness.

It's just a general warning that may or may not apply to your situation.
At least you cannot say I didn't warn you !  ;-p

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[Asterisk-Users] ZapHFC cards not maintaining sync?!

[Francesco Peeters]

My HFC-PCI card is losing connection with the ISDN PSTN for minutes at a
time on a very regular basis...

This is of course unacceptable for any PBX...

The Dutch PSTN disconnects the D-channel once a minute.
About one minute every 5/10 minutes the D channel goes down, and doesn't
get beyond F6 (DEACTIVATED), instead of F7 (ACTIVATED)

Does anybody have any experience in this?

Note that I have 2 cards in the machine, the first one in NT mode, the
second in TE mode. The NT mode one never has issues as far as I can
tell...

I am using * 1.2 BRIstuffed 0.3.0 Pre1

Any suggestions would be appreciated...

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Re: [Asterisk-Users] ZapHFC cards not maintaining sync?!

[Francesco Peeters]

On Fri, December 2, 2005 21:45, Kristof Hardy said:
> Francesco Peeters wrote:
>> Does anybody have any experience in this?
>> I am using * 1.2 BRIstuffed 0.3.0 Pre1
>
> No experience on that, but there's an updated bristuff (0.3.0pre1b),
> maybe try that one?
>
> This is 1 issue that's fixed:
> - chan_zap/libpri fixes (stuck B channels)
>

Just installed 0.3.0pre1c, but no change!  :-/

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Re: [Asterisk-Users] ZapHFC cards not maintaining sync?!

[Francesco Peeters]

On Fri, December 2, 2005 22:50, Francesco Peeters said:
> On Fri, December 2, 2005 21:45, Kristof Hardy said:
>> Francesco Peeters wrote:
>>> Does anybody have any experience in this?
>>> I am using * 1.2 BRIstuffed 0.3.0 Pre1
>>
>> No experience on that, but there's an updated bristuff (0.3.0pre1b),
>> maybe try that one?
>>
>> This is 1 issue that's fixed:
>> - chan_zap/libpri fixes (stuck B channels)
>>
>
> Just installed 0.3.0pre1c, but no change!  :-/
>

I have now got this little ditty running to keep an eye on it:
while true; do grep "(F" /proc/zaptel/2; sleep .1; done

I do see the once a minute down-time come by as a combination of 1 F4, 2
F6's and then F7's.
When it goes down for an extended time, it shows 1 F4 and a lot of F6's
before finally returning F7's again...  :-(

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Re: [Asterisk-Users] [EMAIL PROTECTED] isdn

[Francesco Peeters]

On Sat, December 3, 2005 9:28, Karsten Wemheuer said:
> Hi,
>
> On Tue, November 29, 2005 13:50 Francesco Peeters wrote:
>> BTW: BRIstuff is not included by default as it breaks PRI support.
>> Asterisk is already set up to use zap, so that is easy...
>
> As far as I know, BRIstuff is not included for licencing reasons... Is
> it true, that PRI support and BRIstuff are now incompatible? (In version
> 1.0.9 I had no problems to use two HFC cards and one TE110 in one
> system).
>

I have seen it stated on several sites that it is incompatible, but I have
also heard from different people that it does work...

I do not have a PRI, so I cannot comment on it. It is possible that
they've included that because actually getting it to work may be a hit and
miss thing (correct loading orders, etc.), but again, I cannot comment on
that as I do not have a PRI!  ;-)

All I have are problems with the BRI...  :-o
(But that may (partially) be the Dutch KPN's fault as well, insisting on
bringing down the D channel every minute... I never understood the
reasoning behind it, as bringing down a running connection that often only
increases chances of incompatability, like the 3Com NetBuilder had, and
the chances of instability, like I have now...)

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Re: [Asterisk-Users] ZapHFC cards not maintaining sync?!

[Francesco Peeters]

On Mon, December 5, 2005 7:22, Remco Barende said:
>
>> Already HAVE Florz patch installed!  :-(
>> What version of * and BRIstuff are you using?
>
> Strange, sounds like the florz patch has not been effectively applied or
> it's broken. I'm using an old version of bristuff :
> Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n built by [EMAIL PROTECTED]
> on a x86_64 running Linux

The patch applied fine, No errors, nor .rej files
>
> I had an issue compiling it for x86_64 though, but that's a
> different question.
>
>> I assumed as much when I saw your last name... :-)
>> Whereabouts in NL? I'm in Zoetermeer (ZH)...
>
> Amsterdam, but the ISDN setup I installed near Leiden (ZH) :)
>

Close enough!  ;-)

>>>> 1) Every 10 seconds () the D channel gets torn down, which
>>> That's too slow, it should happen about every 1-2 seconds or so. The d
>>> channel going down and up again is normal behaviour.
>>
>> I know it is. Used to work for a Networking Competence Centre, and we
>> had
>> the same kind of issues with 3Com Netbuilders. The first call attempt
>> after the D Channel was torn down always failed... The only solution was
>> to get KPN to turn on the D Channel permanently...
>
> Strange, I never had that problem before. When the * box gets up I can
> immediately make calls. Also the standard KPN A/B equipment doesn't have
> this problem, sounds like it's more 3Com related.
>

It was... 3Com didn't recover from the tear-down. Only solution was
preventing the teardown  ;-)

> One problem I have found with bristuff (and no solution yet), if you
> disconnect the ISDN line from the * box (or the ISDN line is out of order
> for a short while), bristuff will not re-establish the connection. It is
> then required to unload all the modules and re-load them or even worse,
> reboot the box. I guess that is a specific bristuff problem. All calls to
> the ISDN line fail and it's not possible to make any calls. Even after
> several hours bristuff doesn't setup the line connection.
>

Not seen that yet... I can unplug the line, plug it in again and it'll
bring it back up...

>>>
>>>> 2) Results in the CRC error, which means that
>>>> 3) Every 3 minutes, the D channel goes down for EXACTLY 1 minute.
>>>
>>>> I could try to get the KPN to give me a permanent D channel, but are
>>>> there
>>>> any tricks to try that would/could make asterisk somehow keep up the D
>>>> channel?...
>>
>> I noticed that the 'deactivated' issue doesn't happen for a while after
>> a
>> call has been placed.
>>
>> I am now testing placing a call every minute, with a 100 ms timeout
>> using
>> the manager api. This means it never actually gets a chance to get
>> through, or be picked up, but it does cause activity on the D channel.
>>
>> This has been running for half an hour now, and I haven't seen the
>> channel
>> go down for extended periods since.
>>
>> I'm not sure whether the KPN will like it, but it's an interesting test
>> to
>> run!  
>>>
>>> Good luck with our Royal Dutch KPN, but I would try florz first :)
>>>
>>
>> Tell me about it! Like I said above, we had *extensive* experience with
>> them over the D Channel issue!
>

It is in NL, but that is because the KPN have decided to do it that way.
*Normal* PSTN's keep D channel alive

> Weird, I checked with KPJ before and he mentioned it is normal behaviour
> for ISDN. My console is filled with messages like this :
>== Primary D-Channel on span 1 up
>== Primary D-Channel on span 1 down
>== Primary D-Channel on span 1 up
>== Primary D-Channel on span 1 down
>== Primary D-Channel on span 1 up
>== Primary D-Channel on span 1 down
>
> and it doesn't cause me any issues. It would be nice to 'hide' these
> messages when not in very verbose mode to avoid cluttering up the console.
> The messages indeed do appear about every 10 seconds or so.

Yep...

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Re: [Asterisk-Users] Re: VoipBuster / Finarea

[Francesco Peeters]

On Tue, December 6, 2005 19:46, Wolf N. Paul said:
> Tony Hoyle wrote:
>
>> btw. does anyone have a definitive list of all the finarea VOIP
>> companies?  I can think of:
>>
>> call1899
>> call18866
>> voipbuster
>> sipdiscount
>> voipcheap (note: this one uses a proprietary protocol, similar to IAX
>> but over different ports and not compatibile).
>>
>> ***
> I try to keep a list of Finarea services (I don't think they are
> separate companies, just web presences)
> here: http://www.voip-info.org/wiki/view/Finarea+SA
>


Now all we need is a VoIP provider to give us free calls to cellphones in
Europe...  ;-)

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[Asterisk-Users] Test to see if I'm still on list...

[Francesco Peeters]

As I haven't received any posts since yesterday...


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Re: [Asterisk-Users] oh323 and IAX2

[Francesco Peeters]

On Mon, August 22, 2005 20:21, CM Rahman Jr. said:
> Anybody here using iax2 for one call leg and other call leg for oh323? I
> am
> getting broken sounds from Iax2 call get.
>
> Can somebody here help?
>
> Thanks
>
OT: What does this have to do with the small office / analog thread?

Anyway:
H.323 isn't quite as flexible as SIP or IAX/2. Anything inbetween can -
unfortunately - easily break H.323 (especially NAT routers/firewalls)

Is it possible to test with different protocols like SIP? That way you can
test whether it is the server or something along the way...

Good luck!

-- 
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Re: [Asterisk-Users] How to use * and # as part of number in dial command

[Francesco Peeters]

On Sat, August 27, 2005 23:41, Michel Koenen said:
> Hi all,
>
> I am struggling with the following and I can't get it work:
>
> In the Netherlands where I live it is possible to use special codes
> (aka vertical service codes) to set special 'behaviour' of phonecalls.
> So e.g. when I want to dial out with a normal phone and I dial
> *31* then it will turn off my numberindication
> (CID) at the called party.  They seem to call this the 'keypad
> protocol' but I  cannot find this term when searching through asterisk
> documents.
>
> My asterisk system is connected to an ISDN line with HFC card. I use
> zaphfc module for that.  In my extensions I tried several things to
> dial out and use the *31* but without success.
>

A few others are:
Call forwarding: *21*# / #21#
Delayed forwarding: *61*# / #21#
Busy forwarding: *63*# / #63#

(the ## numbers are to disable the service)

I am glad to see this topic come up before I ran in to it myself!  ;-) I'm
curious about the answers...


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Re: [Asterisk-Users] Calling PSTN lines from VOIP softphone

[Francesco Peeters]

On Sun, August 28, 2005 1:15, Aniket Bhat said:
> Folks,
>
> I am a newbie to the VOIP world and have a question (might as well
> sound silly to some). I would like to set up a PC-to-Phone call from
> my desktop to a regular PSTN number. Does the Asterisk PBX itself act
> as a VOIP-PSTN gateway or do I have to subscribe to a VOIP provider
> for this? Are there any free IP-PSTN gateways which I can subscribe
> to? Do I need any specific hardware to set up such calls?

With correct hardware to interface between the PSTN/ISDN and (*) it is
possible for (*) to do this for you, but you can also get a VOIP provider
to do it for you. The latter means you will not be able to call 911 (for
instance)

The best is usually a combination using least cost routing...
>
> If there is no additional hardware or subscription to a VOIP provider
> required, are there any resources that have information about
> configuring asterisk for such gateway functionality?

There is *no* way asterisk can get a VOIP call onto the PSTN without being
connected to it either directly (hardware) or through a third party (VOIP
provider)

>
> Note: I believe that for receiving calls, you may need some form of a
> FXO card or something, but if my requirement is to just make calls
> from a softphone on my computer to a PSTN no. and NOT vice versa, do I
> still need any additional hardware?
>

See above...

> Thanks in advance,
>
> Aniket.



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RE: [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmoth erboard

[Francesco Peeters]

On Tue, September 6, 2005 22:32, Remco Barende said:
> Just out of interest, how do you run the VIA boards? They only have one
> network connection and if you add a PRI card you cannot have both a LAN
> and NET connection?
>
> (Highly offtopic, sorry!)

I don't have any, but why would you need more than one LAN connection for
a typical POTS/PSTN/ISDN connected PBX?

And IIRC they do have USB connectors, which means you could always use a
USB LAN adapter...

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Re: [Asterisk-Users] How to connect many analog lines to Asterisk?

[Francesco Peeters]

On Wed, September 7, 2005 18:11, Josip Gracin said:
> Darren Wright wrote:
>> Wow, first of all, if you have a hundred analog lines, you are doing
>> yourself a disservice.a 4 T1's would be much much cheaper, and much
>> easier to manage.
>
> Let me clear this up a little bit.  There are hundreds of telephone
> devices inside the building, all connected to a PBX, and there is an
> E1/T1 connection to the PSTN (being statistically multiplexed,
> obviously).  What I'd like to do is to replace the PBX with Asterisk.
>
> I don't see how I can make the situation better by using 4 T1's?
>

You said you had 100 analog lines... What you meant is you have 100 analog
phones... Big difference... (OTOH, only a single letter: FXO -> FXS)  ;-)

But seriously, there really is a big difference whether you are trying to
connect 100 analog lines (i.o.w. 100 incoming POTS lines from the PSTN) or
100 analog phones... If you had 100 incoming POTS lines, 4 PRI spans would
be way cheaper and way easier, hence the advice!

>> Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and fill
>> them with FXO cards, and then buy 1 TE406 Quad T1 card for your asterisk
>> box.
>
> Thanks, I think that's what I need.
>
If you want 100 analog phones, make sure you get FXS cards in stead of FXO
cards... FXO cards are for incoming lines, FXS for phones...

(FXO stands for Foreign Exchange Office, ie PABX or PSTN, FXS for Foreign
Exchange Subscriber, ie telephones)

HTH

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Re: [Asterisk-Users] IAXy - no dailtone

[Francesco Peeters]

On Thu, September 8, 2005 0:11, Jimmy said:
> I have a brand new IAXy I'm playing with.  I do not get a dialtone on
> the phone, or any response at ll on the phone.  No sound, no dialing, no
> ringing.  The phone and wire are tested and known to be good. I think I
> have it setup correctly.  When I give the iaxprov command I get this:
>
> #iaxyprov 192.168.1.90 iaxy.conf
> 02:
> c0 a8 01 5a
That's 192.168.1.90, looks good!
> 05:
> 11 d9
> 03:
> ff ff ff 00
That's 255.255.255.0, looks good too!
> 04:
> c0 a8 01 7d
That's 192.168.1.125, still looks good!
> 0d:
> 00 00 00 04
> 0f:
> c0 a8 01 c7
That's 192.168.1.199, still looks even better!
> 10:
> 11 d9
> 06:
> 69 61 78 79
That's 'iaxy', good!
> 07:
> 70 61 73 73 77 6f 72 64
That's 'password', still good!
> 0c:
> 00 00 00 01
> Provisioning is 60 bytes
> Total packet is 74 bytes
> Got response back from '192.168.1.90'
>

Looks like the iaxy is configured conform the iaxy.conf info...

>
> Here is my iaxy.conf:
> ;
> ; IAXY Provisioning description
> ;
> ;dhcp
> ip: 192.168.1.90
> netmask: 255.255.255.0
> gateway: 192.168.1.125
> codec: ulaw
> ;codec: adpcm
> server: 192.168.1.199
> ;altserver: 192.168.0.2
> user: iaxy
> pass: password
> register
> ;heartbeat
> ;debug
> ;
> ; Feature tuning (default is all enabled)
> ;
> ;disablecid
> ;disablecw
> ;disablecidcw
> ;disable3way
>
>
> The IP addresses here are all correct.
>
> Here's the relevant portion of iax.conf:
>
> [iaxy]
> type=friend
> user=iaxy
> host=dynamic
> secret=password
> context=incoming
> disallow=all
> allow=ulaw
> callerid="My IAXy" <(555) 555-1234>
> trunk=no
>
> The BLUE light on the IAXy is lit.  The ORANGE light blinks about once
> every 7 seconds.  I can dial this extension from another phone, and the
> ORANGE light blinks rapidly while the phone should be ringing, but it
> doesn't ring.  And, as I stated earlier, the phone has no response at
> all.  No dialtone, no dialing, no ringing.
>

I'd almost suspect it is defective or lacking sufficient power... You
*have* used the supplied PSU?

> Have I missed something obvious?  Is there some other test I can try?
>

Not that I know of... You could try dialling from the connected phone even
though there is no dial tone... It could also be that the telephone you
are using is defective or wired in a non-standard way...

When the led blinks quickly, will you get a connection when you pick it
up? What do the leds do when you pick up the phone?

> Thanks in advance for any input.
>
> Jimmy Madden

Good luck!

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Re: [Asterisk-Users] Meetme Question

[Francesco Peeters]

On Tue, September 13, 2005 11:53, Accursio Avona said:
> Hi all,
> I'd like to use the w option of the meetme application.
>  From tiki i read:
>
>'w' -- wait until the marked user enters the conference
>
> * All other connected users will hear MusicOnHold until the marked
>   user enters.
>
> The question  is, how can i indicate the "marked user"?
>

Basically you create 2 different PINs for the meeting. One for normal
users, one for the 'marked' user. It can then differentiate by comparing
PINs.

HTH

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[Asterisk-Users] wiki down?

[Francesco Peeters]

I'm unable to connect to voip-info.org... Anybody else have the same
issues, ro is it just me?

-- 
Francesco Peeters

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RE: [Asterisk-Users] wiki down?

[Francesco Peeters]

On Fri, September 16, 2005 19:53, Wiley Siler said:
> I got right in just fine...
>
> W
>
>
Me too now.  :-/

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Re: [Asterisk-Users] Re: Welcome to the "Asterisk-Users" mailing list

[Francesco Peeters]

On Tue, September 20, 2005 2:10, chuck gelm said:

> What is the URL of the "list archive"?
As listed at the bottom of each email on the list:
http://lists.digium.com/mailman/listinfo/asterisk-users

You can also use google: "SEARCH KEYWORDS site:lists.digium.com"

> The 'wiki' seems broken.
It's working fine from here

>
>   I want to connect to a single home POTS from an internet connection.
> i.e. I want to be able to make telephone calls on my home POTS
> from anywhere that I can obtain an internet connection with
> a device with a microphone and a speaker. e.g. My laptop and a
> headphone with boom microphone.  Does 'Asterisk' do this?
> Should I look elsewhere for this task?

It's a bit overkill, but it can certainly do it!

>
>   If this is answered in the list archive, please respond with the
> URL of the list archive.

As this is the premise of (*), being a software PABX, the entire list is
*full* of posts on this topic...

You might want to search for Asterisk @ Home (aka AAH, [EMAIL PROTECTED]) 
though...

>
> Best regards,
> Chuck
>

PS: Please do not reply to an existing email to start a new thread,
neither with or without changing subjects, etc. It really screws up
threaded email clients. I had to page back all the way to my first email
to find your question, as it was sorted under *my* welcome email... (I
only did it because it showed 1 unread email, and I do not like unread
emails hanging around. Usually I have less time and just click 'mark all
read' when this happens, and I know there are many more. Also your mail
would have never come up if I had the thread marked for deletion... So
it's not only courteous, it's also good sense! )

Just click new and send to [EMAIL PROTECTED] Cleaner,
easier and it doesn't screw up threading in mailclients... (You can add
the list to your address book too!)

Good luck!

-- 
Francesco Peeters

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[Asterisk-Users] What hardware would you recommend?

[Francesco Peeters]

I have 3 locations I want to connect using (*) servers.

1 of those has a single BRI with a Siemens DECT PABX.
1 of those has two BRI's with 2 Siemens DECT PABX's, each serving a
different area.
1 of those has two BRI's and a 2 port Nova Compact PABX with DECT

First step would be to set up the (*) servers and have them
interconnected. When all of that works we'd go on to connect them to the
ISDN and connect the existing PABX's to the servers so we can - for now -
maintain the existing environment but use (*) to route traffic on a least
cost basis, as well as allow SIP/IAX connections from out of office
locations.

The machines themselves will not pose much of a problem, but what ISDN
hardware would you recommend for this? (1 site with 1 TE and 1 NT mode
port, 2 sites with 2 TE and 2 NT mode ports)

TIA!

-- 
Francesco Peeters

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Re: [Asterisk-Users] Asterisk PBX

[Francesco Peeters]

On Wed, September 21, 2005 15:14, Tom Rymes said:
> On Wed, 21 Sep 2005, kapil dhawan wrote:

> If you choose to do it yourself, [EMAIL PROTECTED], voip-info.org, and this 
> list
> will be your best tools.
>
> Tom

I'd like to add Google to that shortlist:

 + site:voip-info.org
or
 + site:lists.difium.com

will help you quickly search the wiki and list archives...

Good luck!


-- 
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Re: [Asterisk-Users] Asterisk PBX

[Francesco Peeters]

On Wed, September 21, 2005 15:19, Francesco Peeters said:
> On Wed, September 21, 2005 15:14, Tom Rymes said:
>> On Wed, 21 Sep 2005, kapil dhawan wrote:
> 
>> If you choose to do it yourself, [EMAIL PROTECTED], voip-info.org, and this 
>> list
>> will be your best tools.
>>
>> Tom
>
> I'd like to add Google to that shortlist:
>
>  + site:voip-info.org
> or
>  + site:lists.difium.com
>
> will help you quickly search the wiki and list archives...
>
> Good luck!
>
Oops! Typo!

 + site:lists.digium.com

is the correct syntax...

Sorry!

-- 
Francesco Peeters

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Re: [Asterisk-Users] Asterisk PBX

[Francesco Peeters]

On Wed, September 21, 2005 15:30, Nathan Pralle said:
>>  + site:lists.difium.com
>
> The above is good when searching for information on Joe Diffie --
> Otherwise, you'll want:
>
>  + site:lists.digium.com
>
> :)
>
> Nathan
>
>


I already corrected myself... I canna help them list servers take so long!
 ;-)

---FP
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[Asterisk-Users] What ISDN hardware would you recommend?

[Francesco Peeters]

Trying again...

*Summary:*
I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE
mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode;
What card(s) should I put in to these servers?

*The long story:*
I have 3 locations I want to connect using (*) servers.

1 of those has a single BRI with a Siemens DECT PABX.
1 of those has two BRI's with 2 Siemens DECT PABX's, each serving a
different area.
1 of those has two BRI's and a 2 port Nova Compact PABX with DECT

First step would be to set up the (*) servers and have them
interconnected. When all of that works we'd go on to connect them to the
ISDN and connect the existing PABX's to the servers so we can - for now -
maintain the existing environment but use (*) to route traffic on a least
cost basis, as well as allow SIP/IAX connections from out of office
locations.

The machines themselves will not pose much of a problem, but what ISDN
hardware would you recommend for this? (1 site with 1 TE and 1 NT mode
port, 2 sites with 2 TE and 2 NT mode ports)

TIA!

-- 
Francesco Peeters

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


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Re: [Asterisk-Users] One-way audio with VPN

[Francesco Peeters]

On Tue, September 27, 2005 20:22, Alex Lake said:
> I've got a one-way audio problem, but I've looked through a few
> documents on the subject and I'm not sure that it's the same issue.
>
> User A calls a local Asterisk user B via a public SIP gateway
> (voiptalk.org) using (sip:[EMAIL PROTECTED])
>
> B is connected to the Asterisk server via VPN
>
> B is registered (and has successful bi-directional conversations with
> other users on the VPN)
>
> Asterisk correctly forwards the call via SIP and B's phone rings and is
> answered, but B can't hear A
>
> So there appears to be an audio-path blockage from A via Asterisk to B.
>
> Now if A leaves a voicemail message on the asterisk box, that's fine
> (the sound file contains a recording of A's voice!)
>
> Therefore, it looks like the problem is to do with the forwarding of RTP
> packets by Asterisk from A (Internet origin) to B (VPN).
>
> Any ideas?
>

If you're not doing NAT on the SOURCE IP of the A before transferring
across the VPN, it is very likely that B is replying DIRECTLY to A rather
than through the VPN. This will cause B to answer with a different Source
IP than A has initiated the call to, causing the packets to be dropped.

You can easily check this by doing a packet trace on the LAN segment of B...

Good luck!

-- 
Francesco Peeters

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Re: [Asterisk-Users] analogue phone with asterisk

[Francesco Peeters]

On Tue, September 27, 2005 19:10, Rajesh Bhairampally said:
> I am a newbee to asterisk. I recently installed [EMAIL PROTECTED] Everything
> went well and my set up is running fine with soft phones, such as kphone
> and XtenLite. Now, i want to be able to connect my analogue phones to my
> asterisk pbx box and use it as if i make a regular Phone call (I do have
> my PSTN gateway account with broadvoice.com and already configured to
> route through it). I do NOT have a PSTN phone connection. I want to use my
> analogue phones as the end points for my asterisk box to make and receive
> calls. All i want is to use my analogue phones instead of soft phones.
>
> Can some one help me what hardware interface i need for that and how
> should i go about it? if there is any HOW-TO for that it will be of great
> help.
>
It depends on how many phones you have...

You can start with a few IAXy's at the low end all the way to channel
banks at the high end...

The wiki is your friend:
http://www.voip-info.org/tiki-index.php?page=Analog+Telephone+Adapters

HTH

-- 
Francesco Peeters

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Re: [Asterisk-Users] Asterisk on windows

[Francesco Peeters]

On Sun, October 2, 2005 12:07, Patrick said:
> On Sun, 2005-10-02 at 10:21 +0100, Wayne wrote:
>> Hiyall,
>> been following this for a while, just thought I would add a bit to the
>> debate, but doesn't the Cisco system (Call Manager?) run on an Windows
>> 2000 based server - if it was that bad why would Cisco choose to run it?
>
> Politics and cluelessness. There are rumours that Cisco's next CCM will
> run Linux. Cisco also used Win2000 on their BBSM product. An amazing
> piece of crap according to those who had to install it and maintain it.
> You had to reboot the thing over and over. Sounds familiar?
>
>> Also 3Com use NT/2000 to run the H323 gateway. Admittedly the call
>> processor runs on VXWorks but to cross the boundary of proprietary 3com
>> and rest of world - they jump onto windows.
>
> VWWorks is as stable as it gets compared to M$. At least they had the
> brains to put the important part on a Unix like OS. About the M$ part,
> well, it's silly decisions like that that contribute to 3Com's fading
> away.
>
> Regards,
> Patrick
>

*shrugs*
SonicWALL (firewall company) have always had their Global Management
System on Sun/Oracle and M$/SQL2000...

The first combination was more stable from the onset, but the second has
been sold many more times, at some point even pushing away
development-time for the Sun/Oracle combination in favor of M$/SQL2000.
They'd probably have dropped it if it hadn't been in use at a few very
large sites (obviously run by people that *did* have a clue what they were
doing!)

>From the onset there have also been many crying for a Linux version, but
again, M$/SQL2000 development took so much time, I still haven't seen a
glimpse of it!

Sometimes a company just doesn't have a choice, and in a market dominated
by M$ manipulation, ehrr... monopolisation, you're quickly condemned to M$
if you need to sell to a large market!

That doesn't make it a better plastform than Linux, but them ITC managers
just don't know there's something out there that is more stable, more
reliable, less costly, etc.

-- 
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Re: [Asterisk-Users] ISDN card

[Francesco Peeters]

On Tue, October 4, 2005 21:19, Emanuele Pucciarelli said:
> Andrea Bencini ha scritto:
>> I would like to know if somebody has already used a C2-ISDN or a 4BRiJUN
>> card to connect asterisk to PSTN network. My problem is that i can't
>> configure my asterisk.
>> Please, help me with some solution!
>
> Non so se riceverai qualche risposta in lista, ma sicuramente lo hanno
> fatto in molti, sia con l'una (se è una AVM) che con l'altra!
>
> Qualche dritta per iniziare: con la AVM ti servirà chan-capi.  Io ti
> consiglio il fork -cm mantenuto da Armin Schindler, disponibile su
> Sourceforge.  Con la scheda quadBRI invece devi usare le patch bristuff,
> con i cui esempi di configurazione dovresti poter partire immediatamente.
>
> Saluti,
>
> --
> Emanuele

And now in English?   ;-)

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Re: [Asterisk-Users] ISDN card

[Francesco Peeters]

On Tue, October 4, 2005 21:46, Emanuele Pucciarelli said:
>
>> And now in English?   ;-)
>
> I'm extremely sorry, I wanted to reply directly in order to cut down on
> FAQ traffic, but after I realized my mistake I did not hit "Cancel"
> quickly enough.  Shame on me :)
>
> It boiled down to "try chan-capi with AVM, try bristuff with the QuadBRI
> and, at least in the latter case, find excellent courtesy start-up
> example configuration files included" :)
>
> (And, yes, I know, with this I'm polluting the ML and its archives with
> not one, but two useless postings!)
>
> --
> Emanuele

ROFL!  Don't feel bad! At least it is ON-TOPIC!   ;-)

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Re: [Asterisk-Users] Re: www.openpbx.org

[Francesco Peeters]

On Sun, October 9, 2005 15:31, Matt Riddell said:
> *PLONK*
>
> --
> Cheers,
>
> Matt Riddell
>

Is that the sound of you dropping out of this list? It can't be a reply to
the previous poster's e-mail, as that was in fact a completely correct
statement...

But back to the topic: I can see the reasons why Digium does what they do,
but I can also appreciate the reasons behind openpbx.org!

It's a risk Digium took willingly and knowingly. They must have known from
the start that this was a risk, but it is a risk they took, I guess with
in the back of their minds the realisation that the risk would most likely
be worth it in return for the contributions they'd get to the
source-tree...

I for one will keep track of both to see which one develops best. For now
I am in the (*) camp, but if they lose ground in regard to openpbx.org, I
may switch.

That is the nice thing about Open Source: you *can* switch, and the best
project(s) will survive...

Just my € 0,02!

-- 
Francesco Peeters

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Re: [Asterisk-Users] PDA softphone....

[Francesco Peeters]

On Tue, October 25, 2005 20:27, [EMAIL PROTECTED] said:
> I have downloaded SJPhone - and well.. it does connect to my system,
> however popping audio is heard when i dial my music on hold extension...
>
> the quality is really really bad..
>
> i have a WLAN-SDIO utility, the signal strength is at 11mb/s. however. is
> that sufficient? The codecs for sjphone are fixed at 64000.. i could not
> change those values.
>
> has anyone had successful attempts with something better?
>
> Thanks...
>
Hello 'pbx'


Just wondering about two things...
1) What's your name? It's nicer to reply to someone by name...
2) What does this have to do with the thread in inter-asterisk
communications?


But seriously, we will need a bit more info, such as:
What version?
What platform?
What type of network? (PS: Signal strength is something in dBi or %,
11mbps is the speed of the connection, but is it the actual speed, or just
an indication of the type of network, ie 802.11b)
What codec are you using?
What config on the *?

Have you looked at <http://www.voip-info.org/wiki-Asterisk+phone+sjphone> ?

And please, next time start a new e-mail, and don't reply to an e-mail
from the list. It screws up e-mail threading in proper mail-clients. (Your
mail and my reply (and any others following) will be intermingled with
those from the 'How to configure the communication between two Asterisk
servers' thread)
Even deleting all old text and the subject won't change that, as the
threading is based on info in the headers or the e-mail you are replying
to... If you cannot remember the list-address, make it in to a 'contact'
for future use...   ;-)

Good luck!

-- 
Francesco Peeters

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RE: [Asterisk-Users] PDA softphone....

[Francesco Peeters]

On Tue, October 25, 2005 20:43, Dean Collins said:
> As a secondary point, I'm looking at buying a Imate Jas Jar running
> windows mobile 5.0 to replace my treo.
>
> Does anyone have any thoughts on Windows mobile 5.0 specific softphones.
>
> Cheers,
> Dean
>
>
LOL! If you wait a bit longer you can buy a WinCE Treo!  ;-)

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Re: [Asterisk-Users] open-source vs. tellme/skype/gNumber et. al.

[Francesco Peeters]

On Thu, October 27, 2005 12:10, Neil Skowronek said:
> Now that Skype and Ebay are one, I feel that they will
> be cherry-picking all the promising open-source
> voip/asterisk development and calling it their own.
>
> There is a company called gNumber that relies
> completely on Asterisk that has also teamed up with
> ebay for cell phone notification of ending auctions,
> they claim to have patents pending on 'transactions
> through voice channel'
>
> I'm new to open-source, perhaps this is the wrong
> forum to ask this question but where does the line
> exist between shared and ownership?
>
> The software that is asterisk has allowed for all this
> to develope, can people then take what freely
> distrubuted and own it?
>
> I know I'm opening a can of worms and need to read
> more on this, but I'd like to hear some learned
> opinions first and at least get a few links to help me
> with my research.
>
> THX
>
> Neil
>
>
>

You sure this isn't a homework question?  :-)

Very short answer: The GPL only allows use and redistribution of any of
the source released under the GPL if it remains under the GPL. So
*legally* they cannot take GPL code and call it their own.

Whether or not it happens is hard to say, esp. with closed source...

But that is only the tip of the iceberg. I hope it helps you along to
start the proper research...

Google is your friend, as is <http://www.gnu.org/copyleft/gpl.html>

-- 
Francesco Peeters

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[Asterisk-Users] I need GoIAX or VoipBuster [EMAIL PROTECTED] examples?

[Francesco Peeters]

Hi all,

I am playing with [EMAIL PROTECTED] / AMP for the first time, and would like
to test GoIAX or VoipBuster (or both) through the use of the AMP
interface.

I have tried several permutations of the possible configs, and sometimes
it logs in to GoIAX and VoipBuster, sometimes it doesn't, however it never
seems to be able to actually set up an IAX channel to start the call.

Using the VoipBuster client or IAXcomm for GoIAX I can dial out just fine.
The FOP shows the Extension calling a number, and shows one of the trunks
(currently always VoipBuster, even though the GoIAX trunk is primary in
the Out Route), but it never actually connects (I don't get a ringtone)
and after a while I get the all channels busy signal.

I can provide the current configs and full log data if anybody wants to
have a look, but I really would prefer to understand myself, and hope a
quick example will help me understand...

TIA!

-- 
Francesco Peeters

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If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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RE: [Asterisk-Users] I need GoIAX or VoipBuster [EMAIL PROTECTED] examples?

[Francesco Peeters]

On Sat, October 29, 2005 1:19, Kerry Garrison said:
> http://voipspeak.net has GOIax example for AMP.
> -Kerry
>
>

Sorry, couldn't find it... Do you have an exact url?

I only found IAX.cc/Sixtel, Teliax and Broadvoice samples...

-- 
Francesco Peeters

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If your program doesn't recognize my signature, please visit
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RE: [Asterisk-Users] I need GoIAX or VoipBuster [EMAIL PROTECTED] examples?

[Francesco Peeters]

On Sat, October 29, 2005 21:10, Francesco Peeters said:
> On Sat, October 29, 2005 1:19, Kerry Garrison said:
>> http://voipspeak.net has GOIax example for AMP.
>> -Kerry
>>
>>
>
> Sorry, couldn't find it... Do you have an exact url?
>
> I only found IAX.cc/Sixtel, Teliax and Broadvoice samples...
>
/IGNORE

I just found it on the Nerd Vittles site...  :-)

Gonna test it right now!

THX!

-- 
Francesco Peeters

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If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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RE: [Asterisk-Users] I need GoIAX or VoipBuster [EMAIL PROTECTED] examples?

[Francesco Peeters]

On Sat, October 29, 2005 21:58, Francesco Peeters said:
> On Sat, October 29, 2005 21:10, Francesco Peeters said:
> /IGNORE
>
> I just found it on the Nerd Vittles site...  :-)
>
> Gonna test it right now!
>
> THX!
>

I am so bloody embarrased! I decided to make a capture on my firewall and
found that data was coming in the LAN, but not going out the WAN port...

Investigating why that might be, I then found out that my IAX2 service in
the firewall was set for TCP instead of UDP!  *blushes*

I changed that and it instantly worked for both GoIAX and VoipBuster
(though they do give me some crap about no credits...)

Sorry for the disturbance on list!  ;-)

-- 
Francesco Peeters

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

[Francesco Peeters]

On Mon, October 31, 2005 8:40, KARIM MOUSLI said:
> hello evryone
>
> can somone help me get asterisk to work with outgoing calls to a voip
> operator
>
> i have tried many stings, but i cant triger the outgoing calls, calls on
> the same pbx are working fine
>
> what did i mis out ?
>
> in advance thanks  :)
>
> _

To start with, is your firewall setup for the correct data streams (IAX2
or SIP and RTP)?

Also what VOIPSP are you using?

Also, to be able to tell what you missed, we'd need to know what you DID
do, but starting with the VOIP SP we should be able to gove you more
hints...  ;-)

I succeeded this weekend (after correcting the IAX protocol type from TCP
to UDP, D'oh!) in setting up outbound with FWD, VoipBuster and GoIAX, all
IAX2 enabled providers (I greatly prefer IAX2 over SIP due to the greater
ease of configuring the firewall. Mine specifically supports SIP and H323
monitoring, and it still gives a huge headache!) So it can definately be
done!

Good luck!

Gopod luck!

-- 
Francesco Peeters

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

[Francesco Peeters]

I'm having some issues, and thought it wise to check with the list before
putting in any more time

Here we go:
1) Do Zaptel BRI (Cologne based cards) support DID routing (I believe they
do, but the behavior of my (*) server is making me doubt, and I want to be
sure before attempting any more permutations)
2) The (*) is parallel to my current Siemens Gigaset4135. Incoming calls
on all MSNs show up on the display for a split second. I assume that means
(*) steals them away before anybody would be able to answer. If all worked
fine, that would not be an issue, but until then, I would prefer parallel
rings, or at least for (*) to leave the main MSN alone, and only capture
the others. Is this possible?
3) Are MSN's the same as DIDs for (*)?

That's it for now, any help is appreciated...

-- 
Francesco Peeters

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

[Francesco Peeters]

Replying to my own post (I kept on trying while waiting for an answer),
just for the record (and the archives)

On Wed, November 2, 2005 19:33, Francesco Peeters said:
> I'm having some issues, and thought it wise to check with the list before
> putting in any more time
>
> Here we go:
> 1) Do Zaptel BRI (Cologne based cards) support DID routing (I believe they
> do, but the behavior of my (*) server is making me doubt, and I want to be
> sure before attempting any more permutations)

Yes

> 2) The (*) is parallel to my current Siemens Gigaset4135. Incoming calls
> on all MSNs show up on the display for a split second. I assume that means
> (*) steals them away before anybody would be able to answer. If all worked
> fine, that would not be an issue, but until then, I would prefer parallel
> rings, or at least for (*) to leave the main MSN alone, and only capture
> the others. Is this possible?

Yes

> 3) Are MSN's the same as DIDs for (*)?
>
Yes

> That's it for now, any help is appreciated...
>

To elaborate: My ZAPATA.CONF initially had immediate=yes. I changed it to
no yesterday, but had not yet rebooted.

Of course when I continued working on the MSN/DID issue, I had totally
forgotten that a reboot had not yet taken place, and as the zapata.conf
had immediate=no, I did not understand.

Of course, after a while something started nagging, and I realized that
the new ZAPATA config had not been loaded yet...

It now works great!

All MSN's/DID's coexist on the Siemens and (*) and I can even call from
the one to the other...

-- 
Francesco Peeters

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If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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Re: [Asterisk-Users] How to configure Asterisk through webmin

[Francesco Peeters]

On Thu, November 3, 2005 17:46, nr k said:
> Hi all
> I configured asterisk and webmin.i dont know how to
> integrate webmin with asterisk and how to access
> asterisk
> through webmin.pls do the needful.
>
> regards
> ramakrishnan.n
>
>

Asterisk is not managed through webmin. Webmin is a tool to help
administer the rest of the server.

There is supposedly a third party webmin module for Asterisk, but I have
not yet been able to get webmin to accept, install and execute it...
(Haven't put much time in, as AMP does a great job IMHO)

If you're starting with Asterisk, it may be helpful to use a dedicated
distro like [EMAIL PROTECTED] (http://asteriskathome.sourceforge.net)
Be careful though, as once you press enter on the boot prompt, it will
wipe the primary harddisk, install CentOS and reboot, after which it will
automatically compile Asterisk and it's modules.

Once it is back up, you have a running Asterisk installation, complete
with AMP and other gadgets...

Good luck!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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[Asterisk-Users] IAX2.FWDNET.NET not responding?

[Francesco Peeters]

Hi all,

Since a few days my (*) no longer seems to log in to FWD through IAX2.

IAX2 DEBUG only shows outbound registration requests, but no replies from
FWD:
Tx-Frame Retry[003] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
   Timestamp: 00014ms  SCall: 6  DCall: 0 [65.39.205.121:4569]
   USERNAME: 715749
   REFRESH : 60

It apparently doesn't reply to the lag requests either:

Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 000 Type: IAX Subclass: LAGRQ
   Timestamp: 30013ms  SCall: 6  DCall: 0 [65.39.205.121:4569]
Tx-Frame Retry[003] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ
   Timestamp: 10013ms  SCall: 6  DCall: 0 [65.39.205.121:4569]
Tx-Frame Retry[001] -- OSeqno: 004 ISeqno: 000 Type: IAX Subclass: LAGRQ
   Timestamp: 30013ms  SCall: 6  DCall: 0 [65.39.205.121:4569]
Tx-Frame Retry[002] -- OSeqno: 002 ISeqno: 000 Type: IAX Subclass: PING
   Timestamp: 20013ms  SCall: 6  DCall: 0 [65.39.205.121:4569]
Tx-Frame Retry[002] -- OSeqno: 003 ISeqno: 000 Type: IAX Subclass: LAGRQ
   Timestamp: 20016ms  SCall: 6  DCall: 0 [65.39.205.121:4569]

(Note the increasing retries)

It *does* connect to VoipBuster and GoIAX...

Is this most likely to be an issue at FWD, my account or something else?

TIA!

-- 
Francesco Peeters

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

[Francesco Peeters]

On Fri, November 4, 2005 11:27, David Acacio said:
> Hi,
>
> I have E1 PRI, When I have an incoming call, how can I know the called
> number (or the destination number) before answer the call?
>
> My provider say that he send it.
>
>   E1
> PRI
> 900XX > 9XXX --> Asterisk
>
>  It appears in some event under the Asterisk Manager API?
>
> Thanks,
>
> David
>

Log in to the CLI (if not on your main system, use 'asterisk -vr') and
watch for the incoming call.

If you want to do DID's you may have to put 'immediate=no' and
'overlapdial=yes' in the zap channel definition (zapata.conf) to ensure
that it waits to receive the DID info and put it in the appropriate
variable.

Do not forget to restart after changing zapata.conf. An 'asterisk -rx
reload' does NOT reload zapata conf!

Once it works, you should see something like this in the CLI:
-- Extension '0793429193' in context 'from-pstn' from '0174287114'
does not exist.  Rejecting call on channel 0/1, span 1

After that all you need to do is define an incoming extension with the
correct DID data, like:
[ext-did]
exten => 0123456788,1,SetVar(FROM_DID=0123456788)   ;
exten => 0123456788,2,Goto(ext-local,200,1) ;
exten => 0123456789,1,SetVar(FROM_DID=0123456789)   ;
exten => 0123456789,2,Goto(aa_1,s,1);

HTH!

-- 
Francesco Peeters

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