[Asterisk-Users] Digium T100P T1 Card

[Wiley Siler]

Hello All,


I could use a recommendation if anyone has a 
moment. I have the T100P but I have not gotten my service yet. I 
want to have at least 12 lines of digital voice with DID. Should I just 
seek out a PRI ISDN provider or is there something else I should look for? 
I want to keep cost as low as possible. Also, I want to own my own router 
for the phones since it is always a hassle to get anything fixed from the 
tele-company. What is a good and cheap router (Cisco maybe) that I would 
use to interface to the T100P? I plan to integrate my system to use our 
old analog lines for fax so I will have questions on that later 
too.

Thanks everyone!

Wiley___
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RE: [Asterisk-Users] Extensions to solve three way calling problem

[Steven Critchfield]

On Wed, 2005-01-05 at 18:38 +1100, PHP Mechanic wrote:
  Have you considered setting up a meetme confrence line for them? :)
 
   analog phone = asterisk/tdm11b = pstn
 
  The meetme option is nice, but it doesn't solve the problem. The TDM11B 
  only
  has one FXO, one FXS. To get the effect the daughter wants requires
  supporting the threeway facility the telco offers. You need to Flash the
  outside line. Zap does have an application for that, but I haven't played
  with what it can do, or how to program it.
 
 I have played with it. But the problem I'm having is as follows
 
 exten = _1800.,1,Dial(Zap/4/${EXTEN},20,Tr) ; call some company 
 willing to pay for my test, preferably get someone with an on hold message
 ; Now I press #* on the analog phone to transfer them to Meetme
 exten = *,1,Meetme,2000   ; send 
 them to meetme
 exten = *,2,Flash()  ; 
 flash the pstn line

What makes you think that would flash the PSTN line? This is your
problem. When you transfer the PSTN line anywhere and then go to dial
again, the flash is actually on the current channel. I wouldn't be
surprised if you hear it in your receiver. I don't know of anyway to
flash the PSTN line from within asterisk that would do as you want. In
fact, to enable it would be a security risk as well. Think of the
possibility of having multiple lines in and then dialing an extension to
flash the line and messing up and flashing someone else's connection. 

Closest thing I could think of is having your PSTN side caller do the
transfer and redial. If the PSTN caller was allowed to transfer the
inside person and then dial a special extension that would initiate the
flash and the dial command. Of course the trouble here is that as soon
as the flash occurs, the new caller is the one going to be stuck in an
odd state and the previous PSTN caller is going to be in unrecoverable
limbo.

Just looks like you will be SOL on utilizing the PSTN 3 way calling.  

-- 
Steven Critchfield [EMAIL PROTECTED]

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[Asterisk-Users] Asterisk with Euro ISDN, etc

[Daniel Nystrm]

Hi folks!

Ourcompanyare going to buy an E1 line 
with Euro ISDN and 30 lines (channels).

This is how it will be configured:
3 Lines, of the total of 30, is going to be for the 
company phones, and share one phonenumber (eg. 555-12340).
1 Line will be dedicated to a specific unique 
phonenumber (Fax) (eg. 555-54321).
The rest of the lines/channels (26) will be used by 
(by, not for) our customers, and will be redirected via our system to mobile 
devices (for more info: http://www.westel.se, 
and choose English in up-left corner).
The large "group" will use a range of 1000 
phonenumbers, which in turn will specify which mobile device it will redirect to 
(eg. 555-4 to 555-40999).

All lines/channels need to be connected to analog 
phones! And with the "large group", it has to deliver all Caller-ID and which of 
the thousend number was called. Preferably with DTMF. And on out-going calls, it 
also has to receive destination phonenumber, and preferably even it's own number 
(one of the thousend numbers).

Since we are going to provide unique numbers to each mobile device (just 
like a cellular), it requires alot of unusual features.
And about that out-going calls receiving the callers number, it would be 
nice to present the number from which mobile device the call is made. Again like 
regular cellulars.

We already have analoge interfacesfor our current exchanger, and each 
of them are connected to regular PSTN lines (each with individual accounts and 
numbers). That's why we need analog interface from Asterisk to our 
exchanger.

Is this possible with Asterisk?

Hope this wasn't too confusing. Just let me know if there are anything 
unclear, and I will try to explain it in a better way.

Happy new year!

Best Regards
Daniel 
Nystrm
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Re: [Asterisk-Users] Displaying incoming e.164 callers number - how?

[Mark Elkins]

On Tue, 2005-01-04 at 15:34 +0100, Erik Versaevel wrote:
 Mark Elkins wrote:
 On Tue, 2005-01-04 at 15:45 +0200, Mark Elkins wrote:
 On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
 I've got asterisk able to make and receive calls via the Internet via
 E164 lookups. If I get such a call - I'd like to display the original

 Playing with myself again - that is - I called myself - and
 got the caller ID of '27128070590'... not quite what I wanted...
 
 In my extensions - I have...
 [fromaix]
 exten = 27128070590,1,Goto(default,s,1)

 And again - changed the above to...
 [fromaix]
 exten = 27128070590,1,GotoIf($[${CALLERIDNUM:0:2} = 27]?2:4)
...
 Default section looks like...
 [default]   ; what people will get when they call me.
 exten = s,1,NoOp(CALLER=${CALLERIDNUM})
 exten = s,2,Answer()

 how about SetCallerId(12345) ;)
 ie
 exten= 27128070590, 2, setcallerid(0${CALLERIDNUM});

This works fine... Thanks.
Incoming AIX looks like...
[fromaix]
exten = 27128070590,1,GotoIf($[${CALLERIDNUM:0:2} = 27]?2:4)
exten = 27128070590,2,setcallerid(0${CALLERIDNUM:2})
exten = 27128070590,3,Goto(default,s,1)
exten = 27128070590,4,setcallerid(09${CALLERIDNUM})
exten = 27128070590,5,Goto(default,s,1)

... and does the right thing...

Of course - this depends on people making e.164+VoIP calls to me
actually setting their Caller ID according to the format '27128070590' -
ie - No plus signs (as for cell/mobile phones), no '00' (or other access
code for international dialling - just the country dialing code followed
by their whole dialing code...   
-- 
  .  . ___. .__  Posix Systems - Sth Africa.  e.164 VOIP ready
 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

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RE: [Asterisk-Users] Extensions to solve three way calling problem

[PHP Mechanic]

On Wed, 2005-01-05 at 18:38 +1100, PHP Mechanic wrote:
 Have you considered setting up a meetme confrence line for them? :)

  analog phone = asterisk/tdm11b = pstn
I have played with it. But the problem I'm having is as follows
exten = _1800.,1,Dial(Zap/4/${EXTEN},20,Tr) ; call some company
willing to pay for my test, preferably get someone with an on hold 
message
; Now I press #* on the analog phone to transfer them to Meetme
exten = *,1,Meetme,2000   ; send
them to meetme
exten = *,2,Flash() 
;
flash the pstn line
What makes you think that would flash the PSTN line?
Because the cli reports that it is executing flash on the Zap/4 - the PSTN 
line

This is your
problem. When you transfer the PSTN line anywhere and then go to dial
again, the flash is actually on the current channel. I wouldn't be
surprised if you hear it in your receiver. I don't know of anyway to
flash the PSTN line from within asterisk that would do as you want. In
fact, to enable it would be a security risk as well. Think of the
possibility of having multiple lines in and then dialing an extension to
flash the line and messing up and flashing someone else's connection.
Closest thing I could think of is having your PSTN side caller do the
transfer and redial. If the PSTN caller was allowed to transfer the
inside person and then dial a special extension that would initiate the
flash and the dial command. Of course the trouble here is that as soon
as the flash occurs, the new caller is the one going to be stuck in an
odd state and the previous PSTN caller is going to be in unrecoverable
limbo.
Just looks like you will be SOL on utilizing the PSTN 3 way calling.
Yeah, I think you are right.
But what is the point of  threewaycalling and transfer in zapata.conf - what 
do they do? 

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Re: [Asterisk-Users] Digium T100P T1 Card

[Steven Critchfield]

On Wed, 2005-01-05 at 01:01 -0700, Wiley Siler wrote:
 Hello All,
  
 I could use a recommendation if anyone has a moment. 

It is preferable to not use HTML in email. Just because a font size
looks good on your monitor doesn't mean it is anywhere close to good
anywhere else. Your choosen font size ends up being 1/2 to maybe 2/3 the
size of standard text. In other words, had it not been for the slightly
interesting to me subject line, this would have been ignored as a user
to clueless to bother with.

Also learn about why paragraphs are good.

  I have the T100P but I have not gotten my service yet.  I want to
 have at least 12 lines of digital voice with DID.  Should I just seek
 out a PRI ISDN provider or is there something else I should look for?
 I want to keep cost as low as possible.  Also, I want to own my own
 router for the phones since it is always a hassle to get anything
 fixed from the tele-company.

This shows you need to learn quite a bit more about phone service. There
is no need for a router on a telephony T1. You will either want to plug
a channelized T1 or PRI into the T100P directly. 

If you want some analog FXS ports, you could also go the route of an
ADIT 600 and plug the T1 into the ADIT and route your incoming 12
channels to the second port of the ADIT and then plug it into the T100P.
The benefit here is you will have 12 channels left over to signal back
from the T100P to the ADIT and have those channels routed to FXS ports.
I used to do something similar to that with a Zhone channel bank before
our company fully trusted asterisk.

   What is a good and cheap router (Cisco maybe) that I would use to
 interface to the T100P?  I plan to integrate my system to use our old
 analog lines for fax so I will have questions on that later too.

You don't need analog lines for FAX. Follow the directions above for the
ADIT and you will be able to have analog ports to plug your fax machines
and route them out the T1. 
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] Call(out) routing

[steve]

On Wed, 5 Jan 2005, Altus Snyman wrote:

 Good day all
 I had a look at the extensions.conf sorting
 http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting
 
 What I'm trying to do is route all my cellphone number threw a  channel
 and all other calls threw the other 3 channels
 Cellphone numbers are 10 number,i.o.w XX.
 This is what I tried but it doesn't seem to work,please help
 Thanks
 Altus
 extensions.conf



Hi Altus,

You;ve done your includes exactly backwards.  The most generic pattern 
must be included at the bottom of the pile, as it were.

In other wrds

[dialout-telkom]
exten = _0.,1,Dial(telkom...)

[dialout]
include = dialout-telkom
exten = _0[78]2.,1,Dial(vodacom)
exten = _0[78]3.,1,Dial(mtn)
etc

The reason is that Asterisk only follows the include links when it can't 
find a match in the current context.

Regards,
Steve Davies
Connection-Telecom CC
Cape Town, South Africa

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RE: [Asterisk-Users] Extensions to solve three way calling problem

[Steven Critchfield]

On Wed, 2005-01-05 at 19:27 +1100, PHP Mechanic wrote:
  On Wed, 2005-01-05 at 18:38 +1100, PHP Mechanic wrote:
   Have you considered setting up a meetme confrence line for them? :)
  
analog phone = asterisk/tdm11b = pstn
 
  I have played with it. But the problem I'm having is as follows
 
  exten = _1800.,1,Dial(Zap/4/${EXTEN},20,Tr) ; call some company
  willing to pay for my test, preferably get someone with an on hold 
  message
  ; Now I press #* on the analog phone to transfer them to Meetme
  exten = *,1,Meetme,2000   ; send
  them to meetme
  exten = *,2,Flash() 
  ;
  flash the pstn line
 
  What makes you think that would flash the PSTN line?
 
 Because the cli reports that it is executing flash on the Zap/4 - the PSTN 
 line
 
 This is your
  problem. When you transfer the PSTN line anywhere and then go to dial
  again, the flash is actually on the current channel. I wouldn't be
  surprised if you hear it in your receiver. I don't know of anyway to
  flash the PSTN line from within asterisk that would do as you want. In
  fact, to enable it would be a security risk as well. Think of the
  possibility of having multiple lines in and then dialing an extension to
  flash the line and messing up and flashing someone else's connection.
 
  Closest thing I could think of is having your PSTN side caller do the
  transfer and redial. If the PSTN caller was allowed to transfer the
  inside person and then dial a special extension that would initiate the
  flash and the dial command. Of course the trouble here is that as soon
  as the flash occurs, the new caller is the one going to be stuck in an
  odd state and the previous PSTN caller is going to be in unrecoverable
  limbo.
 
  Just looks like you will be SOL on utilizing the PSTN 3 way calling.
 
 Yeah, I think you are right.
 
 But what is the point of  threewaycalling and transfer in zapata.conf - what 
 do they do? 

All of it is for doing stuff within asterisk. For example transfer is
for if you have more than one station inside the PBX, then you could
transfer the call from one phone to the other.

Threeway calling is similar. You can make a small impromptu conference
that way with 2 internal phones and an external or 3 internal phones or
even 1 internal and 2 external calls on separate phone lines. All of
these are mixed inside of asterisk and the PSTN is non the wiser.

-- 
Steven Critchfield [EMAIL PROTECTED]

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RE: [Asterisk-Users] Extensions to solve three way calling problem

[PHP Mechanic]

 Threeway calling is similar. You can make a small impromptu conference
that way with 2 internal phones and an external or 3 internal phones or
even 1 internal and 2 external calls on separate phone lines. All of
these are mixed inside of asterisk and the PSTN is non the wiser.
Thanks for clearing this up for me. The thing that still get's me is that 
the pstn is a little bit wise. It can perform the following:

1. Establish a call with the first person. You can call them or they can 
call you.
2. Press Flash/Recall on phone to put the first person on hold.
3. Wait until you hear the dial tone.
4. Dial the number of the second person.
5. Wait until you hear the second line ringing.
6. Press Flash/Recall and talk to the first person (they will hear the 
ringing tone too).

Can I make asterisk play ball with my telco? 

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[Asterisk-Users] Speex codec problem (unresolved ?)

[Walter Klomp]

Hi,

I'm sorry to bring this up again, but I have been googling forever and
whatever solutions are offered don't work for me.

I am using x-lite (the latest build) and trying to use Speex.

When I do call from the x-lite to another SIP phone or PSTN (through Cisco
gateway) My asterisk fills up with this message:
WARNING[1007]: codec_speex.c:196 speextolin_framein: Out of buffer space

The x-lite client can hear the remote end (SIP or PSTN call) quite clearly,
but what comes from the X-Lite is completely garbled and mixed with DTMF
tones.

I had tried the registry fix (which only changes the magic number from 97 to
110 and apparently didn't do anything else), didn't work.

After looking at the source I had also tried to increase the buffer size
from 8000 to 16000, but that made other codecs (like lin_to_g729) choke, and
I still had the problem...

I like speex and would like to use it (as I find ilbc a bit too scratchy)

I am running Asterisk CVS-HEAD-11/16/04-17:19:53 and speex-1.0.4 libraries
on Gentoo Linux.

Can anybody help me further on how to resolve this problem ?

Thanks
Walter

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Re: [Asterisk-Users] Speex codec problem (unresolved ?)

[David Uzzell]

Walter Klomp wrote:
Hi,
I'm sorry to bring this up again, but I have been googling forever and
whatever solutions are offered don't work for me.
I am using x-lite (the latest build) and trying to use Speex.
When I do call from the x-lite to another SIP phone or PSTN (through Cisco
gateway) My asterisk fills up with this message:
WARNING[1007]: codec_speex.c:196 speextolin_framein: Out of buffer space
The x-lite client can hear the remote end (SIP or PSTN call) quite clearly,
but what comes from the X-Lite is completely garbled and mixed with DTMF
tones.
I had tried the registry fix (which only changes the magic number from 97 to
110 and apparently didn't do anything else), didn't work.
After looking at the source I had also tried to increase the buffer size
from 8000 to 16000, but that made other codecs (like lin_to_g729) choke, and
I still had the problem...
I like speex and would like to use it (as I find ilbc a bit too scratchy)
I am running Asterisk CVS-HEAD-11/16/04-17:19:53 and speex-1.0.4 libraries
on Gentoo Linux.
The best sugestion that I can offer is that I saw the same problem and 
could not resolve it but after upgrading * to CVS after the 12/10 it 
went away. Never did find a solution and gave up looking as it solved it.

It also fixed some SIP issues I had and they went away aswell.
Sorry that might not be the answer you are looking for but thats what 
worked for me.

David
Can anybody help me further on how to resolve this problem ?
Thanks
Walter
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Re: [Asterisk-Users] Kirk SIP-DECT gateway

[Remco Barende]

I already did testing with the unit you are talking about and the 
status is that it is NOT working.

Search the list on IP600 and asterisk.
In brief:
Their Skinny implementation makes the IP600 wait for an extra signal and 
refuses to ring any handset without it. The 'original' Cisco 7940 (which 
it emulates) doesn't require this signal. If you use chan_sccp the phones 
will not register (don't know why yet) and Jan Czmok was't sure if this 
extra signal (STATION_CALLINFO) is in chan_sccp anyway. Outgoing calls 
work but soundlevel is too low.

H323 : Works for incoming and outgoing calls but none of the call features 
(hold, transfer etc.) work. This is too little funcitionaly to say that it 
works.

If have had extensive contact about these issues with Kirk support. At the 
moment the units are very interesting but useless with *.

The only thing we can do is keep asking for support to all the Kirk 
offices in every country or to keep enquiring their sales. I can provide 
tcpdumps, logs etc. if anyone is interested.

I haven't heard anything about SIP support, I have it will be in the next 
firmware release for the IP600.

Cheers!
Remco
On Tue, 4 Jan 2005, E s c a u x - Jordi Nelissen wrote:
Hi,
I just got some interesting information from Kirk Telecom
(www.kirktelecom.com). This company has been in the business of providing
DECT solutions (IP gateway, base stations, repeaters and handsets) either to
be used with Cisco CallManager (SCCP protocol) or with the Innovaphone IP
PBX system (H.323).
Two important elements:
1. It seems they foresee a SIP version of their product in Q1 2005.
2. They are open to perform integration tests with asterisk, provided there
is sufficient business potential.
In order to convince them about the business potential of an asterisk
integration, I would like to ask you to drop me an email stating the number
of DECT installations and associated DECT phones you might be able to sell
in 2005, provided the Kirk solution proved to interoperate with asterisk
(either H323 or SIP).
Best regards,
Jordi
--
w w w . e s c a u x . c o m
IPTel : 02 686 09 02
IPFax : 02 686 09 08
Email : [EMAIL PROTECTED]

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Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

[Eric Bishop]

I will certainly try that. Please also let me know your progress..


On Tue, 4 Jan 2005 22:12:23 +0200 (SAST), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
 
 
 On Mon, 3 Jan 2005 [EMAIL PROTECTED] wrote:
 
  Has anyone had success using a TE410P card in an HP-Compaq DL380 G4
  server?
 
 Thanks for all the postings on this thread.
 
 I have a new and completely untested theory - the G4 has the option of a
 non-hotplug or a hotplug PCI riser cage.  My new theory is that my
 problems with the TE410P have something to do with hotplug.
 
 Now I don't know what riser cage my customer ordered, but I see it has two
 slots with quick release clips holding the cards in, one with a screw.
 So just maybe the quick-release implies hotplug.
 
 I don't know much about PCI-Hotplug, but there's an option in the kernel
 config about having a Compaq PCI Hotplug controller.  Do I have one of
 those?...
 
 I see some irq related stuff in the source of the compaq hotplug driver in
 the kernel - so perhaps I'm not loading that, or it doesn't work on the
 DL380 G4 right...
 
 Maybe even if I don't have the hotplug riser I still need this driver...
 
 Steve
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RE: [Asterisk-Users] Kirk SIP-DECT gateway

[B. Vallet - www.acropolistelecom.net]

We have SIP-DECT gateways in Pci:
http://shop.acropolistelecom.net/product_info.php?products_id=30language=en
or PCMCIA cards :
http://shop.acropolistelecom.net/product_info.php?manufacturers_id=11produc
ts_id=29

Regards

Benoit


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Remco Barende
Envoyé : mercredi 5 janvier 2005 11:29
À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Objet : Re: [Asterisk-Users] Kirk SIP-DECT gateway

I already did testing with the unit you are talking about and the 
status is that it is NOT working.

Search the list on IP600 and asterisk.

In brief:
Their Skinny implementation makes the IP600 wait for an extra signal and 
refuses to ring any handset without it. The 'original' Cisco 7940 (which 
it emulates) doesn't require this signal. If you use chan_sccp the phones 
will not register (don't know why yet) and Jan Czmok was't sure if this 
extra signal (STATION_CALLINFO) is in chan_sccp anyway. Outgoing calls 
work but soundlevel is too low.

H323 : Works for incoming and outgoing calls but none of the call features 
(hold, transfer etc.) work. This is too little funcitionaly to say that it 
works.

If have had extensive contact about these issues with Kirk support. At the 
moment the units are very interesting but useless with *.

The only thing we can do is keep asking for support to all the Kirk 
offices in every country or to keep enquiring their sales. I can provide 
tcpdumps, logs etc. if anyone is interested.

I haven't heard anything about SIP support, I have it will be in the next 
firmware release for the IP600.

Cheers!
Remco

On Tue, 4 Jan 2005, E s c a u x - Jordi Nelissen wrote:

 Hi,

 I just got some interesting information from Kirk Telecom
 (www.kirktelecom.com). This company has been in the business of providing
 DECT solutions (IP gateway, base stations, repeaters and handsets) either
to
 be used with Cisco CallManager (SCCP protocol) or with the Innovaphone IP
 PBX system (H.323).

 Two important elements:

 1. It seems they foresee a SIP version of their product in Q1 2005.
 2. They are open to perform integration tests with asterisk, provided
there
 is sufficient business potential.

 In order to convince them about the business potential of an asterisk
 integration, I would like to ask you to drop me an email stating the
number
 of DECT installations and associated DECT phones you might be able to sell
 in 2005, provided the Kirk solution proved to interoperate with asterisk
 (either H323 or SIP).

 Best regards,

 Jordi
 --
 w w w . e s c a u x . c o m

 IPTel : 02 686 09 02
 IPFax : 02 686 09 08
 Email : [EMAIL PROTECTED]



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Re: [Asterisk-Users] Status of SNOM Intercom

[Nils Ohlmeier]

On Tuesday 04 January 2005 14:06, Peer Oliver Schmidt wrote:
 Nils Ohlmeier wrote:
 What are the chances to get intercom working with a Snom 190 with a
 current firmware? Anyone? Is Snom working on this?
 
  yes, Snom is working on this.

 This is very good to hear. Do you have any time frame (2 weeks / 2 month
 / 6 month / 2 years)? Information for this is fairly important, as I
 have another interested party to be deployed during the June/July time
 frame, which needs intercom functionality.

It is allready fixed. So it should work again in the next firmware release 
(which usually results in an availability for the end users in terms of days 
or at max in weeks).

BTW the new snom model (no availability dates yet) will have more LED's then 
the existing ones. What are the most important or interesting features 
(existing or not implemented yet) for the Asterisk community for this 
programmable keys with LED's?

Regards
  Nils Ohlmeier
-- 
snom technology AGPascalstrasse 10bD-10581 Berlin
Nils Ohlmeier
mailto:[EMAIL PROTECTED]  http://www.snom.com
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RE: [Asterisk-Users] Kirk SIP-DECT gateway

[Remco Barende]

Thanks, the price looks attractive but what about Asterisk support?
From your website : All standard Windows operating systems are 
supported. and we are not using Windoze :)
How many concurrent conversations are supported?
Is there a howto anywhere how this could be used with * ?
By looking at the website of the manufacturer this product isn't really 
comparable with the IP600 which has support for 8 simultaneous calls and 
can be used with repeaters to extend it's range.

For small SOHO setups it would be quite nice however.
Cheers!
Remco
On Wed, 5 Jan 2005, B. Vallet - www.acropolistelecom.net wrote:
We have SIP-DECT gateways in Pci:
http://shop.acropolistelecom.net/product_info.php?products_id=30language=en
or PCMCIA cards :
http://shop.acropolistelecom.net/product_info.php?manufacturers_id=11produc
ts_id=29
Regards
Benoit
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Remco Barende
Envoyé : mercredi 5 janvier 2005 11:29
À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Objet : Re: [Asterisk-Users] Kirk SIP-DECT gateway
I already did testing with the unit you are talking about and the
status is that it is NOT working.
Search the list on IP600 and asterisk.
In brief:
Their Skinny implementation makes the IP600 wait for an extra signal and
refuses to ring any handset without it. The 'original' Cisco 7940 (which
it emulates) doesn't require this signal. If you use chan_sccp the phones
will not register (don't know why yet) and Jan Czmok was't sure if this
extra signal (STATION_CALLINFO) is in chan_sccp anyway. Outgoing calls
work but soundlevel is too low.
H323 : Works for incoming and outgoing calls but none of the call features
(hold, transfer etc.) work. This is too little funcitionaly to say that it
works.
If have had extensive contact about these issues with Kirk support. At the
moment the units are very interesting but useless with *.
The only thing we can do is keep asking for support to all the Kirk
offices in every country or to keep enquiring their sales. I can provide
tcpdumps, logs etc. if anyone is interested.
I haven't heard anything about SIP support, I have it will be in the next
firmware release for the IP600.
Cheers!
Remco
On Tue, 4 Jan 2005, E s c a u x - Jordi Nelissen wrote:
Hi,
I just got some interesting information from Kirk Telecom
(www.kirktelecom.com). This company has been in the business of providing
DECT solutions (IP gateway, base stations, repeaters and handsets) either
to
be used with Cisco CallManager (SCCP protocol) or with the Innovaphone IP
PBX system (H.323).
Two important elements:
1. It seems they foresee a SIP version of their product in Q1 2005.
2. They are open to perform integration tests with asterisk, provided
there
is sufficient business potential.
In order to convince them about the business potential of an asterisk
integration, I would like to ask you to drop me an email stating the
number
of DECT installations and associated DECT phones you might be able to sell
in 2005, provided the Kirk solution proved to interoperate with asterisk
(either H323 or SIP).
Best regards,
Jordi
--
w w w . e s c a u x . c o m
IPTel : 02 686 09 02
IPFax : 02 686 09 08
Email : [EMAIL PROTECTED]

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Re: [Asterisk-Users] Can't initiate a call with X-Lite.

[Rich Adamson]

   I'm trying to place a call to asterisk using X-Lite. Asterisk is  setup 
 with some Grandstream phones. I can call from one grandstream extension 
 to another. When I try to an extension with X-Lite, it comes back with 
 Status of SIP/2.0 404 Not Found. X-Lite is not registered as asterisk 
 extension. It is just sending a sip invite to [EMAIL PROTECTED] Does the 
 X-Lite need to connect to via a proxy?

No. You should work on configuring xlite to register with asterisk.
In the xlite Sip Proxy menu, you will need a User Name, Password,
Sip Proxy, and Domain/Realm defined to match entries in your
sip.conf definitions.

Your sip.conf for xlite should look something like:
[3005]
type=friend
host=dynamic
username=3005
secret=yourpassword
context=from-sip
canreinvite=no
mailbox=3005 
 
 After several days of reading RFCs and looking at packet traces, I know 
 a bit more about SIP, but not quite enough to make this work.
 
 Is there a way to get asterisk to say what its doing? I tried 
 -vv etc, but the only messages are see are when I use one of my 
 my Grandstream phones. On the wire, is see the same To:  header from 
 both the grandstream and the X-Lite soft phone. I don't understand why 
 its found by one, and not the other.

From your asterisk CLI, try sip debug to see the flow of packets to/from
asterisk; sip no debug will shut it off.



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[Asterisk-Users] Cannot Hear at all

[enunes]

Hi all,

I am attempting to call from softphone to softphone, I am using X-lite to call
another X-lite.

I get the phones to call each other and finnaly connecting, but cannot hear the
voice at all. Is there any ideas as to why this is happening. 

(I don't have sound card in my linux server. I need one in my linux server ??)

PS: callonhold is working but cannot hear the music too.

look at my messages file:


Dec 21 22:31:36 NOTICE[65540]: Request to schedule in the past?!?!
Dec 21 22:31:36 WARNING[16384]: Unable to get our IP address, Skinny disabled
Dec 21 22:31:36 NOTICE[65540]: Request to schedule in the past?!?!
Dec 21 22:31:37 NOTICE[65540]: Request to schedule in the past?!?!
Dec 21 22:31:37 WARNING[16384]: Unable to open /dev/dsp: No such device   
Dec 21 22:31:37 NOTICE[65540]: Request to schedule in the past?!?!
Dec 21 22:32:46 NOTICE[213005]: RFC3389 support incomplete.  Turn off on client
if possible
Dec 21 22:34:42 WARNING[16384]: Unable to get our IP address, Skinny disabled
Dec 21 22:34:42 NOTICE[65540]: Request to schedule in the past?!?!
Dec 21 22:34:42 NOTICE[65540]: Request to schedule in the past?!?!
Dec 21 22:34:42 WARNING[16384]: Unable to open /dev/dsp: No such device
Dec 21 22:34:43 NOTICE[65540]: Request to schedule in the past?!?!
Dec 21 22:34:43 NOTICE[65540]: Request to schedule in the past?!?!
Dec 21 22:35:51 NOTICE[213005]: RFC3389 support incomplete.  Turn off on client


Thanks,

 

Eduardo Nunes


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[Asterisk-Users] New asterisk installation but no audible voicemail prompts?

[Remco Barende]

Hi List!
I installed Asterisk 1.0.3 stable on a RHEL rebuild. Due to problems with 
* modules refusing to build I replaced the RHEL kernel with stock 2.6.10.

Asterisk seems to be working but when I dial voicemail I hear nothing. 
When I hangup I see a message on the console that the calller did not 
specify a mailbox number so I guess voicemail app is working.

The phone(Grandstream BT100) is connected directly to the * server so it's 
not any NAT or firewall trouble (no firewall installed).

Any ideas?  Which kernel options are required for Asterisk to function 
properly. Any recommendations on that?

Several options do come to mind like, also to prevent timing problems 
like:
- HPET Timer Support (CONFIG_HPET_TIMER)
- Provide RTC interrupt (CONFIG_HPET_EMULATE_RTC)
- Preemptible Kernel (CONFIG_PREEMPT)
  (even though the info in kernel describes this for desktop)
- Message Signaled Interrupts (MSI and MSI-X) (CONFIG_PCI_MSI)
- Enhanced Real Time Clock Support (CONFIG_RTC)
- HPET - High Precision Event Timer (CONFIG_HPET)

For general Astrisk with ISDN operation:
Is telephony support and ISDN support in the kernel required?
Thanks for any suggestions!
Remco
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[Asterisk-Users] chan_cornet

[Joao Pereira]

Hi
but did anyone have ever used a Siemens HiPath PBX with Asterisk?
If you made it, please tell me how...

I read that chan_cornet does exist...
http://lists.digium.com/pipermail/asterisk-users/2004-October/069559.html

Is there any Digium Hardware solution for the Asterisk HiPath connection?

Thanks
Joao Pereira

- Original Message -
From: Luís Palma [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 04, 2005 10:30 PM
Subject: Re: [Asterisk-Users] connect Asterisk with Siemens HiPath HG1500


 Hi,

 It doesn't tell you much but it looks like that you are not alone when
 trying to integrate with Siemens Hicom. It seems someone has decided
 to make it by himself.

 http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom

 Regards
 Luis Palma

 On Tue, 4 Jan 2005 12:26:38 -, Joao Pereira [EMAIL PROTECTED]
wrote:
  Hi
  I want to know the best way to connect Asterisk to a Siemens HiPath
HG1500
  PBX. Until now I came out with 3 solutions:
 
  1-Asterisk being a H.323 client of the Siemens PBX (I believe it needs
  Siemens licences and Digium hardware)
  2-Asterisk connecting to the PSTN phones with Voice Modems (good
ideia!!!
  but its analog... doesnt have caller information...)
  3-Using RDIS interfaces to connect the Siemens PBX
 
  does someone have other ideias?
 
  Thanks
  Joao Pereira
 
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Re: [Asterisk-Users] New asterisk installation but no audible voicemail prompts?

[John Middleton]

use - on the command line for debugging information, there
should be detailed tracking information provided that will help


On Wed, 5 Jan 2005 12:41:23 +0100 (CET), Remco Barende
[EMAIL PROTECTED] wrote:
 Hi List!
 
 I installed Asterisk 1.0.3 stable on a RHEL rebuild. Due to problems with
 * modules refusing to build I replaced the RHEL kernel with stock 2.6.10.
 
 Asterisk seems to be working but when I dial voicemail I hear nothing.
 When I hangup I see a message on the console that the calller did not
 specify a mailbox number so I guess voicemail app is working.
 
 The phone(Grandstream BT100) is connected directly to the * server so it's
 not any NAT or firewall trouble (no firewall installed).
 
 Any ideas?  Which kernel options are required for Asterisk to function
 properly. Any recommendations on that?
 
 Several options do come to mind like, also to prevent timing problems
 like:
 - HPET Timer Support (CONFIG_HPET_TIMER)
 - Provide RTC interrupt (CONFIG_HPET_EMULATE_RTC)
 - Preemptible Kernel (CONFIG_PREEMPT)
   (even though the info in kernel describes this for desktop)
 - Message Signaled Interrupts (MSI and MSI-X) (CONFIG_PCI_MSI)
 - Enhanced Real Time Clock Support (CONFIG_RTC)
 - HPET - High Precision Event Timer (CONFIG_HPET)
 
 For general Astrisk with ISDN operation:
 Is telephony support and ISDN support in the kernel required?
 
 Thanks for any suggestions!
 
 Remco
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Re: [Asterisk-Users] CallerID in Australia Analogue PSTN Phone

[Howard Lowndes]

On Wed, 2005-01-05 at 17:02, PHP Mechanic wrote:
  Howard Lowndes wrote:
  Is there anyone using * in AU that has successfully extracted the CLID
  from an incoming analogue PSTN phone call, and would like to spread the
  word?
 
  Yes, I have. I'm using a Voicetronix OpenLine4 card. I did have an issue 
  with the config file, you have to set callerid=yes before each channel, 
  unless you're running CVS from 2004/12/13 21:04:12 or later. What hardware 
  are you using? chan_vpb has useful debugging info for callerid at debug 
  level 4.
 
 I fixed it using this: 
 http://www.voip-info.org/tiki-index.php?page=Asterisk+and+Australian+Caller+ID
 

Done that.

What I need more though is examples of anything that needs to go into
extensions.conf 

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LANNet Computing Associates;
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--
When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.


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[Asterisk-Users] Re: How can I silently use ASTCC?

[Barry Flanagan]

Ronald Wiplinger wrote:

 The idea:
 If I have a customer, who is registered to my Asterisk box as extension,
 than I should not need to ask him for a pin code for ASTCC.
 
 How can I set this up?
 
Have a look at the astcc.agi file:

 # Usage-example:
#
# ;
# ; Card-number and number to dial derived from command-line.
# ; Call script with the card-number as first arg and the number
# ; to dial as the second arg.
# ;
# exten = _00X,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
# exten = _00X,2,Hangup
#

...so simply create cards with numbers that equal your extension numbers

Hope this helps.


-Barry Flanagan
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Re: [Asterisk-Users] chan_cornet

[richard Coco]

Hi,

The HG1500 is a HiPath3000 board and i don't have experience with Asterisk and HiPath3K.
What we have is an Asterisk connected to a Siemens HiPath4000 over a H.323 trunk using oh323 and the HG3550 board. It works fine. But the Siemens HG3550 only supports H.323 V2.0 (so not a lot of features are available). May be Steffen's chan_cornet will change this.
Are there any news about this project?Joao Pereira [EMAIL PROTECTED] wrote:
Hibut did anyone have ever used a Siemens HiPath PBX with Asterisk?If you made it, please tell me how...I read that chan_cornet does exist...http://lists.digium.com/pipermail/asterisk-users/2004-October/069559.htmlIs there any Digium Hardware solution for the Asterisk HiPath connection?ThanksJoao Pereira- Original Message -From: "Luís Palma" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"Sent: Tuesday, January 04, 2005 10:30 PMSubject: Re: [Asterisk-Users] connect Asterisk with Siemens HiPath HG1500 Hi, It doesn't tell you much but it looks like that you are not alone when trying to integrate with Siemens Hicom. It seems someone has decided to make it by himself.
 http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom Regards Luis Palma On Tue, 4 Jan 2005 12:26:38 -, Joao Pereira <[EMAIL PROTECTED]>wrote:  Hi  I want to know the best way to connect Asterisk to a Siemens HiPathHG1500  PBX. Until now I came out with 3 solutions:   1-Asterisk being a H.323 client of the Siemens PBX (I believe it needs  Siemens licences and Digium hardware)  2-Asterisk connecting to the PSTN phones with Voice Modems (goodideia!!!  but its analog... doesnt have caller information...)  3-Using RDIS interfaces to connect the Siemens PBX   does someone have other ideias?   Thanks  Joao Pereira   ___  Asterisk-Users mailing list 
 Asterisk-Users@lists.digium.com  http://lists.digium.com/mailman/listinfo/asterisk-users  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users  ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] CallerID in Australia Analogue PSTN Phone

[PHP Mechanic]

 Is there anyone using * in AU that has successfully extracted the CLID
 from an incoming analogue PSTN phone call, and would like to spread 
 the
 word?
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten = s,1,NoOp(Caller ID on the PSTN line is ${CALLERID}) 

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Re: [Asterisk-Users] CallerID in Australia Analogue PSTN Phone System

[Howard Lowndes]

On Wed, 2005-01-05 at 16:50, James Andrewartha wrote:
 Howard Lowndes wrote:
  Is there anyone using * in AU that has successfully extracted the CLID
  from an incoming analogue PSTN phone call, and would like to spread the
  word?
 
 Yes, I have. I'm using a Voicetronix OpenLine4 card. I did have an issue 
 with the config file, you have to set callerid=yes before each channel, 

done

 unless you're running CVS from 2004/12/13 21:04:12 or later.

later

  What hardware 
 are you using? chan_vpb has useful debugging info for callerid at debug level 
 4.

X101P

My callerid settings from zapata.conf are:

usecallerid = yes
callerid = yes
useincomingcalleridonzaptransfer = yes


 
 James Andrewartha
 DAA Sysadmin
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--
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Get rid of the Australian states.


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Re: [Asterisk-Users] chan_cornet

[Joao Pereira]

Hi
I dont knowif Steffen's chan_cornet is working. I emailed him, but with no 
result.
Yesterday I read this article
http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom

It has some solutions... but not yet a direct 
Asterisk-HiPath connection.

But doesnt Digium have Asterisk-HiPath 
solutions?

Joao

  - Original Message - 
  From: 
  richard 
  Coco 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, January 05, 2005 12:13 
  PM
  Subject: Re: [Asterisk-Users] chan_cornet 
  
  
  Hi,
  
  The HG1500 is a HiPath3000 board and i don't have experience with 
  Asterisk and HiPath3K.
  What we have is an Asterisk connected to a Siemens HiPath4000 over a 
  H.323 trunk using oh323 and the HG3550 board. It works fine. But the Siemens 
  HG3550 only supports H.323 V2.0 (so not a lot of features are available). May 
  be Steffen's chan_cornet will change this.
  Are there any news about this project?Joao Pereira 
  [EMAIL PROTECTED] wrote:
  Hibut 
did anyone have ever used a Siemens HiPath PBX with Asterisk?If you made 
it, please tell me how...I read that chan_cornet does 
exist...http://lists.digium.com/pipermail/asterisk-users/2004-October/069559.htmlIs 
there any Digium Hardware solution for the Asterisk HiPath 
connection?ThanksJoao Pereira- Original Message 
-From: "Luís Palma" <[EMAIL PROTECTED]>To: "Asterisk Users 
Mailing List - Non-Commercial 
Discussion"Sent: Tuesday, January 
04, 2005 10:30 PMSubject: Re: [Asterisk-Users] connect Asterisk with 
Siemens HiPath HG1500 Hi, It doesn't tell 
you much but it looks like that you are not alone when trying to 
integrate with Siemens Hicom. It seems someone has decided to make 
it by himself. 
http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom 
Regards Luis Palma On Tue, 4 Jan 2005 12:26:38 
-, Joao Pereira <[EMAIL PROTECTED]>wrote:  Hi 
 I want to know the best way to connect Asterisk to a Siemens 
HiPathHG1500  PBX. Until now I came out with 3 
solutions:   1-Asterisk being a H.323 client of the 
Siemens PBX (I believe it needs  Siemens licences and Digium 
hardware)  2-Asterisk connecting to the PSTN phones with Voice 
Modems (goodideia!!!  but its analog... doesnt have caller 
information...)  3-Using RDIS interfaces to connect the Siemens 
PBX   does someone have other ideias? 
  Thanks  Joao Pereira   
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Re: [Asterisk-Users] Status of SNOM Intercom

[Peer Oliver Schmidt]

Nils Ohlmeier wrote:
What are the chances to get intercom working with a Snom 190 with a
current firmware? Anyone? Is Snom working on this?
[..]
It is allready fixed. So it should work again in the next firmware release 
(which usually results in an availability for the end users in terms of days 
or at max in weeks).
Using the new firmware is there still the issue with needing to patch 
chan_sip.c, or does it work out of the box? Do you have details on how 
it should be implemented within *?
--
Best regards

Peer Oliver Schmidt
the internet company
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RE: [Asterisk-Users] Extensions to solve three way calling problem

[Michael Graves]

On Wed, 5 Jan 2005 07:54:43 +0100, Florian Overkamp wrote:

Hi, 

 -Original Message-
 Have you considered setting up a meetme confrence line for them? :)

  analog phone = asterisk/tdm11b = pstn

The meetme option is nice, but it doesn't solve the problem. The TDM11B only
has one FXO, one FXS. To get the effect the daughter wants requires
supporting the threeway facility the telco offers. You need to Flash the
outside line. Zap does have an application for that, but I haven't played
with what it can do, or how to program it.

*CLI show application Flash

  -= Info about application 'Flash' =-

[Synopsis]:
Flashes a Zap Trunk

[Description]:
  Flash(): Sends a flash on a zap trunk.  This is only a hack for
people who want to perform transfers and such via AGI and is generally
quite useless otherwise.  Returns 0 on success or -1 if this is not
a zap trunk

I've never bothered with Flash myself. I'd setup an account with an
ITSP like VoipJet or Sixtel. Then you can initial the calls from
in-house and make multiple outgoing connections, transfering each into
meetme. Then again, if you used a sip phone (as opposed to an ata or
TDM) you could conference one the phone withour resorting to meetme.

Michael

--
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Re: [Asterisk-Users] CallerID in Australia Analogue PSTN Phone

[Howard Lowndes]

On Wed, 2005-01-05 at 23:22, PHP Mechanic wrote:
   Is there anyone using * in AU that has successfully extracted the CLID
   from an incoming analogue PSTN phone call, and would like to spread 
   the
   word?
 
  What I need more though is examples of anything that needs to go into
  extensions.conf
 
 You could add this line if you want
 exten = s,1,NoOp(Caller ID on the PSTN line is ${CALLERID}) 

M.  Tried that, but it didn't deliver ${CALLERID}

 
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[Asterisk-Users] Usage Of Additional LEDs For Snom (was; Status of SNOM Intercom)

[Peer Oliver Schmidt]

Nils Ohlmeier wrote:
 BTW the new snom model (no availability dates yet) will have more
 LED's then  the existing ones. What are the most important or
 interesting features (existing or not implemented yet) for the
 Asterisk community for this programmable keys with LED's?
The only missing feature would be an integration into the Queue 
management,ie. my customers use the callback functionality of the queue. 
It would be nice to show the status of this on the phone,ie. if one of 
the extensions currently mapped to my phone is part of a callback in a 
queue.

Other than that I am a happy camper, once intercom is working
--
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Peer Oliver Schmidt
the internet company
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Re: [Asterisk-Users] CallerID in Australia Analogue PSTN Phone

[PHP Mechanic]

 What I need more though is examples of anything that needs to go into
 extensions.conf
You could add this line if you want
exten = s,1,NoOp(Caller ID on the PSTN line is ${CALLERID}) 
M.  Tried that, but it didn't deliver ${CALLERID}
Did the caller have callerid enabled by their telco ?
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RE: [Asterisk-Users] Digium T100P T1 Card

[Wiley Siler]

Apologies if the format of the email was troublesome.  I am accessing my
email remotely via Outlook Web Access otherwise the format would have
been plain text.

So, I need to learn more about voice T1s?  Reeally?  That would be why I
am posting to the user group in the first place.  To learn more.  The
wiki says nothing about how PRI works because it is expected that
someone will know.  Well, I didn't.  Had to ask.  After cruising ebay
for 30 minutes looking at routers and reading the tech spec on the
T100P, I figured out the very same thing regarding the fact that no
router was needed.

Using my analog line for fax is not a matter of needs.  It is a matter
of using available lines that we will have for another 18 months because
that T1 is under a long contract.  The ISP company wants an arm and a
leg to upgrade the T1 from analog lines to digital so that is why I am
getting a separate voice T1 altogether.  That will leave these analog
lines unused so I may as well dedicate them to my fax system and keep
all the digitals for our voice.

Finally, let me say thank you.  Your info is exactly what I needed and I
truly appreciate it.  People who take time to help others should truly
be applauded.  I have seen scores of replies from you to others so I
know you are one of the best contributors here.  In fact, I usually read
yours first just because of the quality of your replies.

However, was there that much need for the criticism and arrogance in
your reply?  Wouldn't it just be esier not to reply at all than start
off with a complaint about my HTML formatting, go to a critique of how I
formatted my 4 sentence email (paragraph for 4 sentences?), and finish
up by pointing out that I don't know much about voice T1s?

Regards,
Wiley


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, January 05, 2005 1:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Digium T100P T1 Card

On Wed, 2005-01-05 at 01:01 -0700, Wiley Siler wrote:
 Hello All,
  
 I could use a recommendation if anyone has a moment. 

It is preferable to not use HTML in email. Just because a font size
looks good on your monitor doesn't mean it is anywhere close to good
anywhere else. Your choosen font size ends up being 1/2 to maybe 2/3 the
size of standard text. In other words, had it not been for the slightly
interesting to me subject line, this would have been ignored as a user
to clueless to bother with.

Also learn about why paragraphs are good.

  I have the T100P but I have not gotten my service yet.  I want to 
 have at least 12 lines of digital voice with DID.  Should I just seek 
 out a PRI ISDN provider or is there something else I should look for?
 I want to keep cost as low as possible.  Also, I want to own my own 
 router for the phones since it is always a hassle to get anything 
 fixed from the tele-company.

This shows you need to learn quite a bit more about phone service. There
is no need for a router on a telephony T1. You will either want to plug
a channelized T1 or PRI into the T100P directly. 

If you want some analog FXS ports, you could also go the route of an
ADIT 600 and plug the T1 into the ADIT and route your incoming 12
channels to the second port of the ADIT and then plug it into the T100P.
The benefit here is you will have 12 channels left over to signal back
from the T100P to the ADIT and have those channels routed to FXS ports.
I used to do something similar to that with a Zhone channel bank before
our company fully trusted asterisk.

   What is a good and cheap router (Cisco maybe) that I would use to 
 interface to the T100P?  I plan to integrate my system to use our old 
 analog lines for fax so I will have questions on that later too.

You don't need analog lines for FAX. Follow the directions above for the
ADIT and you will be able to have analog ports to plug your fax machines
and route them out the T1. 
--
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] configuring sample time period for codecs?

[Roy Sigurd Karlsbakk]

how can I configure the sample time period (10ms,20ms etc) for codecs?
kapejod told me on IRC that this could not be achived with 
configuration, and that I needed to dig into the source to do this.

Can someone please tell me what asterisk normally uses here?
Should the client setting always match asterisk's sample period, or 
will this be handled by the codec(s), if so, which?

thanks
roy
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Re: [Asterisk-Users] chan_cornet

[richard Coco]

What do you mean with *But doesnt Digium have Asterisk-HiPath solutions?*. If you are meaning a connection with Digium cards so...sorry i've never usedDigium cards. Buti thought (see your first thread *connect Asterisk with Siemens HiPath HG1500*), you are looking for a way to connectAsterisk to the HG.If you use HG1500 you have to configure a h323 channel (h.323 or oh.323). If not, you can try toconfigurechan_capi and try to connect Asterisk (e.g with an EICONDiva card)to a STLS4 (for HiPath3500) or a STMD8(forHiPath3700).
hope it will help...
if in a few days you have additional informations about chan_cornet, please let me (the list) know.

thx
Joao Pereira [EMAIL PROTECTED] wrote:




Hi
I dont knowif Steffen's chan_cornet is working. I emailed him, but with no result.
Yesterday I read this article
http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom

It has some solutions... but not yet a direct Asterisk-HiPath connection.

But doesnt Digium have Asterisk-HiPath solutions?

Joao

- Original Message - 
From: richard Coco 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Sent: Wednesday, January 05, 2005 12:13 PM
Subject: Re: [Asterisk-Users] chan_cornet 

Hi,

The HG1500 is a HiPath3000 board and i don't have experience with Asterisk and HiPath3K.
What we have is an Asterisk connected to a Siemens HiPath4000 over a H.323 trunk using oh323 and the HG3550 board. It works fine. But the Siemens HG3550 only supports H.323 V2.0 (so not a lot of features are available). May be Steffen's chan_cornet will change this.
Are there any news about this project?Joao Pereira [EMAIL PROTECTED] wrote:
Hibut did anyone have ever used a Siemens HiPath PBX with Asterisk?If you made it, please tell me how...I read that chan_cornet does exist...http://lists.digium.com/pipermail/asterisk-users/2004-October/069559.htmlIs there any Digium Hardware solution for the Asterisk HiPath connection?ThanksJoao Pereira- Original Message -From: "Luís Palma" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"Sent: Tuesday, January 04, 2005 10:30 PMSubject: Re: [Asterisk-Users] connect Asterisk with Siemens HiPath HG1500 Hi, It doesn't tell you much but it looks like that you are not alone when trying to integrate with Siemens Hicom. It seems someone has decided to make it by himself.
 http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom Regards Luis Palma On Tue, 4 Jan 2005 12:26:38 -, Joao Pereira <[EMAIL PROTECTED]>wrote:  Hi  I want to know the best way to connect Asterisk to a Siemens HiPathHG1500  PBX. Until now I came out with 3 solutions:   1-Asterisk being a H.323 client of the Siemens PBX (I believe it needs  Siemens licences and Digium hardware)  2-Asterisk connecting to the PSTN phones with Voice Modems (goodideia!!!  but its analog... doesnt have caller information...)  3-Using RDIS interfaces to connect the Siemens PBX   does someone have other ideias?   Thanks  Joao Pereira   ___  Asterisk-Users mailing list 
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[Asterisk-Users] Versions of * what do they do/where is the change history/docs?

[John Middleton]

Could you please explain or tell me where it is explained the version
and contents of * that is retrieved with CVS.

I am wondering whether there is a change list or something. If you
tell me here I will update the Wiki ;-)

Thanks

John
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[Asterisk-Users] TDM400P + Asterisk + zaptel timer ?

[Samudra E. Haque]

Hello, I thought that my Digium TDM400P would be the right hardware to
support the zaptel timer, and put the following IAX.CONF entry to test,
(trunk=yes) in the example below

[VHAX]
type=peer
auth=md5
username=whoknows
jitterbuffer=yes
;trunk=yes
secret=terriblesecret
host=4.5.6.7
qualify=1200
disallow=all
allow=ulaw
allow=gsm
;allow=g711u
;allow=g711a

But, it didn't work. So I had to comment it out. When trunk=yes was set, the
following result happened, and literally the audio was very unstable.
Otherwise the system has been working great. I am using ax1*CLI show
version
Asterisk CVS-HEAD-12/02/04-17:57:31 built by [EMAIL PROTECTED] on a i686 running
Linux


Jan  5 13:40:53 WARNING[1928]: Unable to open pseudo channel for timing...
Sound may be choppy.
Jan  5 13:40:54 WARNING[1928]: Unable to open IAX timing interface: No such
file or directory
Jan  5 13:40:57 NOTICE[1928]: Request to schedule in the past?!?!
Jan  5 13:41:00 WARNING[1928]: Unable to support trunking on peer 'VHAX'
without zaptel timing
Jan  5 13:41:06 WARNING[1928]: Unable to specify channel 1: No such device
or address
Jan  5 13:41:06 ERROR[1928]: Unable to open channel 1: No such device or
address
Jan  5 13:41:06 ERROR[1928]: Unable to register channel '1'
Jan  5 13:41:06 WARNING[1928]: chan_zap.so: load_module failed, returning -1
Jan  5 13:41:06 WARNING[1928]: Loading module chan_zap.so failed!

However, in my /dev/zap directory from original installation: (Fedora Core
3)
[EMAIL PROTECTED] asterisk]# ls -l /dev/zap
total 0
crw---  1 root root 196,   1 Jan  5 15:36 1
crw---  1 root root 196,   2 Jan  5 15:36 2
crw---  1 root root 196,   3 Jan  5 15:36 3
crw---  1 root root 196,   4 Jan  5 15:36 4
crw---  1 root root 196, 254 Jan  5 15:36 channel
crw---  1 root root 196,   0 Jan  5 15:36 ctl
crw---  1 root root 196, 255 Jan  5 15:36 pseudo
crw---  1 root root 196, 253 Jan  5 15:36 timer
[EMAIL PROTECTED] asterisk]#

so, I'm puzzled, what / where is the zaptel timer supposed to be defined ?
the TDM400P has FXO and FXS interfaces and is a Digium.

-samudra



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Re: [Asterisk-Users] chan_cornet

[richard Coco]

Hello Steffen,

hey it sounds very good...!!! but what do you mean with *new hipath version doesn't support H.323 anymore*? What version are you talking about? As far as i know the new version of HiPath4000 V2.0 still supports H.323 (STMI2).Steffen Koepf [EMAIL PROTECTED] wrote:
Hello, I dont know if Steffen's chan_cornet is working. I emailed him, but with no result.You are not patient enough ;)You got an answer one minute ago ;)No it is not ready, it is work in progress.At the moment i'm forced to get a Asterisk-SMS Gateway working here,with our old PBX, but i hope it works soon so that i can proceed withchan_cornet. At the moment, Optipoint 400 and 600s can register tothe chan_cornet, and one can call them so that they ring. There is somelittle work to be done, to get the voice working (a bit H.323 stuff),and with a little editor (for entering the numbers, the phone can't handle this), the Phone2PBX part should work. And then the next goalis the PBX2PBX stuff.That newer HiPath PBXs are worse, coz Siemens dropped the H.323-Support,that means they do not support one stand
 ard voip
 protocol. They saidthat they will support SIP in the future, but they say this for morethan a year now. That means, one can connect now that PBXs with TDM-Lines(S2M, BRI) or to another cornet-ip supporting device like IPDAs (smallersiemens pbxs that connect to a main PBX) or PBXs and hopefully, chan_cornetsometime ;). cu,Steffen___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] TDM400P + Asterisk + zaptel timer ?

[Adam Goryachev]

On Wed, 2005-01-05 at 19:52 +0600, Samudra E. Haque wrote:
 Hello, I thought that my Digium TDM400P would be the right hardware to
 support the zaptel timer, and put the following IAX.CONF entry to test,
 (trunk=yes) in the example below
 
[snip]
 But, it didn't work. So I had to comment it out. When trunk=yes was set, the
 following result happened, and literally the audio was very unstable.
 Otherwise the system has been working great. I am using ax1*CLI show
 version
 Asterisk CVS-HEAD-12/02/04-17:57:31 built by [EMAIL PROTECTED] on a i686 
 running
 Linux
 
 Jan  5 13:41:06 WARNING[1928]: chan_zap.so: load_module failed, returning -1
 Jan  5 13:41:06 WARNING[1928]: Loading module chan_zap.so failed!

You should first solve this problem. If chan_zap.so can't load, then you
don't have a timing device.

Do this:
Get zaptel source/compile/install
in /usr/src/asterisk make clean;make install
then modprobe zaptel and modprobe wcfxs (or wctdm if using current CVS)
then ztcfg
If you get *NO* error messages, then continue, else report back to the
list your /etc/zaptel.conf and details of your TDM card modules, and the
errors
Start asterisk
If you can do a zap show channels and see your channels, then continue,
else report back to the list the output of ztcfg -vv, and
your /etc/asterisk/zapata.conf and the error messages when starting
asterisk.

If you have done all the above correctly, then it will probably work.
Remember that you need to enable trunk'ing at BOTH ends as well.

Hope this helps you...

Regards,
Adam
-- 
 -- 
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Website Managers
Ph:  +61 2 9345 4395[EMAIL PROTECTED]
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Re: [Asterisk-Users] Polycom Buddy Feature

[Jon Radon]

I think you need to subscribe to the context where exten 200 exists. 
I'm not sure if it'll work with an arbitrary context.  You may also
want to try sending the hint to just one phone.  I'm not 100% on the
format for sending the hint to multiple phones.


On Tue, 04 Jan 2005 21:40:00 -0700, Nihal [EMAIL PROTECTED] wrote:
 I'm still trying to work this out.
 
 I've got this in my sip.conf
 [1003polycom]
 type=peer
 secret=abc123
 host=dynamic
 defaultip=192.168.1.215
 context=default
 mailbox=1003
 subscribecontext=phonestatus
 
 [1004polycom]
 type=peer
 secret=abc123
 host=dynamic
 defaultip=192.168.1.214
 context=default
 mailbox=1004
 subscribecontext=phonestatus
 
 And this in my extensions.conf
 [phonestatus]
 exten = 200,hint,SIP/1003polycom
 exten = 200,hint,SIP/1004polycom
 
 Then I added a contact to my phone of 1004, speed dial 1004.
 It shows the phone in my buddies list, but the status doesnt update.
 
 Did I miss something or am I doing something wrong?
 
 Thanks!
 Nihal
 
 On Tue Jan  4 17:13 , Matt Gibson  sent:
 
 Hi Jared,
 
 
 Jared Armstrong wrote:
  Matt,
  Can you explain how you were able to get this functionality to work? I
  would like also possibly an example configuration file if you could
  manage it.
 
 I followed the instructions that Jon put on the list a few weeks ago:
 
 in the phone's sip.conf entry add
 subscribecontext=context_name
 
 and in extensions.conf add a hint (info can be found near the bottom
 of the following page)
 http://www.voip-info.org/wiki-Asterisk+standard+extensions
 
 Hope this helps!
 
 Matt
 
 
 --
 Matt Gibson
 VOIP Administrator
 NJ Tech Solutions
 1.314.480.4550 ex. 6400
 1.877.999.4678 ex. 6400
 
 
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-- 
Is it something someone said, was it something someone said?
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[Asterisk-Users] Asterisk as Nortel option 11 Autoattendant, question

[Voip Business]

Hello List

I have a customer with an Option 11 with a E1 Card (MFCR2) to a
Audiocodes Mediant 2000 gateway and a Sip netowrk.

at this moment I have working an Asterisk Box as Voicemail for  all
the PBX and working like a Charm. but my customer wants to have an
autoattendant IVR  for the option 11 within the Asterisk.

for me is clear that if I use 2 channels of the Audiocodes (Inbound -
outbound) I can be able to do that, but is there any way to configure
* in order to trasnfer  the call to the desired extension and then
frees the E1 trunk?

if someone has this answer I will need Consulting Services for this.

thanks

Humberto
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[Asterisk-Users] Problems with msn's, did not find device for msn

[Sebastian Buntin]

Hello everybody!

happy new year ;-)

I have a problem with inbound calls on my Diva Server PRI

This is my problem:

2005-01-05 15:38:04 ERROR[31716]: chan_capi.c:1696 pipe_msg: did not find
device for msn = 4132


capi.conf:

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de

[interfaces]
msn=4132
incomingmsn=*
controller=1
context=demo
mode=immediate
isdnmode=ptp
devices=30

extensions.conf:

[demo]
exten = 4132,6,Dial(SIP/test)




so, I'm a bit confused now.
what can I do???


thanks for helping me out!

greetings, Sebastian


Capi Debug output:

CAPI Debugging Enabled
-- CONNECT_IND ID=001 #0x0009 LEN=0050
  Controller/PLCI/NCCI= 0x201
  CIPValue= 0x10
  CalledPartyNumber   = 814132
  CallingPartyNumber  = 21 8003621423132
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = 80 90 a3
  LLC = default
  HLC = 91 81
  AdditionalInfo
   BChannelinformation= default
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default

2005-01-05 15:41:32 NOTICE[31716]: chan_capi.c:1932 capi_handle_msg:
CONNECT_IND ID=001 #0x0009 LEN=0050
  Controller/PLCI/NCCI= 0x201
  CIPValue= 0x10
  CalledPartyNumber   = 814132
  CallingPartyNumber  = 21 8001234567891233
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = 80 90 a3
  LLC = default
  HLC = 91 81
  AdditionalInfo
   BChannelinformation= default
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default

  == CONNECT_IND
(PLCI=0x201,DID=4132,CID=03621423132,CIP=0x10,CONTROLLER=0x1)
-- INFO_IND ID=001 #0x000a LEN=0020
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x70
  InfoElement = 814132

2005-01-05 15:41:32 ERROR[31716]: chan_capi.c:1696 pipe_msg: did not find
device for msn = 4132
-- INFO_IND ID=001 #0x000b LEN=0018
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x18
  InfoElement = a9 83 8a

-- INFO_IND ID=001 #0x000c LEN=0015
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x8005
  InfoElement = default

-- INFO_IND ID=001 #0x000d LEN=0017
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x8
  InfoElement = 80 95

-- INFO_IND ID=001 #0x000e LEN=0015
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x805a
  InfoElement = default

-- DISCONNECT_IND ID=001 #0x0010 LEN=0014
  Controller/PLCI/NCCI= 0x201
  Reason  = 0x3495

  == DISCONNECT_IND PLCI=0x201 REASON=0x3495
activehangingup
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Re: [Asterisk-Users] chan_cornet

[Steffen Koepf]

Hello,

 but what do you mean with *new hipath version doesn't 
 support H.323 anymore*? What version are you talking about? 
 As far as i know the new version of HiPath4000 V2.0 still 
 supports H.323 (STMI2).

HiPath 3000 - H.323 Support
HiPath 4000 - NO H.323 Support and nothing else but cornet (voip).
   cornet-ip is basic h.323 + addons, but it does not
   work with standard h.323 devices.


cu,

Steffen

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[Asterisk-Users] ISDN/SS7 book?

[Roy Sigurd Karlsbakk]

hi
some time ago, I asked the list of a good book for learning ISDN and 
SS7. I don't need to know how to write a channel driver or something; I 
just want to know more about the possibilities and what's really sent 
back and forth. I was told the book ISDN and SS7: Architectures for 
Digital Signaling Networks by Uyless Black (ISBN 0132591936) was a 
good choice, but this seems sold out. Does anyone know about another 
book about the subject?

thanks
roy
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Re: [Asterisk-Users] Cannot Hear at all

[Giovanni Powell]

The asterisk server doesn't need a sound card. The machines that have
the xlite softphone need to have a sound card.

E.g. you have 2 pc's running xlite + the asterisk server. The asterisk
server doesn't need a sound card but the 2 pc's will. (sounblaster
live solved similar problems i was having)


On Wed,  5 Jan 2005 09:33:56 -0200, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
 
 Hi all,
 
 I am attempting to call from softphone to softphone, I am using X-lite to call
 another X-lite.
 
 I get the phones to call each other and finnaly connecting, but cannot hear 
 the
 voice at all. Is there any ideas as to why this is happening.
 
 (I don't have sound card in my linux server. I need one in my linux server ??)
 
 PS: callonhold is working but cannot hear the music too.
 
 look at my messages file:
 
 Dec 21 22:31:36 NOTICE[65540]: Request to schedule in the past?!?!
 Dec 21 22:31:36 WARNING[16384]: Unable to get our IP address, Skinny disabled
 Dec 21 22:31:36 NOTICE[65540]: Request to schedule in the past?!?!
 Dec 21 22:31:37 NOTICE[65540]: Request to schedule in the past?!?!
 Dec 21 22:31:37 WARNING[16384]: Unable to open /dev/dsp: No such device   
 
 Dec 21 22:31:37 NOTICE[65540]: Request to schedule in the past?!?!
 Dec 21 22:32:46 NOTICE[213005]: RFC3389 support incomplete.  Turn off on 
 client
 if possible
 Dec 21 22:34:42 WARNING[16384]: Unable to get our IP address, Skinny disabled
 Dec 21 22:34:42 NOTICE[65540]: Request to schedule in the past?!?!
 Dec 21 22:34:42 NOTICE[65540]: Request to schedule in the past?!?!
 Dec 21 22:34:42 WARNING[16384]: Unable to open /dev/dsp: No such device
 Dec 21 22:34:43 NOTICE[65540]: Request to schedule in the past?!?!
 Dec 21 22:34:43 NOTICE[65540]: Request to schedule in the past?!?!
 Dec 21 22:35:51 NOTICE[213005]: RFC3389 support incomplete.  Turn off on 
 client
 
 Thanks,
 
 Eduardo Nunes
 
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Re: [Asterisk-Users] ISDN/SS7 book?

[Roger Schreiter]

Roy Sigurd Karlsbakk schrieb:
...
Digital Signaling Networks by Uyless Black (ISBN 0132591936) was a good 
choice, but this seems sold out. Does anyone know about another book 

Hi,
since it's sold out for a longer time, I sold mine used at
amazon.
Roger.
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[Asterisk-Users] Happy Wednesday Morning SMS question, slightly OT

[Jay Milk]

We all mostly know that * as well as various SIP phones support SMS.
While the final setup is somewhat of a mystery, there are reports of
those lucky souls who have it working.  We also know that in order to
send an SMS to a mobile phone, we need to connect to some SMS message
center and get the word out that way.  

Now, here's the new (?) element:  How can I *accept* messages on my
voip-based US landline?  I know that if I send an SMS from my T-Mobile
phone to a friend's Verizon phone, the message goes through, so
somewhere there must exist a national message center that knows which
carrier to hand the message off to.  Technically it should be possible
to register a phone number with them to receive messages sent from
cell-phones or from other * systems, and then to receive these messages
through * and onto a SMS capable IP phone...?

Who knows more about this?

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[Asterisk-Users] Re: Speex codec problem (unresolved ?) = Fixed

[Walter Klomp]

After looking at the source I had also tried to increase the buffer size
from 8000 to 16000, but that made other codecs (like lin_to_g729) choke, 
and
I still had the problem...

I like speex and would like to use it (as I find ilbc a bit too scratchy)
I am running Asterisk CVS-HEAD-11/16/04-17:19:53 and speex-1.0.4 
libraries
on Gentoo Linux.
The best sugestion that I can offer is that I saw the same problem and
could not resolve it but after upgrading * to CVS after the 12/10 it
went away. Never did find a solution and gave up looking as it solved it.
It also fixed some SIP issues I had and they went away aswell.
Sorry that might not be the answer you are looking for but thats what
worked for me.
David
You're right... after wiping my asterisk directory and doing a new CVS 
checkout, installing Speex 1.1.6, it now works... I do wonder what has 
changed though as on no list there is a mention of a fix (other than yours 
David)

Walter. 

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[Asterisk-Users] Asterisk Pbx Manager Equivalent

[Paul Brock]

http://www.thirdlane.com/screenshots.htm
(Asterisk PBX Manager from Thirdlane) looks like a 

great program for eye candy configuration of
Asterisk.



However it costs lost of $, and Im currently only an experimenter
so to speak.



Anyone advice of a decent alternative that is similar??
Currently, we only have VOIP connections, 

but will have a couple of Digium fxs/fxos soon to
have a play with, so would be advantageous if it 

worked with these too



Finally, Anyone know of a Digium hardware Reseller in the Uk at all?? 



Thx



Paul 






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Re: [Asterisk-Users] Status of SNOM Intercom

[Raymond McKay]

Using the new firmware is there still the issue with needing to patch 
chan_sip.c, or does it work out of the box? Do you have details on how it 
should be implemented within *?

As of now, the hack still applies.  It would be wonderful though if somebody 
could implement a command line variable that allows you to append anything 
to the SIP URI in the form of variable=variable.  Right now the patch 
essentially breaks the VXML_URL functionality right now as stands.

Raymond McKay
President
RAYNET Technologies LLC
http://www.raynettech.com
(860) 693-2226 x 31
Toll Free (877) 693-2226 

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[Asterisk-Users] Asterisk Pbx Manager Equivalent (in plain text - apologies to those that dont like HTML mail!!)

[Paul Brock]

http://www.thirdlane.com/screenshots.htm (Asterisk PBX Manager from
Thirdlane) looks like a great program for eye candy configuration of
Asterisk.

However it costs lost of $, and I'm currently only an experimenter so to
speak.

Anyone advice of a decent alternative that is similar?? Currently, we only
have VOIP connections, but will have a couple of Digium fxs/fxo's soon to
have a play with, so would be advantageous if it worked with these too.

Finally, Anyone know of a Digium hardware Reseller in the Uk at all?? 

Thx

Paul

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Re: [Asterisk-Users] Happy Wednesday Morning SMS question, slightly OT

[Michael Welter]

Jay Milk wrote:
We all mostly know that * as well as various SIP phones support SMS.
While the final setup is somewhat of a mystery, there are reports of
those lucky souls who have it working.  We also know that in order to
send an SMS to a mobile phone, we need to connect to some SMS message
center and get the word out that way.  

Now, here's the new (?) element:  How can I *accept* messages on my
voip-based US landline?  I know that if I send an SMS from my T-Mobile
phone to a friend's Verizon phone, the message goes through, so
somewhere there must exist a national message center that knows which
carrier to hand the message off to.  Technically it should be possible
to register a phone number with them to receive messages sent from
cell-phones or from other * systems, and then to receive these messages
through * and onto a SMS capable IP phone...?
Who knows more about this?
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Here's one solution if you have a EUR1600 to spend:
http://www.2n.cz/products/gsm_gateways/voip.html
For outbound SMS, I've thought about having an * plug-in open an HTTP 
connection with the cellular provider's web site.  From there I could 
send the message (kinda like the old 3270 screen scrape).

Is SMS part of SS7, or is it a cellular protocol only?  I've seen the 
SMS functions in Asterisk--how are these intended to be used?  Is it 
Europe/GSM only?

Cheers
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
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[Asterisk-Users] Do Not Disturb

[Shawn Dillon]

We have just finished installing some Sayson 480i phones ( 
will post a review soon) and I have one issue. I cannot seem to get the *78 , 
*79 ( and like) functions to work. Are these automatically installed with 
Asterisk?

Anything required in the extensions.conf or 
sip.conf?

When I dial *78 I get a 404 error on the phone ( 
Call failed). Nothing shows in the Asterisk console.

Thanks in advance,

Shawn Dillon
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[Asterisk-Users] asterisk - oh323 driver

[João Amaro]

Hi List


I'm having problems starting asterisk  with asterisk-oh323-0.6.4.
I'm using this versions:
 asterisk-1.0.3
 asterisk-oh323-0.6.4
 openh323-Janus_patch4 + asterisk-0h323 patch
 pwlib-Janus_patch4

At starting time, i've this error message

# /srv/usr/sbin/asterisk -vvvc
[chan_oh323.so]
Jan  3 17:06:26 WARNING[5817]: loader.c:258 ast_load_resource: /srv/usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: _ZNK13PSoundChannel6IsOpenEv

Jan  3 17:06:26 WARNING[5817]: loader.c:440 load_modules: Loading module chan_oh323.so failed!


I've tried to upgrade to version 0.6.5, but i got a compile error. 
Anyone know how to solve this error ?

Thanks in advance, and have a GOOD 2005

Regardz,

Joo Amaro



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[Asterisk-Users] One way audio [Asterisk + Innovaphone IP3000 + asterisk-oh323/h323]

[Silviu Herchi]

Hello everybody,

Ive been trying to solve a problem for several weeks now but it really
beats me.

There are several hard phones connected to an Innovaphone 3000 VoIP gateway.
On the other side I have a SIP softphone connected to Asterisk. The problem
I have is that on incoming calls (hardphones to softphone) I only have
outgoing audio (from soft to hardphone); everything is OK when I call the
other way round.

Asterisk (I'm using 1.0.3) uses H323 to access to the Innovaphone 3000. I
tried both asterisk-oh323 0.6.5 and the supplied h323 channel with the same
results. All my machines are on the same LAN connected by a hub (there is no
NAT or firewall involved at all).

Here is the setup:

- Innovaphone 3000 (10.253.30.254)
- Test1 3760 hardphone (10.253.10.102)
- GnuGK gatekeeper 2.2.0 (10.253.30.11), compiled with the required versions
of pwlib and openh323
- Asterisk 1.0.3 with either asterisk-oh323 0.6.5 or h323 channels
(10.253.30.1)
- SJphone SIP softphone on windows (10.253.30.10)

As both asterisk-oh323 and h323 behave the same way, I'm wondering whether
this could be an OpenH323 problem.

I tried an Ethereal trace, and there is no RTP whatsoever from Asterisk to
the hardphone (the only RTP streams are Asterisk -- softphone (both ways)
and hardphone -- Asterisk (one-way!!)).

The one strange thing I noticed when I enabled debug on asterisk-h323 is
that at some point when the outgoing logical channel is open the remote ip
address is 127.0.0.1 (have a look at the attached log).

Needless to say, I googled my a** off for the few last weeks to no avail...

Thank you for your help!

Best regards,

Silviu


*CLI   == New H.323 Connection created.
-- Received SETUP message
-- Setting up Call
-- Call token:  [ip$10.253.30.11:1119/284]
-- Calling party name:  [Test1]
-- Calling party number:  [3760]
-- Called  party name:  [377]
-- Called  party number:  [377]
=-= In OnAnswerCall for call 284
-- Executing StripMSD(H323/ip$10.253.30.11:1119/284, 3) in new stack
-- Executing Goto(H323/ip$10.253.30.11:1119/284, SIP||1) in new
stack
-- Goto (SIP,,1)
-- Executing Dial(H323/ip$10.253.30.11:1119/284, SIP/silviu.herchi)
in new stack
-- Called silviu.herchi
-- SIP/silviu.herchi-b027 is ringing
Sending alerting
-- Received Facility message... 
-- Received Facility message... 
-- Received Facility message... 
-- Received Facility message... 
-- Received Facility message... 
=*= In CreateRealTimeLogicalChannel for call 284
-- externalIpAddress: 10.253.30.1
-- externalPort: 16246
-- SessionID: 1
-- Direction: IsReceiver
 -- Started logical channel: receiving G.711-ALaw-64k{sw}
-- channelsOpen = 1
RTP channel id 1 parameters:
-- remoteIpAddress: 10.253.30.102
-- remotePort: 16722
-- ExternalIpAddress: 10.253.30.1
-- ExternalPort: 16246
-- SIP/silviu.herchi-b027 answered H323/ip$10.253.30.11:1119/284
answering call
=*= In CreateRealTimeLogicalChannel for call 284
-- externalIpAddress: 10.253.30.1
-- externalPort: 16246
-- SessionID: 1
-- Direction: IsTransmitter
 -- Started logical channel: sending G.711-ALaw-64k{sw}
-- channelsOpen = 2
RTP channel id 1 parameters:
-- remoteIpAddress: 127.0.0.1   #WHY 127.0.0.1 ???
-- remotePort: 2070
-- ExternalIpAddress: 10.253.30.1
-- ExternalPort: 16246
=-= In OnConnectionEstablished for call 284
-- Connection Established with Test1 (3760) [10.253.30.11]
-- Received Facility message... 
=-= In OnReceivedAckPDU for call 284
  == Spawn extension (SIP, , 1) exited non-zero on
'H323/ip$10.253.30.11:1119/284'
-- ClearCall: Request to clear call with token
ip$10.253.30.11:1119/284
-- Sending RELEASE COMPLETE
channelsOpen = 1
channelsOpen = 0
 -- Call with Test1 (3760) [10.253.30.11] completed (EndedByLocalUser)
== H.323 Connection deleted.

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.6.8 - Release Date: 03/01/2005
 

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Re: [Asterisk-Users] ISDN/SS7 book?

[Dorn Hetzel]

On Wed, Jan 05, 2005 at 03:56:39PM +0100, Roy Sigurd Karlsbakk wrote:
 hi
 
 some time ago, I asked the list of a good book for learning ISDN and 
 SS7. I don't need to know how to write a channel driver or something; I 
 just want to know more about the possibilities and what's really sent 
 back and forth. I was told the book ISDN and SS7: Architectures for 
 Digital Signaling Networks by Uyless Black (ISBN 0132591936) was a 
 good choice, but this seems sold out. Does anyone know about another 
 book about the subject?

amazon.com links to plenty of used copies this book for under $15.00.

I just put ISDN and SS7 in the search box and it went straight to it.

-Dorn
 
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Re: [Asterisk-Users] manager API

[Christopher L. Wade]

Guys,
After connecting to the * manager, each and every event is sent to the
connected client, right?
This means that if I install a client on each PC for monitoring incoming
calls, or pretty much anything else, it will create a lot of excess
traffic on my LAN.
Can I connect to the manager and tell it to send only events regarding
specific extension(s) my way?
I'd like to provide a popup display say incoming call from CALLERID
and provide a way for the callee to divert it to voicemail or something
like that.
Thanks
Another option would be to write a manager proxy (look on the wiki - 
several simple examples already exist) that only sends particular events 
on to clients who request them.  Basically have a script connect to * 
one time.  Clients then connect to this script instead of * and tell the 
script what messages they want to know about.  At this point the script 
handles the end clients - not * - a double benefit (less net traffic and 
less work for *).

-Chris
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RE: [Asterisk-Users] QOS / Cisco / Asterisk

[Matt Schulte]

Title: Message



Yes 
yes, we've been through all that actually :-) We did find out it was one of the 
3550's reseting the TOS.

  
  -Original Message-From: 
  [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 
  Tuesday, January 04, 2005 2:40 PMTo: 
  asterisk-users@lists.digium.comSubject: RE: [Asterisk-Users] QOS / 
  Cisco / Asterisksnip  What's 
  wrong with doing it by port? We're actually using SIP to terminate 
  calls, going by rtp.conf the portscould range several thousand ports. What 
  we're going for is onlyhonoring TOS for that particular customer, luckily 
  these are T1customers hosted on our routers. They understand that their 
  firewallscannot pass TOS, if they do (ie: we packet sniff and see this) 
  thenthey're on their own.In a nutshell we wanted to avoid using 
  hardcoded ports, what if say agame server was in that port range (and used 
  udp lol), you would berather screwed. /snip Ahh OK. Well, 
  how about configuring a laptop with ethereal (http://www.ethereal.com/) and 
  capturing the packets you have in mind? It even runs on Windows. :p It's 
  pretty easy to specify a particular destination or so, for limiting which 
  traffic you sniff. You could use an old hub and start plugging the laptop in 
  between routers using the hub so it can capture the packets. Should be fairly 
  quick to isolate which router is modifying the TOS value. Just an idea... of 
  course you have to have physical access to the network... 
  HTH, -Ron 
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Re: [Asterisk-Users] Versions of * what do they do/where is the change history/docs?

[Steven Critchfield]

On Wed, 2005-01-05 at 13:51 +, John Middleton wrote:
 Could you please explain or tell me where it is explained the version
 and contents of * that is retrieved with CVS.

CVS is sometimes a pain in the but, but it is possible to grab that
information via the log command in CVS.

 I am wondering whether there is a change list or something. If you
 tell me here I will update the Wiki ;-)

As someone else pointed out there is a -cvs mailing list to see changes
as they are committed and give you a bit of a heads up.
-- 
Steven Critchfield [EMAIL PROTECTED]

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RE: [Asterisk-Users] Do Not Disturb

[Senad Jordanovic]

When I dial *78 I get a 404 error on the phone ( Call failed). Nothing
shows in the Asterisk console.


You need to check dial plan in the 480i...


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Re: [Asterisk-Users] Happy Wednesday Morning SMS question, slightly OT

[Rich Adamson]

 We all mostly know that * as well as various SIP phones support SMS.
 While the final setup is somewhat of a mystery, there are reports of
 those lucky souls who have it working.  We also know that in order to
 send an SMS to a mobile phone, we need to connect to some SMS message
 center and get the word out that way.  
 
 Now, here's the new (?) element:  How can I *accept* messages on my
 voip-based US landline?  I know that if I send an SMS from my T-Mobile
 phone to a friend's Verizon phone, the message goes through, so
 somewhere there must exist a national message center that knows which
 carrier to hand the message off to.  Technically it should be possible
 to register a phone number with them to receive messages sent from
 cell-phones or from other * systems, and then to receive these messages
 through * and onto a SMS capable IP phone...?
 
 Who knows more about this?

Based on previous postings, the SMS thingie is primarly a european thing
and is rather different from the US cellular implementation. Since you
mentioned T-Mobile, I'm assuming you're in the US.

If that assumption is correct, then its not likely you're going to be
able to accomplish your objective without implementing some sort of
site-specific role-your-own mechanism (eg, I don't know of any US cellular
company that would sell you a sms address for your pbx).


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[Asterisk-Users] Last callers script?

[Mike Dent]

Hi,
Is there some script which can be called from a * extension to
playback the recent incoming
callers on a particular PSTN line?

In the UK 1471 is a BT number which plays back the most recent callers
number, it also
gives you the option to call this number back (now charging you for
this service too!).

Is there anything similar in asterisk-land?
thanks
Mike
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[Asterisk-Users] does TE405P support 3Bit CAS?

[Paradise Dove]

does TE405P support 3Bit CAS?
what are the configuration tips?

thanx,
Paradise Dove
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RE: [Asterisk-Users] asterisk - oh323 driver

[Silviu Herchi]

Hi,

The key to this stuff is using the exact versions of the required libs and
following blindly the instructions (the pwlib and openh323 libraries from
sourceforge.net worked better in my case than the ones from
innaccessnetworks.com). What is the error message you get when you try to
compile asterisk-oh 0.6.5?

Regards,

Silviu


De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de João Amaro
Envoyé : mercredi 5 janvier 2005 16:38
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [Asterisk-Users] asterisk - oh323 driver

Hi List


I'm having problems starting asterisk  with asterisk-oh323-0.6.4.
I'm using this versions:
 asterisk-1.0.3
 asterisk-oh323-0.6.4
 openh323-Janus_patch4 + asterisk-0h323 patch
 pwlib-Janus_patch4


At starting time, i've this error message

# /srv/usr/sbin/asterisk -vvvc
[chan_oh323.so]
Jan  3 17:06:26 WARNING[5817]: loader.c:258 ast_load_resource:
/srv/usr/lib/asterisk/modules/chan_oh323.so: undefined symbol:
_ZNK13PSoundChannel6IsOpenEv

Jan  3 17:06:26 WARNING[5817]: loader.c:440 load_modules: Loading module
chan_oh323.so failed!


I've tried to upgrade to version 0.6.5, but i got a compile error. 
Anyone know how to solve this error ?

Thanks in advance, and have a GOOD 2005

Regardz,

João Amaro

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.6.8 - Release Date: 03/01/2005
 

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[Asterisk-Users] CAPI Question

[Aldo Bergamini]

Dear list,

I am starting to setup an asterisk pbx, using a Fritz ISDN card through
chan_capi (0.3.5). The underlying OS is SUSE 9.2; I installed asterisk
with the RPMs supplied on the DVD.

While I can dial out (I had successful outside calls), through the ISDN
card, so far I could not answer a phone call on the card.

My capi.conf file is quite fantasyless:

;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]

msn=221591030
incomingmsn=221591030
controller=1
devices=2
softdtmf=1
callgroup=1
context=from-chan_capi
;accountcode=
;echosquelch=1
;echocancel=yes
;echotail=64
;deflect=02-fastweb !!

Now if I try from one SIP extension to call my self (on 0221591030) , I
can obtain the 'ringing' of the office number. But I was not able to see
Asterisk answering the call. There is a second ISDN physical phone
connected, as well as a Zyxel ISDN router (with two analog phones
attached). Everything rings but the internal SIP extension...

After some fiddling I did activate capi debugging; here is what prints
out on the CLI during an attempt:

gamma-stargate*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
gamma-stargate*CLI reload
Jan  5 16:52:40 NOTICE[1110690736]: indications.c:397
ast_unregister_indication_country: Removed default indication country 'it'
-- CONNECT_CONF ID=002 #0x0041 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

-- INFO_IND ID=002 #0x17cc LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x800d
  InfoElement = default

-- INFO_IND ID=002 #0x17cd LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

-- CONNECT_IND ID=002 #0x17ce LEN=0049
  Controller/PLCI/NCCI= 0x201
  CIPValue= 0x10
  CalledPartyNumber   = a1221591030
  CallingPartyNumber  = 21 81221591030
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = 80 90 a3
  LLC = default
  HLC = 91 81
  AdditionalInfo  = default

Jan  5 16:52:57 NOTICE[1088080816]: chan_capi.c:1932 capi_handle_msg:
CONNECT_IND ID=002 #0x17ce LEN=0049
  Controller/PLCI/NCCI= 0x201
  CIPValue= 0x10
  CalledPartyNumber   = a1221591030
  CallingPartyNumber  = 21 81221591030
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = 80 90 a3
  LLC = default
  HLC = 91 81
  AdditionalInfo  = default

Jan  5 16:52:57 ERROR[1088080816]: chan_capi.c:2051 capi_handle_msg: did
not find device for msn = 221591030
-- INFO_IND ID=002 #0x17cf LEN=0025
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x70
  InfoElement = a1221591030

Jan  5 16:52:57 ERROR[1088080816]: chan_capi.c:1198 find_pipe: unable to
find a pipe for PLCI = 0x201 MN = 0x17cf
Jan  5 16:52:57 NOTICE[1088080816]: chan_capi.c:1302 pipe_msg: INFO_IND
ID=002 #0x17cf LEN=0025
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x70
  InfoElement = a1221591030
-- INFO_IND ID=002 #0x17d0 LEN=0016
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x18
  InfoElement = 8a

Jan  5 16:52:57 ERROR[1088080816]: chan_capi.c:1198 find_pipe: unable to
find a pipe for PLCI = 0x201 MN = 0x17d0
Jan  5 16:52:57 NOTICE[1088080816]: chan_capi.c:1302 pipe_msg: INFO_IND
ID=002 #0x17d0 LEN=0016
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x18
  InfoElement = 8a
-- DISCONNECT_IND ID=002 #0x17d3 LEN=0014
  Controller/PLCI/NCCI= 0x201
  Reason  = 0x0

Jan  5 16:52:57 ERROR[1088080816]: chan_capi.c:1198 find_pipe: unable to
find a pipe for PLCI = 0x201 MN = 0x17d3
-- INFO_IND ID=002 #0x17d4 LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8001
  InfoElement = default

-- DISCONNECT_CONF ID=002 #0x0042 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

-- DISCONNECT_IND ID=002 #0x17d5 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Reason  = 0x3400

gamma-stargate*CLI 

Unfortunately I do not understand where the problem is.

The context inside extension.conf that I would like to obtain the call is
the following:


; *** INBOUND CONTEXT: DIAL INTERNAL PHONES
[from-chan_capi]

; reach the internal dialplan context!
include = incoming
include = nba_plan


; *** INCOMING CONTEXT: FROM FRITZ! CARD
[incoming]

; 

[Asterisk-Users] Re: ISDN/SS7 book?

[Aldo Bergamini]

[EMAIL PROTECTED] is believed to have said: 

some time ago, I asked the list of a good book for learning ISDN and 
SS7. I don't need to know how to write a channel driver or something; I 
just want to know more about the possibilities and what's really sent 
back and forth. I was told the book ISDN and SS7: Architectures for 
Digital Signaling Networks by Uyless Black (ISBN 0132591936) was a 
good choice, but this seems sold out. Does anyone know about another 
book about the subject?

thanks

roy

Roy,

if you look up on Amazon you'll find it used.

HTH
Aldo


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RE: [Asterisk-Users] asterisk - oh323 driver

[Kanuri, Seshu (Company IT)]

1) Download latest openh323 Libraries, 
untar to a folder and configure them running 
./configure
2) Download latest pwlib libraries
3) Download Asterisk-oh323 untar to a folder
Edit the Makefile and configure the folder names that has pwlib 
and openh323 libraries
4) make install
The pwlib and openh323 libraries can be downloaded from : 

http://www.inaccessnetworks.com/projects/asterisk-oh323
and
http://www.openh323.org/
Latest 
Linux patches can be downloaded from
http://ftp.freshrpms.net/pub/freshrpms/redhat/8.0/apt/
Check the blog here. 
http://www.openh323.org/pipermail/openh323/2004-April/067415.html
http://www.openh323.org/pipermail/openh323/2004-April/067422.html
Seshu Kanuri


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of João 
AmaroSent: Wednesday, January 05, 2005 10:38 AMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] asterisk - oh323 driver
Hi ListI'm having problems starting asterisk  with asterisk-oh323-0.6.4.
I'm using this versions:
 asterisk-1.0.3
 asterisk-oh323-0.6.4
 openh323-Janus_patch4 + asterisk-0h323 patch
 pwlib-Janus_patch4At starting time, i've this error message

# /srv/usr/sbin/asterisk -vvvc
[chan_oh323.so]
Jan  3 17:06:26 WARNING[5817]: loader.c:258 ast_load_resource: /srv/usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: _ZNK13PSoundChannel6IsOpenEv

Jan  3 17:06:26 WARNING[5817]: loader.c:440 load_modules: Loading module chan_oh323.so failed!


I've tried to upgrade to version 0.6.5, but i got a compile error. 
Anyone know how to solve this error ?

Thanks in advance, and have a GOOD 2005

Regardz,

João Amaro





NOTICE: If received in error, please destroy and notify sender.  Sender does not waive confidentiality or privilege, and use is prohibited.

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RE: [Asterisk-Users] Kirk SIP-DECT gateway

[B. Vallet - www.acropolistelecom.net]

It works with *. But indeed must be installed on a windows client only. This
gateway is user oriented, not server oriented.
Imagine while visiting a customer abroad, you connect to a wifi hostpot with
your laptop (sorry under windows) you take your dect phone or dect earset
only and you can receive or make calls from your office. Great application
no ?

Benoit


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Remco Barende
Envoyé : mercredi 5 janvier 2005 12:02
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [Asterisk-Users] Kirk SIP-DECT gateway

Thanks, the price looks attractive but what about Asterisk support?

From your website : All standard Windows operating systems are 
supported. and we are not using Windoze :)

How many concurrent conversations are supported?

Is there a howto anywhere how this could be used with * ?

By looking at the website of the manufacturer this product isn't really 
comparable with the IP600 which has support for 8 simultaneous calls and 
can be used with repeaters to extend it's range.

For small SOHO setups it would be quite nice however.

Cheers!
Remco

On Wed, 5 Jan 2005, B. Vallet - www.acropolistelecom.net wrote:


 We have SIP-DECT gateways in Pci:

http://shop.acropolistelecom.net/product_info.php?products_id=30language=en
 or PCMCIA cards :

http://shop.acropolistelecom.net/product_info.php?manufacturers_id=11produc
 ts_id=29

 Regards

 Benoit


 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De la part de Remco
Barende
 Envoyé : mercredi 5 janvier 2005 11:29
 À : [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
 Discussion
 Objet : Re: [Asterisk-Users] Kirk SIP-DECT gateway

 I already did testing with the unit you are talking about and the
 status is that it is NOT working.

 Search the list on IP600 and asterisk.

 In brief:
 Their Skinny implementation makes the IP600 wait for an extra signal and
 refuses to ring any handset without it. The 'original' Cisco 7940 (which
 it emulates) doesn't require this signal. If you use chan_sccp the phones
 will not register (don't know why yet) and Jan Czmok was't sure if this
 extra signal (STATION_CALLINFO) is in chan_sccp anyway. Outgoing calls
 work but soundlevel is too low.

 H323 : Works for incoming and outgoing calls but none of the call features
 (hold, transfer etc.) work. This is too little funcitionaly to say that it
 works.

 If have had extensive contact about these issues with Kirk support. At the
 moment the units are very interesting but useless with *.

 The only thing we can do is keep asking for support to all the Kirk
 offices in every country or to keep enquiring their sales. I can provide
 tcpdumps, logs etc. if anyone is interested.

 I haven't heard anything about SIP support, I have it will be in the next
 firmware release for the IP600.

 Cheers!
 Remco

 On Tue, 4 Jan 2005, E s c a u x - Jordi Nelissen wrote:

 Hi,

 I just got some interesting information from Kirk Telecom
 (www.kirktelecom.com). This company has been in the business of providing
 DECT solutions (IP gateway, base stations, repeaters and handsets) either
 to
 be used with Cisco CallManager (SCCP protocol) or with the Innovaphone IP
 PBX system (H.323).

 Two important elements:

 1. It seems they foresee a SIP version of their product in Q1 2005.
 2. They are open to perform integration tests with asterisk, provided
 there
 is sufficient business potential.

 In order to convince them about the business potential of an asterisk
 integration, I would like to ask you to drop me an email stating the
 number
 of DECT installations and associated DECT phones you might be able to
sell
 in 2005, provided the Kirk solution proved to interoperate with asterisk
 (either H323 or SIP).

 Best regards,

 Jordi
 --
 w w w . e s c a u x . c o m

 IPTel : 02 686 09 02
 IPFax : 02 686 09 08
 Email : [EMAIL PROTECTED]



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[Asterisk-Users] VoIP Provider Peering

[David Ishmael]

I have several contacts that use Vonage and was wondering how I can peer
with Vonage (assuming that's possible) so that I can contact these people
through the * rather than PSTN.  Can that be done?  What about other
providers (Skype, etc)?  Is there something on the Wiki that discusses this?

Thanks,
Dave

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Re: [Asterisk-Users] chan_cornet

[Joao Pereira]

Hi
Yes, Im looking a way to connect Asterisk to HG. 

I have already oh323 configured in Asterisk, but I 
cant connect to the Siemens HG PBX by ethernet, because the HG doesnt 
support normal H.323. 
How are you connecting Asterisk with the HG PBX? 
Are you connecting thru witch port ?ethernet, Analog, or 
digital?

Whats STLS4and STMD8? All pages with theese 
products are in german

Thanks
João Pereira

PS: Sorry Steffen, but I didnt saw your email when I sent 
the email to tyhe list.

  - Original Message - 
  From: 
  richard 
  Coco 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, January 05, 2005 1:38 
  PM
  Subject: Re: [Asterisk-Users] chan_cornet 
  
  
  What do you mean with *But doesnt Digium have 
  Asterisk-HiPath solutions?*. If you are meaning a connection with Digium cards 
  so...sorry i've never usedDigium cards. 
  
  Buti thought (see your first thread 
  *connect Asterisk with Siemens HiPath HG1500*), you are looking for a way to 
  connectAsterisk to the HG.If you use HG1500 you have to configure 
  a h323 channel (h.323 or oh.323). 
  
  If not, you can try 
  toconfigurechan_capi and try to connect Asterisk (e.g with an 
  EICONDiva card)to a STLS4 (for HiPath3500) or a 
  STMD8(forHiPath3700).
  hope it will help...
  if in a few days you have additional informations about chan_cornet, 
  please let me (the list) know.
  
  thx
  Joao Pereira [EMAIL PROTECTED] wrote:
  



Hi
I dont knowif Steffen's chan_cornet is working. I emailed 
him, but with no result.
Yesterday I read this article
http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom

It has some solutions... but not yet a direct 
Asterisk-HiPath connection.

But doesnt Digium have Asterisk-HiPath 
solutions?

Joao

  - Original Message - 
  From: 
  richard Coco 
  To: Asterisk Users Mailing List 
  - Non-Commercial Discussion 
  Sent: Wednesday, January 05, 2005 
  12:13 PM
  Subject: Re: [Asterisk-Users] 
  chan_cornet 
  
  Hi,
  
  The HG1500 is a HiPath3000 board and i don't have experience with 
  Asterisk and HiPath3K.
  What we have is an Asterisk connected to a Siemens HiPath4000 over a 
  H.323 trunk using oh323 and the HG3550 board. It works fine. But the 
  Siemens HG3550 only supports H.323 V2.0 (so not a lot of features are 
  available). May be Steffen's chan_cornet will change this.
  Are there any news about this project?Joao Pereira 
  [EMAIL PROTECTED] wrote:
  Hibut 
did anyone have ever used a Siemens HiPath PBX with Asterisk?If you 
made it, please tell me how...I read that chan_cornet does 
exist...http://lists.digium.com/pipermail/asterisk-users/2004-October/069559.htmlIs 
there any Digium Hardware solution for the Asterisk HiPath 
connection?ThanksJoao Pereira- Original Message 
-From: "Luís Palma" <[EMAIL PROTECTED]>To: "Asterisk Users 
Mailing List - Non-Commercial 
Discussion"Sent: Tuesday, 
January 04, 2005 10:30 PMSubject: Re: [Asterisk-Users] connect 
Asterisk with Siemens HiPath HG1500 Hi, 
It doesn't tell you much but it looks like that you are not alone 
when trying to integrate with Siemens Hicom. It seems someone 
has decided to make it by himself. 
http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom 
Regards Luis Palma On Tue, 4 Jan 2005 12:26:38 
-, Joao Pereira <[EMAIL PROTECTED]>wrote:  
Hi  I want to know the best way to connect Asterisk to a 
Siemens HiPathHG1500  PBX. Until now I came out with 3 
solutions:   1-Asterisk being a H.323 client of 
the Siemens PBX (I believe it needs  Siemens licences and 
Digium hardware)  2-Asterisk connecting to the PSTN phones 
with Voice Modems (goodideia!!!  but its analog... 
doesnt have caller information...)  3-Using RDIS interfaces 
to connect the Siemens PBX   does someone have 
other ideias?   Thanks  Joao 
Pereira   
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mailing 

[Asterisk-Users] Asterisk consultant wanted - S. California

[Scott Stingel]

Hello-
I have a client in Orange County California who will soon need some 
consulting assistance with their new asterisk system.  I've been asked 
to help them find someone.   Skills needed would be, in order of 
importance:  Basic experience configuring and using asterisk, coding 
experience in Perl, experience with MySQL or equiv., and a knowledge of 
telephony terminology and technologies.  Would be very nice for the 
consultant to be located in Southern California to meet with customer 
occasionally.

I have developed and delivered a working prototype of the system to 
their spec., but an increasing workload prevents me from carrying it 
much further.  A number of customized (non-PBX) features will make this 
an interesting system to work on.  'C' coding or changing the asterisk 
internals should not be necessary as far as I can tell.

Please contact me OFF-LINE (ie: NOT on this mailing list):scott  at 
 evtmedia.com,

ie: do not reply to this, just send me a new email and please put: 
asterisk consulting or something in the subject line so I can see it 
among the spam!

Thanks
Scott Stingel
President
Emerging Voice Technology, Inc.
www.evtmedia.com
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RE: [Asterisk-Users] 8 pstn lines+ on Asterisk supported hardware.

[brian]

This is a pretty nice looking solution Is anyone else using it?  If
so, how is the quality?  I do like the idea of keeping the phone lines
where they are and not using the TDM400 series cards.

So far Digium's support has been underwhelming.  Today is day 3 since I
tried to contact them.  I'm not amused. 


Brian Greul
Texas Shirt Company
www.txshirts.com
713-802-0369 / 713-861-6261 (fax)

-Original Message-
From: Erik Espinoza [mailto:[EMAIL PROTECTED] 
Sent: Monday, January 03, 2005 6:19 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 8 pstn lines+ on Asterisk supported
hardware.

Or just get a couple of these:

http://www.ipeya.com/VOIP_Products.htm

(Specifically the 4 Ports FXO SIP VOIP-PSTN Gateway)

Available from eBay at a discount at:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=61839item=574196
6868rd=1

And do it all without worrying about irq's or the motherboard. Just let
the device do it's job.

Erik


On Mon, 3 Jan 2005 15:48:47 -0500, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
 I have an Asterisk with 2 TDM with 8 FXO modules and I don't have any
problems.
 
 One thing to look for is that the cards don't share any IRQ.
 
 Use a motherboard where you can assign IRQ to the PCI slot. I used an 
 Intel board.
 
 Hope this help
 
 On Mon, 3 Jan 2005 19:43:12 +0200, Hadi Jadallah
[EMAIL PROTECTED] wrote:
  Hi all,
 
  I have this project that requires me to use 8 PSTN lines and
possible more. I was thinking 2 TDM cards with FXO modules.
  The I got to read the Qs about FXO/FXS cards thread and that
scared me.
  Can anybody recommend anything that is known to work ok with no
mysterious problems?
  I was thinking OpenSwitch12 cards. What do you guys think?
  Any help is appreciated.
 
  Regards,
  Hadi
 
  --
  No virus found in this outgoing message.
  Checked by AVG Anti-Virus.
  Version: 7.0.290 / Virus Database: 265.6.7 - Release Date: 
  12/30/2004
 
 
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Re: [Asterisk-Users] asterisk - oh323 driver

[João Amaro]

Hello,

Meanwhile i've downloaded ,again, the 0.6.5 version. I'm using pwlib
and openh323 versions from sourceforge.
It compiled without errors, but the error at startup it's the same

This is the ldd output for the driver.

Shouldn't this be linked to the wrapper ?

# ldd chan_oh323.so
 libstdc++.so.5 = /usr/lib/libstdc++.so.5 (0x009b3000)
 libpthread.so.0 = /lib/tls/libpthread.so.0 (0x00a69000)
 libldap.so.2 = /usr/lib/libldap.so.2 (0x00a79000)
 libldap_r.so.2 = /usr/lib/libldap_r.so.2 (0x00aa3000)
 liblber.so.2 = /usr/lib/liblber.so.2 (0x00efb000)
 libsasl.so.7 = /usr/lib/libsasl.so.7 (0x00c36000)
 libssl.so.4 = /lib/libssl.so.4 (0x00ad1000)
 libcrypto.so.4 = /lib/libcrypto.so.4 (0x00b05000)
 libexpat.so.0 = /usr/lib/libexpat.so.0 (0x00bf6000)
 libresolv.so.2 = /lib/libresolv.so.2 (0x00fa6000)
 libdl.so.2 = /lib/libdl.so.2 (0x00f0b000)
 libc.so.6 = /lib/tls/libc.so.6 (0x00c42000)
 libm.so.6 = /lib/tls/libm.so.6 (0x00d97000)
 libgcc_s.so.1 = /lib/libgcc_s.so.1 (0x00c16000)
 /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x00d81000)
 libgdbm.so.2 = /usr/lib/libgdbm.so.2 (0x00c1f000)
 libcrypt.so.1 = /lib/libcrypt.so.1 (0x00f17000)
 libpam.so.0 = /lib/libpam.so.0 (0x00c26000)
 libgssapi_krb5.so.2 = /usr/kerberos/lib/libgssapi_krb5.so.2
(0x00db9000)
 libkrb5.so.3 = /usr/kerberos/lib/libkrb5.so.3 (0x00dcc000)
 libcom_err.so.3 = /usr/kerberos/lib/libcom_err.so.3
(0x00c2e000)
 libk5crypto.so.3 = /usr/kerberos/lib/libk5crypto.so.3
(0x00f56000)
 libz.so.1 = /usr/lib/libz.so.1 (0x00e2a000)
 liblaus.so.1 = /lib/liblaus.so.1 (0x00c3)

Regards


Silviu Herchi wrote:

  Hi,

The key to this stuff is using the exact versions of the required libs and
following blindly the instructions (the pwlib and openh323 libraries from
sourceforge.net worked better in my case than the ones from
innaccessnetworks.com). What is the error message you get when you try to
compile asterisk-oh 0.6.5?

Regards,

Silviu


De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] De la part de Joo Amaro
Envoy: mercredi 5 janvier 2005 16:38
: Asterisk Users Mailing List - Non-Commercial Discussion
Objet: [Asterisk-Users] asterisk - oh323 driver

Hi List


I'm having problems starting asterisk  with asterisk-oh323-0.6.4.
I'm using this versions:
 asterisk-1.0.3
 asterisk-oh323-0.6.4
 openh323-Janus_patch4 + asterisk-0h323 patch
 pwlib-Janus_patch4


At starting time, i've this error message

# /srv/usr/sbin/asterisk -vvvc
[chan_oh323.so]
Jan  3 17:06:26 WARNING[5817]: loader.c:258 ast_load_resource:
/srv/usr/lib/asterisk/modules/chan_oh323.so: undefined symbol:
_ZNK13PSoundChannel6IsOpenEv

Jan  3 17:06:26 WARNING[5817]: loader.c:440 load_modules: Loading module
chan_oh323.so failed!


I've tried to upgrade to version 0.6.5, but i got a compile error. 
Anyone know how to solve this error ?

Thanks in advance, and have a GOOD 2005

Regardz,

Joo Amaro

  



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RE: [Asterisk-Users] Digium T100P T1 Card

[Steven Critchfield]

On Wed, 2005-01-05 at 05:59 -0700, Wiley Siler wrote:
 Apologies if the format of the email was troublesome.  I am accessing my
 email remotely via Outlook Web Access otherwise the format would have
 been plain text.

Thats good as this message was very easy to read.

 Using my analog line for fax is not a matter of needs.  It is a matter
 of using available lines that we will have for another 18 months because
 that T1 is under a long contract.  The ISP company wants an arm and a
 leg to upgrade the T1 from analog lines to digital so that is why I am
 getting a separate voice T1 altogether.  That will leave these analog
 lines unused so I may as well dedicate them to my fax system and keep
 all the digitals for our voice.

So lets back up and look at another option here. Don't bother that ISP
T1 at all. Look at your analog lines. Depending on the location you are
at, 12 lines will be delivered via a T1 and broke out to analog lines
via a channel bank of some sort. If so, then you are already a ways to
getting closer to what you want.

Either way, I wouldn't bother the ISP for voice. Your phone lines should
come from a ILEC(former baby bell) or a CLEC(competes with ILEC). Your
ISP will probably charge you so much more because they have to pay for
the phone lines and then put the lines onto your data T1 with
specialized equipment. Depending on where the other end of your T1 is,
that can be fairly expensive for them.

If your analog lines are delivered via a T1 interface and split with a
channel bank, your phone company will probably love to upgrade your
service. You will probably still want to pick up a channel bank, and if
you already have the T100P, you will want to get a channelized T1 to
take advantage of passing the T1 through the channel bank and coming
back for the FXS ports. 

On a channelized T1 you will want to talk about getting an EM wink
lines and you can then have your DIDs.

 Finally, let me say thank you.  Your info is exactly what I needed and I
 truly appreciate it.  People who take time to help others should truly
 be applauded.  I have seen scores of replies from you to others so I
 know you are one of the best contributors here.  In fact, I usually read
 yours first just because of the quality of your replies.
 
 However, was there that much need for the criticism and arrogance in
 your reply?  Wouldn't it just be esier not to reply at all than start
 off with a complaint about my HTML formatting, go to a critique of how I
 formatted my 4 sentence email (paragraph for 4 sentences?), and finish
 up by pointing out that I don't know much about voice T1s?

No it isn't better to not reply. The complaint about HTML formatting is
important. Too many people don't understand what their formatting means
to other peoples readers. Maybe I am a bit sensitive about it as one of
our main clients has almost exclusively older ladies working for them
that have eye problems on track for the age. This has caused me to be
very aware of color choices and font sizes, or specifically choosing
relative sizes instead of hard choices. 

Consider that to be less of a complaint about you specifically and more
about the list in general. I placed in a response to you as it was
convenient and a fair portion of the list would see it. There are too
many people who compound the problem when there are several in a thread
with all kinds of alternating font sizes. 

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Steven
 Critchfield
 Sent: Wednesday, January 05, 2005 1:28 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Digium T100P T1 Card
 
 On Wed, 2005-01-05 at 01:01 -0700, Wiley Siler wrote:

 If you want some analog FXS ports, you could also go the route of an
 ADIT 600 and plug the T1 into the ADIT and route your incoming 12
 channels to the second port of the ADIT and then plug it into the T100P.
 The benefit here is you will have 12 channels left over to signal back
 from the T100P to the ADIT and have those channels routed to FXS ports.
 I used to do something similar to that with a Zhone channel bank before
 our company fully trusted asterisk.

-- 
Steven Critchfield [EMAIL PROTECTED]

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RE: [Asterisk-Users] Bootable Asterisk CD ?

[Anders F Eriksson]

Check out http://www.voip-info.org/wiki-Asterisk+Bootable+CDROM.

/Anders 

 
 A while ago, I saw some threads on booting linux w/ asterisk 
 from a CF card.
 
 I have also seen CD installs of Asterisk, which require a hdd.
 
 Has anyone come up with a bootable cd (like a Live CD), that 
 creates a ramdisk and runs asterisk, without touching the hard disk ?
 
 It would be a good tool to demo asterisk, without actuall 
 installing linux.
 I looked at AstWind, but I dont think you can use the Console 
 Channel driver with that.

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RE: [Asterisk-Users] Digium T100P T1 Card

[Peter Svensson]

On Wed, 5 Jan 2005, Wiley Siler wrote:

 So, I need to learn more about voice T1s?  Reeally?  That would be why I
 am posting to the user group in the first place.  To learn more.  The
 wiki says nothing about how PRI works because it is expected that
 someone will know.  Well, I didn't.  Had to ask.  After cruising ebay
 for 30 minutes looking at routers and reading the tech spec on the
 T100P, I figured out the very same thing regarding the fact that no
 router was needed.

[snip]

 However, was there that much need for the criticism and arrogance in
 your reply?  Wouldn't it just be esier not to reply at all than start
 off with a complaint about my HTML formatting, go to a critique of how I
 formatted my 4 sentence email (paragraph for 4 sentences?), and finish
 up by pointing out that I don't know much about voice T1s?

Normally I can be quite critical of the sometimes brusque replies on this 
list but the reply Steven sent was filled with information. He started out 
by saying that he found your email hard to read and the reasons why. He 
then stated that you have a lot to learn about T1/isdn pri which is 
probably true. This is a complex subject and if you are not familiar with 
it it may be a good idea to hire a consultant who is. 

This list is really not meant as a general educational tool for digital 
telecom. There are such resources elsewhere on the net. Once you have done 
your homework and is more knowledgeable on the topics of telecommunications 
you are in a better position to ask questions regarding Asterisk. At that 
point you will probably receive a lot more help from the members of this 
list.

Peter



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Re: [Asterisk-Users] Last callers script?

[Roger Gulbranson]

On Wed, 2005-01-05 at 11:00, Mike Dent wrote:
 Hi,
 Is there some script which can be called from a * extension to
 playback the recent incoming
 callers on a particular PSTN line?
 
 In the UK 1471 is a BT number which plays back the most recent callers
 number, it also
 gives you the option to call this number back (now charging you for
 this service too!).
 
 Is there anything similar in asterisk-land?

I have an AGI script (a modified version of calleridnamelookup.agi)
that, among other things, stores the channel and callerid in a mysql
DB.  The AGI is called from within my IVR processing on all the inbound
channels.  I happen to use this for a web page that displays the most
recent 20 calls.

Writing an AGI script to take a channel and find the last inbound
callerid should be an easy thing to do (once you have the data).

No doubt there are other ways to achieve the same result.  DBget/DBput
could be used, for example.

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[Asterisk-Users] Re: Asterisk Pbx Manager Equivalent (in plain text - apologies to those that dont like HTML mail!!)

[Tony Mountifield]

In article [EMAIL PROTECTED],
Paul Brock [EMAIL PROTECTED] wrote:
 
 Finally, Anyone know of a Digium hardware Reseller in the Uk at all?? 

www.telappliant.com

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] asterisk - oh323 driver

[João Amaro]

Hello,

Meanwhile i've downloaded ,again, the 0.6.5 version. I'm using pwlib
and openh323 versions from sourceforge.
It compiled without errors, but the error at startup it's the same

This is the ldd output for the driver.

Shouldn't this be linked to the wrapper ?

# ldd chan_oh323.so
 libstdc++.so.5 = /usr/lib/libstdc++.so.5 (0x009b3000)
 libpthread.so.0 = /lib/tls/libpthread.so.0 (0x00a69000)
 libldap.so.2 = /usr/lib/libldap.so.2 (0x00a79000)
 libldap_r.so.2 = /usr/lib/libldap_r.so.2 (0x00aa3000)
 liblber.so.2 = /usr/lib/liblber.so.2 (0x00efb000)
 libsasl.so.7 = /usr/lib/libsasl.so.7 (0x00c36000)
 libssl.so.4 = /lib/libssl.so.4 (0x00ad1000)
 libcrypto.so.4 = /lib/libcrypto.so.4 (0x00b05000)
 libexpat.so.0 = /usr/lib/libexpat.so.0 (0x00bf6000)
 libresolv.so.2 = /lib/libresolv.so.2 (0x00fa6000)
 libdl.so.2 = /lib/libdl.so.2 (0x00f0b000)
 libc.so.6 = /lib/tls/libc.so.6 (0x00c42000)
 libm.so.6 = /lib/tls/libm.so.6 (0x00d97000)
 libgcc_s.so.1 = /lib/libgcc_s.so.1 (0x00c16000)
 /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x00d81000)
 libgdbm.so.2 = /usr/lib/libgdbm.so.2 (0x00c1f000)
 libcrypt.so.1 = /lib/libcrypt.so.1 (0x00f17000)
 libpam.so.0 = /lib/libpam.so.0 (0x00c26000)
 libgssapi_krb5.so.2 = /usr/kerberos/lib/libgssapi_krb5.so.2
(0x00db9000)
 libkrb5.so.3 = /usr/kerberos/lib/libkrb5.so.3 (0x00dcc000)
 libcom_err.so.3 = /usr/kerberos/lib/libcom_err.so.3
(0x00c2e000)
 libk5crypto.so.3 = /usr/kerberos/lib/libk5crypto.so.3
(0x00f56000)
 libz.so.1 = /usr/lib/libz.so.1 (0x00e2a000)
 liblaus.so.1 = /lib/liblaus.so.1 (0x00c3)

Regards


Silviu Herchi wrote:

  Hi,

The key to this stuff is using the exact versions of the required libs and
following blindly the instructions (the pwlib and openh323 libraries from
sourceforge.net worked better in my case than the ones from
innaccessnetworks.com). What is the error message you get when you try to
compile asterisk-oh 0.6.5?

Regards,

Silviu


De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] De la part de Joo Amaro
Envoy: mercredi 5 janvier 2005 16:38
: Asterisk Users Mailing List - Non-Commercial Discussion
Objet: [Asterisk-Users] asterisk - oh323 driver

Hi List


I'm having problems starting asterisk  with asterisk-oh323-0.6.4.
I'm using this versions:
 asterisk-1.0.3
 asterisk-oh323-0.6.4
 openh323-Janus_patch4 + asterisk-0h323 patch
 pwlib-Janus_patch4


At starting time, i've this error message

# /srv/usr/sbin/asterisk -vvvc
[chan_oh323.so]
Jan  3 17:06:26 WARNING[5817]: loader.c:258 ast_load_resource:
/srv/usr/lib/asterisk/modules/chan_oh323.so: undefined symbol:
_ZNK13PSoundChannel6IsOpenEv

Jan  3 17:06:26 WARNING[5817]: loader.c:440 load_modules: Loading module
chan_oh323.so failed!


I've tried to upgrade to version 0.6.5, but i got a compile error. 
Anyone know how to solve this error ?

Thanks in advance, and have a GOOD 2005

Regardz,

Joo Amaro

  



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[Asterisk-Users] Getting Agent Channel information

[Asterisk]

Given that all agents are on SIP phones (Cisco 7940), and all outside lines
are EuroISDN channels (using a TE405P), is there any way of finding what zap
channel is being used by an agent channel within a dialplan ?

If you type show agents on the CLI you get information like:

6038 (Agent 6038) logged in on SIP/6908-8445 talking to Zap/63-1
(musiconhold is 'default')

What I want to get is the Zap/63-1 information in some form of variable -
I want to do this so that I can use ZapBarge on the Zap/63 channel.

I want to be able to let a supervisor enter an agent number, for the
dialplan to obtain the zap channel of that agent, and then to zap barge it.

I know that I can modify the C code to set a channel variable with this
info, but would prefer to do this within a dialplan, not by modifying the
source.

Thanks.

Julian

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RE: [Asterisk-Users] chan_cornet

[GIBERT Frédéric]

Hi,

You 
should connect HG1500 with Ethernet port. But be careful, because in the HG1500 
configuration, you need to declare nodes. This node should be the asterisk H323 
[EMAIL PROTECTED] The asterisk need to be declare as a gateway, and not as a 
phone!
I 
never do such config, but as I was working for siemens, I know the HG1500. Be 
careful, because siemens had cornet over H323 in is protocol, and it's an old 
version of H323.

STLS4 
is a 4 BRI ports card to connect to carrier.
STMD8 
is a card to connect 8 ISDN Siemens phones (optiset)

Bye


  -Message d'origine-De: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]De la part de Joao 
  PereiraEnvoyé: mercredi 5 janvier 2005 
  17:23À: Asterisk Users Mailing List - Non-Commercial 
  DiscussionObjet: Re: [Asterisk-Users] chan_cornet 
  
  Hi
  Yes, Im looking a way to connect Asterisk to HG. 
  
  I have already oh323 configured in Asterisk, but 
  I cant connect to the Siemens HG PBX by ethernet, because the HG doesnt 
  support normal H.323. 
  How are you connecting Asterisk with the HG PBX? 
  Are you connecting thru witch port ?ethernet, Analog, or 
  digital?
  
  Whats STLS4and STMD8? All pages with theese 
  products are in german
  
  Thanks
  João Pereira
  
  PS: Sorry Steffen, but I didnt saw your email when I sent 
  the email to tyhe list.
  
- Original Message - 
From: 
richard 
Coco 
To: Asterisk Users Mailing List - 
Non-Commercial Discussion 
Sent: Wednesday, January 05, 2005 1:38 
PM
Subject: Re: [Asterisk-Users] 
chan_cornet 

What do you mean with *But doesnt Digium have 
Asterisk-HiPath solutions?*. If you are meaning a connection with Digium 
cards so...sorry i've never usedDigium cards. 

Buti thought (see your first thread 
*connect Asterisk with Siemens HiPath HG1500*), you are looking for a way to 
connectAsterisk to the HG.If you use HG1500 you have to 
configure a h323 channel (h.323 or oh.323). 

If not, you can try 
toconfigurechan_capi and try to connect Asterisk (e.g with an 
EICONDiva card)to a STLS4 (for HiPath3500) or a 
STMD8(forHiPath3700).
hope it will help...
if in a few days you have additional informations about chan_cornet, 
please let me (the list) know.

thx
Joao Pereira [EMAIL PROTECTED] 
wrote:

  
  

  Hi
  I dont knowif Steffen's chan_cornet is working. I emailed 
  him, but with no result.
  Yesterday I read this article
  http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom
  
  It has some solutions... but not yet a direct 
  Asterisk-HiPath connection.
  
  But doesnt Digium have Asterisk-HiPath 
  solutions?
  
  Joao
  
- Original Message - 
From: 
richard Coco 
To: Asterisk Users Mailing 
List - Non-Commercial Discussion 
Sent: Wednesday, January 05, 2005 
12:13 PM
Subject: Re: [Asterisk-Users] 
chan_cornet 

Hi,

The HG1500 is a HiPath3000 board and i don't have experience with 
Asterisk and HiPath3K.
What we have is an Asterisk connected to a Siemens HiPath4000 over 
a H.323 trunk using oh323 and the HG3550 board. It works fine. But the 
Siemens HG3550 only supports H.323 V2.0 (so not a lot of features are 
available). May be Steffen's chan_cornet will change this.
Are there any news about this project?Joao Pereira 
[EMAIL PROTECTED] wrote:
Hibut 
  did anyone have ever used a Siemens HiPath PBX with Asterisk?If 
  you made it, please tell me how...I read that chan_cornet does 
  exist...http://lists.digium.com/pipermail/asterisk-users/2004-October/069559.htmlIs 
  there any Digium Hardware solution for the Asterisk HiPath 
  connection?ThanksJoao Pereira- Original 
  Message -From: "Luís Palma" <[EMAIL PROTECTED]>To: 
  "Asterisk Users Mailing List - Non-Commercial 
  Discussion"Sent: Tuesday, 
  January 04, 2005 10:30 PMSubject: Re: [Asterisk-Users] connect 
  Asterisk with Siemens HiPath HG1500 
  Hi, It doesn't tell you much but it looks like that 
  you are not alone when trying to integrate with Siemens Hicom. 
  It seems someone has decided to make it by 
  himself. 
  http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom 
  Regards Luis Palma On Tue, 4 Jan 2005 12:26:38 
  -, Joao Pereira <[EMAIL PROTECTED]>wrote:  
  Hi  I want to know the best way to connect Asterisk to a 
  Siemens HiPathHG1500  PBX. Until now I came out with 3 
  solutions:   1-Asterisk being a H.323 client 
  of the Siemens PBX (I believe it needs  Siemens licences 
  and Digium hardware)  2-Asterisk 

RE: [Asterisk-Users] Bootable Asterisk CD ?

[Jeff R Glassman]

Try http://knopsterisk.com/

Jeff

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Shahed
Moolji
Sent: Wednesday, January 05, 2005 11:23 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Bootable Asterisk CD ?


Hi All,

A while ago, I saw some threads on booting linux w/ asterisk from a CF card.

I have also seen CD installs of Asterisk, which require a hdd.

Has anyone come up with a bootable cd (like a Live CD), that
creates a ramdisk and runs asterisk, without touching the hard disk ?

It would be a good tool to demo asterisk, without actuall installing linux.
I looked at AstWind, but I dont think you can use the Console Channel
driver with that.

Thanks
Shahed.

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Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

[Tais M. Hansen]

On Monday 03 January 2005 19:34, [EMAIL PROTECTED] wrote:
 Has anyone had success using a TE410P card in an HP-Compaq DL380 G4
 server?

We're struggeling with the same thing right now. We have several TE410Ps 
working on DL380G3s, but have so far been unsuccessful in getting it to work 
on the G4.

Our G4 config is dual xeon 3.6ghz, 2gb ram, kernel 2.6.10 and 2.4.28.

zaptel and wct4xxp modules loads fine. At this point the flashing red lights 
on the wct4xxp are turned off. zttool shows all spans are OK, no matter if 
there are anything plugged in.

-- 
Regards,
Tais M. Hansen
ComX Networks A/S
Tel: +45-70257474
Fax: +45-70257374


pgpBbG4BeTDZW.pgp
Description: PGP signature
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Re: [Asterisk-Users] IAXy Static... and other issues

[Wilson Pickett]

 Anyone with any thoughts I would really appreciate any help.

Having owned one of these for several months, here's what I've learned.

If you don't have a power supply that is stable at 1200ma the IAXy
won't work. It kinda/sorta worked on a 1amp supply but it works more
consistently on a 1500ma (1.5A) supply. Symptoms of bad power supply:
noise, hoorible static at random moments, being perfectly connected
with ring (but not ringing phone), apparent total playing dead but
accepting provisioning, etc. Basic flaky behaviour has for the moment
been stopped by a new power supply.

Further, many older European phones will not ring on the regular
asterisk setup and this includes when connected to an IAXy. I have
several three year old Siemens cordless phones that will NOT ring on
the IAXy. They don't ring on asterisk either without using the 25hz
ring patch in wcfxs.c

Every once in a while, the IAXy is no longer online. Someone has
pinned this to bootp and not renewing DHCP properly. This is possible,
I am trying to *prove* it now. However, giving the IAXy a fixed
internal ip will be an easy fix.
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Re: [Asterisk-Users] Status of SNOM Intercom

[Nils Ohlmeier]

On Wednesday 05 January 2005 16:25, Raymond McKay wrote:
 As of now, the hack still applies.  It would be wonderful though if
 somebody could implement a command line variable that allows you to append
 anything to the SIP URI in the form of variable=variable.  Right now the
 patch essentially breaks the VXML_URL functionality right now as stands.

Ok in that case the patch still has to be applied, because the procedure to 
start an intercom call still remains the same (parameter intercom=true in the 
request URI).

Greetings
  Nils Ohlmeier
-- 
snom technology AGPascalstrasse 10bD-10581 Berlin
Nils Ohlmeier
mailto:[EMAIL PROTECTED]  http://www.snom.com
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RE: [Asterisk-Users] Digium T100P T1 Card

[Wiley Siler]

LOL - Thanks for not getting mad about my email.  I just felt a little
stung for being uneducated about T1s but we have to learn somewhere!  
I completely understand your concerns and will try to comply as best as
I can.  
Again, thanks for being such a contributor to the this support system!! 

To further explain my siutation, I should give you some more background
on my setup.  My current setup has an AdTran 616 on the wall breaking
out my 6 analog lines and delivering my data to the office.  I have two
TDM400P cards receiving 6 analog lines which are used for both fax and
voice.  I have had numerous problems with this ISP and I just want to
get away as soon as possible.  Problem is, I have a contract that won't
expire for a while so I need to use these lines for something.  The ISP
wants a contract extension and some setious cash to do the upgrade.
Better to just seek alternate service.

I originally bought my T100P thinking I would get digital lines and all
the goodies involved.  Then budget constraints and an ISP that wants too
mcuh to convert me to Digital lead to a temporary solution.  I would use
the analog lines for a while longer.  Well, that has run its course and
I have to get to something more stable.  The PRI card looks pretty good
at this point.

So getting back to the T1 PRI issue (and I am playing catch up here), my
goal is to just deliver new service into this office over my T100P and
just dump nothing but fax out those old lines.  That way I can reserve
the digitals for our truly important calls and still reap the benefit of
having those old analog lines.

I will have to google up ILEC and CLEC for more info b/c that is new to
me as well.

Thanks again,
Wiley



 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, January 05, 2005 9:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Digium T100P T1 Card

On Wed, 2005-01-05 at 05:59 -0700, Wiley Siler wrote:
 Apologies if the format of the email was troublesome.  I am accessing 
 my email remotely via Outlook Web Access otherwise the format would 
 have been plain text.

Thats good as this message was very easy to read.

 Using my analog line for fax is not a matter of needs.  It is a matter

 of using available lines that we will have for another 18 months 
 because that T1 is under a long contract.  The ISP company wants an 
 arm and a leg to upgrade the T1 from analog lines to digital so that 
 is why I am getting a separate voice T1 altogether.  That will leave 
 these analog lines unused so I may as well dedicate them to my fax 
 system and keep all the digitals for our voice.

So lets back up and look at another option here. Don't bother that ISP
T1 at all. Look at your analog lines. Depending on the location you are
at, 12 lines will be delivered via a T1 and broke out to analog lines
via a channel bank of some sort. If so, then you are already a ways to
getting closer to what you want.

Either way, I wouldn't bother the ISP for voice. Your phone lines should
come from a ILEC(former baby bell) or a CLEC(competes with ILEC). Your
ISP will probably charge you so much more because they have to pay for
the phone lines and then put the lines onto your data T1 with
specialized equipment. Depending on where the other end of your T1 is,
that can be fairly expensive for them.

If your analog lines are delivered via a T1 interface and split with a
channel bank, your phone company will probably love to upgrade your
service. You will probably still want to pick up a channel bank, and if
you already have the T100P, you will want to get a channelized T1 to
take advantage of passing the T1 through the channel bank and coming
back for the FXS ports. 

On a channelized T1 you will want to talk about getting an EM wink
lines and you can then have your DIDs.

 Finally, let me say thank you.  Your info is exactly what I needed and

 I truly appreciate it.  People who take time to help others should 
 truly be applauded.  I have seen scores of replies from you to others 
 so I know you are one of the best contributors here.  In fact, I 
 usually read yours first just because of the quality of your replies.
 
 However, was there that much need for the criticism and arrogance in 
 your reply?  Wouldn't it just be esier not to reply at all than start 
 off with a complaint about my HTML formatting, go to a critique of how

 I formatted my 4 sentence email (paragraph for 4 sentences?), and 
 finish up by pointing out that I don't know much about voice T1s?

No it isn't better to not reply. The complaint about HTML formatting is
important. Too many people don't understand what their formatting means
to other peoples readers. Maybe I am a bit sensitive about it as one of
our main clients has almost exclusively older ladies working for them
that have eye problems on track for the age. This has caused me to be
very aware of color choices 

Re: [Asterisk-Users] Polycom Buddy Feature

[Matt Gibson]

Jon Radon wrote:
I think you need to subscribe to the context where exten 200 exists. 
I'm not sure if it'll work with an arbitrary context.  You may also
want to try sending the hint to just one phone.  I'm not 100% on the
format for sending the hint to multiple phones.



Just a little bit I got from our Polycom reseller - Maybe this makes 
sense. Anyone have any contacts at Polycom so we can get them to change 
this? :)

-
Hi Matt,
I have checked the Polycom Admin Manual and did not see any hard
limit for the buddy list.  The manual mentioned that it will add entries
to the buddy list when they are added to the contact list.  The only thing
I could think of that limits the number to 7 is that it corresponds to the
number of lines available for the IP500.  It would make sense that you are
only allowed to monitor 7 since the phone can only handle 6 calls max.
--
Matt Gibson
VOIP Administrator
NJ Tech Solutions
1.314.480.4550 ex. 6400
1.877.999.4678 ex. 6400
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Re: [Asterisk-Users] Happy Wednesday Morning SMS question, slightly OT

[David Boyd]

On Wed, 2005-01-05 at 10:23, Jay Milk wrote:
 We all mostly know that * as well as various SIP phones support SMS.
 While the final setup is somewhat of a mystery, there are reports of
 those lucky souls who have it working.  We also know that in order to
 send an SMS to a mobile phone, we need to connect to some SMS message
 center and get the word out that way.  
 
 Now, here's the new (?) element:  How can I *accept* messages on my
 voip-based US landline?  I know that if I send an SMS from my T-Mobile
 phone to a friend's Verizon phone, the message goes through, so
 somewhere there must exist a national message center that knows which
 carrier to hand the message off to.  Technically it should be possible
 to register a phone number with them to receive messages sent from
 cell-phones or from other * systems, and then to receive these messages
 through * and onto a SMS capable IP phone...?
 
 Who knows more about this?
 
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isn't SMS sent out via SS7?
dave

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Re: [Asterisk-Users] Happy Wednesday Morning SMS question, slightly OT

[Wilson Pickett]

 How can I *accept* messages on my
 voip-based US landline? 

I doubt it. SMS depends upon the sender and receiver talking via FSK
*before* the phone is answered. I wish fax worked this way, by the
way.
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RE: [Asterisk-Users] Re: Asterisk Pbx Manager Equivalent (in plain text- apologies to those that dont like HTML mail!!)

[Paul Brock]

Thx Tony, have dropped them a line :)

Paul

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: 05 January 2005 16:41
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Asterisk Pbx Manager Equivalent (in plain
text- apologies to those that dont like HTML mail!!)

In article [EMAIL PROTECTED],
Paul Brock [EMAIL PROTECTED] wrote:
 
 Finally, Anyone know of a Digium hardware Reseller in the Uk at all?? 

www.telappliant.com

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [Asterisk-Users] asterisk - oh323 driver

[Kanuri, Seshu (Company IT)]

Joao wrote:



Meanwhile i've downloaded ,again, the 0.6.5 version. 
I'm using pwlib and openh323 versions from sourceforge.It compiled without 
errors, but the error at startup it's the sameThis 
is the ldd output for the driver.Shouldn't this be linked to the wrapper 
?

I'm having problems starting asterisk  with asterisk-oh323-0.6.4.
I'm using this versions:
 asterisk-1.0.3
 asterisk-oh323-0.6.4
 openh323-Janus_patch4 + asterisk-0h323 patch
 pwlib-Janus_patch4

I am trying to compile thge latest h323 libraries from openh323.org site and also from sourceforge and I get only one error as under:/usr/include/ptlib/syncthrd.h:356: error: 'PDictionary' is used as a
type, but is not defined as a type.
The error seems to be in synthrd.h file at line 356, where it is used but not declared in the beginning. Does nyoneone know how to fix this?My be we need to declare PDictionary as a type in the file. Does anyone know how to declare this in the header file?Seshu Kanuri




NOTICE: If received in error, please destroy and notify sender.  Sender does not waive confidentiality or privilege, and use is prohibited.

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[Asterisk-Users] Broadvoice / * re-register issues

[kevin]

HELP!

Ok, so I have the following SIP.CONF:

[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw

register =
[EMAIL PROTECTED]:X:[EMAIL PROTECTED]/1234

[sip.broadvoice.com]
type=peer
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=##
context=default
dtmfmode=inband
canreinvite=no
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw

[kevin]
type=friend
regexten=1001
username=kevin
fromuser=Kevin Marvin   ; Specify user to put in from instead of
callerid
secret=XX
host=dynamic
canreinvite=no
defaultip=10.1.1.16
amaflags=default; Choices are default, omit, billing,
documentation
dtmfmode=inband

[laptop2]
type=friend
regexten=1005
;username=notebook
fromuser=notebook
secret=XXX
auth=md5
host=dynamic
qualify=1000
callerid=Notebook 1005
disallow=all
allow=gsm
context=default
dtmfmode=inband
reinvite=no
canreinvite=no

[1002]
type=friend
username=1002
secret=
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=1002
nat=0


My server, on startup, registers with BV and works great.  Calls come in,
calls go out, life is good.

After about 4 or 5 minutes, however, I cannot get incoming calls.  It
either just rings or goes busy, and never executes the dialplan in
extensions.conf.

I have recompiled, applied the BV patch, all to no avail.  Please help me
understand what I am missing before the wife kills me, which would be very
very bad for my racquetball game.


Thanks,

- Kevin
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[Asterisk-Users] (no subject)

[kevin]

HELP!

Ok, so I have the following SIP.CONF:

[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw

register =
[EMAIL PROTECTED]:X:[EMAIL PROTECTED]/1234

[sip.broadvoice.com]
type=peer
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=##
context=default
dtmfmode=inband
canreinvite=no
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw

[kevin]
type=friend
regexten=1001
username=kevin
fromuser=Kevin Marvin   ; Specify user to put in from instead of
callerid
secret=XX
host=dynamic
canreinvite=no
defaultip=10.1.1.16
amaflags=default; Choices are default, omit, billing,
documentation
dtmfmode=inband

[laptop2]
type=friend
regexten=1005
;username=notebook
fromuser=notebook
secret=XXX
auth=md5
host=dynamic
qualify=1000
callerid=Notebook 1005
disallow=all
allow=gsm
context=default
dtmfmode=inband
reinvite=no
canreinvite=no

[1002]
type=friend
username=1002
secret=
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=1002
nat=0


My server, on startup, registers with BV and works great.  Calls come in,
calls go out, life is good.

After about 4 or 5 minutes, however, I cannot get incoming calls.  It
either just rings or goes busy, and never executes the dialplan in
extensions.conf.

I have recompiled, applied the BV patch, all to no avail.  Please help me
understand what I am missing before the wife kills me, which would be very
very bad for my racquetball game.


Thanks,

- Kevin


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RE: [Asterisk-Users] Digium T100P T1 Card

[Wiley Siler]

Peter,

I also made it a point to voice my appreciation and recognize the fact
that Stephen is major contributor here.  I also acknowledged his
generous explanations.  I have also since replied to his reply and
thanked him again as well.  

A consultant so I can get a T1 PRI on my wall and use it with my
Asterisk box?  LMAO.  That is the dumbest thing I have ever heard.  I
need a consultant so I can get a T1 with PRI?  Please.  I am just trying
to better understand how the Digium PRI card works and how it
interconnects to the ISP.

I checked the Wiki and I check Digium.  Neither one said install PRI
card and no other router is needed.  Or rather, what I did find was the
reference that said that your * box will act as a router with the PRI
card.  Then it clicked and I got it.  Having never had a PRI T1, I did
not know it would be unlike my current T1 which has an AdTran to break
out my voice from the data.  So asking how to connect the Digium card
seemed natural for this discussion. 

Again, thank you for you contribution to the discussion.  If my previous
response was offensive to anyone, especially Stephen, I apologize.  If
it is not clear, I view the gurus here as generous contributors.  I just
generally don't like criticism with my answers.  Maybe that is the price
to apy but really.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Svensson
Sent: Wednesday, January 05, 2005 9:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Digium T100P T1 Card

On Wed, 5 Jan 2005, Wiley Siler wrote:

 So, I need to learn more about voice T1s?  Reeally?  That would be why

 I am posting to the user group in the first place.  To learn more.  
 The wiki says nothing about how PRI works because it is expected that 
 someone will know.  Well, I didn't.  Had to ask.  After cruising ebay 
 for 30 minutes looking at routers and reading the tech spec on the 
 T100P, I figured out the very same thing regarding the fact that no 
 router was needed.

[snip]

 However, was there that much need for the criticism and arrogance in 
 your reply?  Wouldn't it just be esier not to reply at all than start 
 off with a complaint about my HTML formatting, go to a critique of how

 I formatted my 4 sentence email (paragraph for 4 sentences?), and 
 finish up by pointing out that I don't know much about voice T1s?

Normally I can be quite critical of the sometimes brusque replies on
this list but the reply Steven sent was filled with information. He
started out by saying that he found your email hard to read and the
reasons why. He then stated that you have a lot to learn about T1/isdn
pri which is probably true. This is a complex subject and if you are not
familiar with it it may be a good idea to hire a consultant who is. 

This list is really not meant as a general educational tool for digital
telecom. There are such resources elsewhere on the net. Once you have
done your homework and is more knowledgeable on the topics of
telecommunications you are in a better position to ask questions
regarding Asterisk. At that point you will probably receive a lot more
help from the members of this list.

Peter



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RE: [Asterisk-Users] Bootable Asterisk CD ?

[Brian West]

What no download?  Just wait AsterLinux will be out soon.

bkw

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jeff R Glassman
 Sent: Wednesday, January 05, 2005 11:24 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Bootable Asterisk CD ?
 
 Try http://knopsterisk.com/
 
 Jeff
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Shahed
 Moolji
 Sent: Wednesday, January 05, 2005 11:23 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Bootable Asterisk CD ?
 
 
 Hi All,
 
 A while ago, I saw some threads on booting linux w/ asterisk from a CF
 card.
 
 I have also seen CD installs of Asterisk, which require a hdd.
 
 Has anyone come up with a bootable cd (like a Live CD), that
 creates a ramdisk and runs asterisk, without touching the hard disk ?
 
 It would be a good tool to demo asterisk, without actuall installing
 linux.
 I looked at AstWind, but I dont think you can use the Console Channel
 driver with that.
 
 Thanks
 Shahed.
 
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[Asterisk-Users] lcdproc and asterisk

[Corvin]

Hi!

I would like to use lcdproc and asterisk. 
Any hints or links? Maybe someone
has experience in such matter. I am working
on such solution. I've heard of SAPBX.
Thanks for any help.

Regards,
Corvin
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[Asterisk-Users] Broadvoice / * re-register issues

[kevin]

HELP!

Ok, so I have the following SIP.CONF:

[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw

register =
[EMAIL PROTECTED]:X:[EMAIL PROTECTED]/1234

[sip.broadvoice.com]
type=peer
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=##
context=default
dtmfmode=inband
canreinvite=no
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw

[kevin]
type=friend
regexten=1001
username=kevin
fromuser=Kevin Marvin   ; Specify user to put in from instead of
callerid
secret=XX
host=dynamic
canreinvite=no
defaultip=10.1.1.16
amaflags=default; Choices are default, omit, billing,
documentation
dtmfmode=inband

[laptop2]
type=friend
regexten=1005
;username=notebook
fromuser=notebook
secret=XXX
auth=md5
host=dynamic
qualify=1000
callerid=Notebook 1005
disallow=all
allow=gsm
context=default
dtmfmode=inband
reinvite=no
canreinvite=no

[1002]
type=friend
username=1002
secret=
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=1002
nat=0


My server, on startup, registers with BV and works great.  Calls come in,
calls go out, life is good.

After about 4 or 5 minutes, however, I cannot get incoming calls.  It
either just rings or goes busy, and never executes the dialplan in
extensions.conf.

I have recompiled, applied the BV patch, all to no avail.  Please help me
understand what I am missing before the wife kills me, which would be very
very bad for my racquetball game.


Thanks,

- Kevin


p.s. - my mail system is having trouble too, so I may have sent this more
than once.  1 thing at a time :)
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Re: [Asterisk-Users] Can't initiate a call with X-Lite.

[Andy Howell]

Rich Adamson wrote:
It is just sending a sip invite to [EMAIL PROTECTED] Does the 
X-Lite need to connect to via a proxy?

No. You should work on configuring xlite to register with asterisk.
Thanks. I can get it to work that way. What I was trying to simulate was 
an external user calling in. Sorry, I should have stated that.

From your asterisk CLI, try sip debug to see the flow of packets to/from
asterisk; sip no debug will shut it off.
That is just what I needed. I found that asterisk is looking in the 
'default' context for the extension, whereas our extensions are under 
[from-sip]. I've got a little more configuring to do. Thanks.

Andy
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[Asterisk-Users] Sending DTMF to PSTN problem with SIP

[CClarke]

Dear All ~

I have * setup  running ok (with two Wildcard X100P's to PSTN). I also have
two analog phones connected into same through a SIPURA 2000. These work fine,
except that when I call out through PSTN  try to send DTMF tones to (say) a
remote PBX to dial an extension, the gain seems to go wild (high), and the
DTMF tones are not recognized at the other end.

I tried setting the SIP2000 to use inband dtmfmode (as opposed to auto), and
likewise in sip.conf, but no success.

btw, I've also set relaxdtmf=yes in zapata.conf since inbound calls sometimes
seem to have trouble dialling extensions.

A soft IAX phone (e.g. DIAXPhone) works ok, so I suspect my SIP2000/sip.conf
setup, but can't see what I'm doing wrong.

Christina.
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Re: [Asterisk-Users] ISDN/SS7 book?

[Storer, Darren]

Hi Roy,

On Wed, 5 Jan 2005 15:56:39 +0100, Roy Sigurd Karlsbakk
[EMAIL PROTECTED] wrote:

 I was told the book ISDN and SS7: Architectures for
 Digital Signaling Networks by Uyless Black (ISBN 0132591936) was a
 good choice, but this seems sold out. Does anyone know about another
 book about the subject?

For a book on SS7/VoIP only, Signaling System #7 by Travis Russell
is good (ISBN 0071361197) but for a good overview of the whole TDM
telephone network including SS7, R2 and ISDN, I recommend Signaling
in Telecommunication Networks by John G. van Bosse (ISBN
0-471-57377-9).

The Travis Russell book can be found very cheaply on eBay:

http://tinyurl.com/3zd87

HTH

Darren
-- 
Darren Storer
Comgate
Telco|Internet|Broadcast
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