Re: [Asterisk-Users] E1's and span - what questions to ask my service provider

[Peter Svensson]

On Fri, 4 Feb 2005, Michael Bielicki wrote:

> It's even easier. If you are taking an E1 with euroisdn, the D-channel
> will allways be nr. 16. In neary all cases the rest of the settings
> will be: ccs,hdb3 and depending on the carrier crc4.

Other things you will need to agree on / know:

 * Number of DID digits sent (usually the last 3-4 or all digits)
 * Number of digits and numbering plan for your outgoing callerid 
   (Calling Number on outgoing calls)
 * The numbering plans used for incoming caller id 
   on national/international calls.
 * The numbering plan expected for the called number on outgoing calls.
 * channel selection order. Normally a net end will hunt ascending 
   and a cpe interface will hunt descending. Make sure you are hunting
   in the opposite direction of the other end. This removes the risk of
   both ends trying to sieze the same B channel at the same time.
   Configured in Asterisk through the use of 'g' or 'G' on the channel 
   specification.

Peter


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RE: [Asterisk-Users] Manager API

[Bill Seddon]

<< Has anyone tried the API before?>>

Of course.  Here is a snippet taken from Nicolas Gudino's op_server.pl to
originate a call.  

my $comando = "Action: Originate\r\n";
$comando .= "Channel: $originate\r\n";
$comando .= "Exten: $canal\r\n";
$comando .= "Context: $meetme_context\r\n";
$comando .= "Priority: 1\r\n";
$comando .= "\r\n";
send_command_to_manager($comando);


Mondial Software Limited
020 7043 2795
www.mondialsoftware.com


Click here to view our presentation of Cash Controller showing its
forecasting and automated bank reconciliation features


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: February 04, 2005 7:41 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Manager API

Hi,

I try to use Manager API to originate a call from a channel to an 
existing extension. Based on "sip channel show" command, the Manager 
initiates a call to the channel only. It doesn't generate a call to 
the extension. So the originate call API of Manager is failed. I think 
I pretty much follow the API description at http://www.voip-
info.org/wiki-Asterisk+Manager+API+Action+Originate. Has anyone tried 
the API before? Thanks.

Jason
-- 

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[Asterisk-Users] TMD card to buy.

[asterisk asterisk]

Hello ,
I want to install a littel office . I have some question regarding to it.
My office has 8 analog line in and we would 20 line out (analog ).
So as a sow I need 2 TMD fxo card  and 5 TMD txs card , Am I right ?
Can I use these card in one PC ?, or I need more PC s ? Can I install driver in one PC for 7 TMD cards ?
Iam from Hungary , where can I purchase cards ? Is there any company in Hungary or nearby hungary ?
 
Thanks a lot !
 
 
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[Asterisk-Users] why asterisk and ser

[Altus Snyman]

Good day all
Why would u use asterisk and ser together and what is the big
difference?
Thanks
altus

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Re: [Asterisk-Users] TMD card to buy.

[Rich Adamson]

> I want to install a littel office . I have some question regarding to it.
> My office has 8 analog line in and we would 20 line out (analog ).
> So as a sow I need 2 TMD fxo card  and 5 TMD txs card , Am I right ?

That would be one way to do it, but I don't think you'll find many
PC's that will have 7 slots for those cards.

You might consider using 20 SIP phones instead of the analog phones.
Twenty Grandstream phones would cost about $1500 (US) while five of
the TDM fxs cards would cost about $2000 (US).

Another way would be to purchase a Digium T1/E1 card and a Channel
Bank. A channel bank with fxs cards will handle 24 analog phones,
and would connect to asterisk via a T1/E1 cable.

> Can I use these card in one PC ?, or I need more PC s ? Can I install 
> driver in one PC for 7 TMD cards ?

I would not suggest running any more then two TDM cards in a single
PC as you're likely to have interrupt/bus problems.

> Iam from Hungary , where can I purchase cards ? Is there any 
> company in Hungary or nearby hungary ?

I don't know. But, you might look over the documentation and
references on the Wiki ( http://www.voip-info.org ). Lots of
good information and some references to where to purchase stuff.



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Re: [Asterisk-Users] Nortel i2004 support asterisk?

[Matt Darnell]

> > Simple Answer:  No.  the i2004 uses the proprietary nortel UNISTIM
> > protocol.  Asterisk uses SIP, IAX, SCCP, H.323, but not UNISTIM.
> >
> > Complex answer:  It depends on how much you really want it.  There has
> > been an open-sourced implementation of a UNITSTIM server done by
> > Cedric Hans.  It is located at http://www.mlkj.net/UNISTIM/voi.tar.bz2
> > (Note: I have not tried it myself yet).  With some work it could be
> > modified
> 
> Yup, it works. I took a copy of voi to our local Nortel distributor's
> office and showed their engineers how their phones can be used without
> their call server.
> 
> Honestly, I'm not sure if chan_unistim makes much sense.
> a) The phones aren't cheap - over here they cost as much as Cisco 7940's.
> b) Nortel is already going to SIP. Their latest switches all support
> SIP. I guess in the end they might produce firmware to upgrade the i2004
> to SIP.


You were able to complete calls from one phone to another?

The installation doesn't look that difficult.

It looks like it was a lot of work to reverse engineer it.

-Matt
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Re: [Asterisk-Users] Q: How to get the preset callerid from a CLID-no-screen E1-PRI

[Frank Sautter]

Kevin P. Fleming wrote:
Frank Sautter wrote:
our customer uses this feature to show the callerid of the original 
caller when redirecting a call to a mobile phone.
That is RDNIS, it shows the "redirected number". In other words, it's 
not CLID (Calling Line ID).

Check the RDNIS channel variable to see what it holds when you receive 
one of these calls.
no, they are not deflecting the call.
they are answering the call and making a new one to the mobile phone. 
RDNIS is empty.

the main problem are not the redirected calls, but 'normal' calls from 
there showing the trunk CLID instead of the trunk CLID plus the local 
extension in the CLID.

if anyone is interested, i can arrange calls from there for debugging 
purpose.
the CLID shown should be +497031714717 but actually shown is +4970317145 
(the assigned number range they have is +497031714500 to +497031714999)

regards
 frank sautter
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[Asterisk-Users] Re: E1's and span - what questions to ask my service provider

[Vikram Rangnekar]

thanks you for the information its extremely useful.


+++ Peter Svensson [04/02/05 08:59 +0100]:
> On Fri, 4 Feb 2005, Michael Bielicki wrote:
> 
> > It's even easier. If you are taking an E1 with euroisdn, the D-channel
> > will allways be nr. 16. In neary all cases the rest of the settings
> > will be: ccs,hdb3 and depending on the carrier crc4.
> 
> Other things you will need to agree on / know:
> 
>  * Number of DID digits sent (usually the last 3-4 or all digits)
>  * Number of digits and numbering plan for your outgoing callerid 
>(Calling Number on outgoing calls)
>  * The numbering plans used for incoming caller id 
>on national/international calls.
>  * The numbering plan expected for the called number on outgoing calls.
>  * channel selection order. Normally a net end will hunt ascending 
>and a cpe interface will hunt descending. Make sure you are hunting
>in the opposite direction of the other end. This removes the risk of
>both ends trying to sieze the same B channel at the same time.
>Configured in Asterisk through the use of 'g' or 'G' on the channel 
>specification.
> 
> Peter
> 
> 
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-- 
regards
Vikram (http://www.vicramresearch.com)
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[Asterisk-Users] Q: charge info on E1-PRI

[Frank Sautter]

hi,
how can the charge info from a E1-PRI be received and be forwarded to a 
classic PBX?

regards
 frank
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[Asterisk-Users] Q: how to receice the number of the called party back?

[Frank Sautter]

hi,
a feature of euroisdn is, that you dail a number e.g. 0732194490 (where 
0 is the extension of the call dispatcher) and the phone is forwarded to 
someone with an extension of 26.
our ericsson showed after the call was picked up 07321944926 and no 
longer the dialled 0732194490.

another example is, that you are dialling a number within the same area 
code e.g. 9876543 and after the phone is picked up the number in the 
display of the caller change to 07119876543.

how can this be achieved?
regards
 frank
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[Asterisk-Users] New Asterisk user with a goal

[Ryan Coates]

Hi All, I am rather new to the asterisk world, and new to VoIP in
general, my question seems rather simple compared to some of the
topics under discussion here :)

I have done quite a bit of reading and fiddling trying to get a system
set up, to no avail yet
basically myself and a friend (both behind NAT and Firewalls, but able
to set up the firewall rules/port mappings ourself) are interesting in
setting up a PBX each, initially just for IP based calls over the net,
so that we can call each other for free
eventually we will look at plugging this into the POTS system and
repacing all our phones with  IP phones and rouitng the calls
appropriately depending on destination

at present I am trying to test with some software phones and an
asterisk box on a virtual network (no nat, no firewalls) to see if we
can get anything working (client 1 calling client 2), before we splash
out a fair ammount on some decent IP Phones, but am not having much
joy

if anyone could give me some help/advice on the matter I would be greatful

if you need any more details please do not hesitate to ask, ill try to
answer whatever I can
-- 
Regards,

Ryan "Phoenix" Coates
[EMAIL PROTECTED]
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[Asterisk-Users] A: PRI leading 0 (area access code) or 00 (country access code) missing on incoming callerid

[Frank Sautter]

Frank Sautter wrote:
on our incoming E1-PRI from german telco Arcor the leading 0 for the 
(area access code in europe) and the 00 (country accescode in europe) 
are missing on incoming callerids.
only prepending a single 0 is not the solution as suggested by some 
writers on this list, because there is no way to differ between national 
and international callerids and it's not possible to make the decission 
based on the length of the presented callerid, as the length of the 
callerid can vary in most countries.

e.g.: i'm getting signalled 4123456789 which could be a call from 
"Barmstedt (Germany)" which has the areacode '4123' or from Switzerland 
which has the countrycode '41'

somehow our ericsson businessphone 250 fromerly connected to the same 
E1-PRI was capable of showing the correct number of leading 0s?!?
the patch i made is now available through CVS-HEAD.
thanks again to peter svensson who gave me the relevant hints where to 
look after!

it is now possible to define prefixes in zapata.conf
internationalprefix=00
nationalprefix=0
localprefix=089
privateprefix=0891234
unknownprefix=
is also made the channel restart interval per span configurable
resetinterval=86400
regards
 frank
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[Asterisk-Users] Multi Office Configuration

[Nathan Alberti]

We currently have an office which has a happily working Asterisk setup 
and all is good. We are looking to deploy the same solution to our other 
5 offices, my question relates to the interconnections between all those 
sites. The requirement is seamless dialing of extensions and the ability 
to have members of multiple offices participate in call queues. I wish 
to keep the configuration as simple and easy to troubleshoot as possible.

The options I see are as follows:
1) IAX trunks between all offices in a full mesh configuration
   - This is probably the most workable and easy to set up but quite a 
bit of config and not a "dynamic" as I'd like it.

2) DUNDi
   - Concerns about its maturity and I found it hard to trouble shoot, 
there was not allot of documentation available. I got it working 
successfully but I'm not 100% sure its the way to go.

3) Configure internal DNS servers for ENUM and have asterisk use this 
for call routing.
   - Don't know enough about it yet to test but from what I have read 
it might be a possibility.

4) "Switch" configuration on all stub servers back to the central 
Asterisk box.

Does anyone have any examples of how they have configured Asterisk in a 
similar situation or better ideas that the ones above ?

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Re: [Asterisk-Users] Re: IAX2 Rejected connection attempt (voiptalk.org)

[Mark Benson]

Cheers Tony,
That sorted it! - have passed this info onto voiptalk to update their 
help pages

You wouldn't care to add an explanation as to why user works over friend?
Cheers,
Mark
Tony Mountifield wrote:
Mark Benson <[EMAIL PROTECTED]> wrote:
 

Tony Mountifield wrote:
   

Mark Benson <[EMAIL PROTECTED]> wrote:
 

Have a problem that I have been battling with for a few days now with 
help from voiptalk.org support.but I thought someone here might have 
seen this before.

I have an asterisk box running on a real non nat'ed ip address with an 
incoming number from voiptalk.org on IAX2.

The problem I am seeing with or without firewall rules in place (port 
4569 udp open or all ports open ie firewall rules flushed) is rejected 
connection attempt from xxx.xxx.xxx.xxx which is voiptalks IAX server...
   

Where is this message coming from? Asterisk? kernel IPtables? Also, where
is it appearing?  e.g. /var/log/messages, console, or Asterisk log file.
What does your iax.conf look like?
 

The message is on the asterisk console - this is all I see even if iax2 
debug is on and verbose is 30+
   

OK, that says that it is Asterisk that is rejecting the connection, not
any firewall.
 

iax.conf looks like this (more or less - comments removed)
[general]
bindport=4569
allow=all   ; same as bandwidth=high
disallow=lpc10
jitterbuffer=no
[voiptalk]
type=peer
username=
secret=xx
context=default
host=iax.voiptalk.org
[08700nn]
type=peer
username=08700nn
context=default
host=iax.voiptalk.org
Last two items as per voiptalks' instructions (user and pass and 0870 no 
removed for list)
   

The [08700nn] section should be type=user, not type=peer. That is
almost certainly the cause of your problem. Also, some items are not
required, and will be ignored (e.g. [voiptalk] doesn't need context, and
[08700nn] doesn't need username or host).
Here is my working setup:
[08700nn]
type=user
notransfer=yes
context=voiptalk-incoming
[voiptalk]
type=peer
username=
secret=xx
host=iax.voiptalk.org
notransfer=yes
;trunk=yes
qualify=yes
Hope this helps!
Cheers
Tony
 

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Re: [Asterisk-Users] Q: how to receice the number of the called party back?

[Peter Svensson]

On Fri, 4 Feb 2005, Frank Sautter wrote:

> a feature of euroisdn is, that you dail a number e.g. 0732194490 (where 
> 0 is the extension of the call dispatcher) and the phone is forwarded to 
> someone with an extension of 26.
> our ericsson showed after the call was picked up 07321944926 and no 
> longer the dialled 0732194490.
> 
> another example is, that you are dialling a number within the same area 
> code e.g. 9876543 and after the phone is picked up the number in the 
> display of the caller change to 07119876543.

Your pstn provider (and in at least the first case) the remote end 
supports passing of "Connected Line Identification Presentation" (COLP). 
I.e. the destination pbx returns the actual isdn address to which the 
calling user was connected. It is sent in the CONNECT message. See the 
ITU q.951 standard.

It is not implemented in libpri at the moment, but it should not be that 
hard to implement. The IT code is 0x4c and quick glance at the IE 
description implies that it is coded the same as calling and called 
numbers.

My suggestion would be to pass incoming COLP:s (that is, for outgoing
calls) up to the chan_zap driver which then sets a variable. The variable
can be read through a M() macro in Dial().

For outgoing COLP (when answering incoming calls) chan_zap can check a 
channel varaible when it answers the channel (instructs libpri to send a 
CONNECT) and pass COLP if available. Also COLR (Connected Line 
Identification Restriction) may be implemented.

Other more integrated way of handling COLP are certainly possible.

Peter


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Re: [Asterisk-Users] why asterisk and ser

[Nathan Alberti]

http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20not-proxy
Altus Snyman wrote:
Good day all
Why would u use asterisk and ser together and what is the big
difference?
Thanks
altus
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[Asterisk-Users] ASTCC Apllication

[Daniel Eboa]

Hello,

I have some problem using ASTCC application. I’ve
installed the application and everything works well. I’ve created card
numbers, routes trunk and others. When I dial the desired number (77) in my
case, I’m prompted to enter my card number. All goes well till I’m
prompted to enter the destination number. When I enter a destination number,
the system says it’s not a recognized number and the call doesn’t
go through. Can any one help me out with this issue? Is there a file where I can
define extensions like in extensions.conf? 

 

Thanks.

 

Daniel.

 






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Re: [Asterisk-Users] Q: charge info on E1-PRI

[Peter Svensson]

On Fri, 4 Feb 2005, Frank Sautter wrote:

> how can the charge info from a E1-PRI be received and be forwarded to a 
> classic PBX?

Advice Of Charge, see the standard q.956 clause 2. There are several 
flavours: AOC-S gives the expected per time unit charge at the start of 
the call and any time the tarrif changes. AOC-D gives a running total 
during the call. AOC-E reports the total at the end of the call.

These are dscribed in an ASN.1 notation. I think some other functions like 
this are starting to make an appearance in libpri. Passing it around 
asterisk would need a few new data structures and a new frame type, I 
think.

Peter


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Re: [Asterisk-Users] Q: How to get the preset callerid from a CLID-no-screen E1-PRI

[Peter Svensson]

On Fri, 4 Feb 2005, Frank Sautter wrote:

> no, they are not deflecting the call.
> they are answering the call and making a new one to the mobile phone. 
> RDNIS is empty.
> 
> the main problem are not the redirected calls, but 'normal' calls from 
> there showing the trunk CLID instead of the trunk CLID plus the local 
> extension in the CLID.
> 
> if anyone is interested, i can arrange calls from there for debugging 
> purpose.
> the CLID shown should be +497031714717 but actually shown is +4970317145 
> (the assigned number range they have is +497031714500 to +497031714999)

So the operator sets an incomplete callerid? Sounds like a 
misconfiguration at the operators end.

Do a "pri intense debug span XXX" on one of the calls and post the log 
of the SETUP to CONNECT_ACK messages.

Are you sure that the RDNIS is empty? 

Peter


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[Asterisk-Users] Intertex IX66 incoming IAX

[John Middleton]

Hi,
Has anyone got incoming IAX to work on the above router.
I can call out, but incoming calls are not reaching the * box.
Has anyone got this working? Could they give me some configuration hints.
Thanks
John
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[Asterisk-Users] Re: IAX2 Rejected connection attempt (voiptalk.org)

[Tony Mountifield]

Mark Benson <[EMAIL PROTECTED]> wrote:
> Cheers Tony,
> 
> That sorted it! - have passed this info onto voiptalk to update their 
> help pages
> 
> You wouldn't care to add an explanation as to why user works over friend?

Putting friend instead of user would have worked just as well, but been
less obvious. It was peer that didn't work in the [08700nn] section.

The use of friend is as follows:

If you have two identically-named sections, one of type=peer and the other
of type=user, and there are no entries that are in both with the same name
but different values, then you can combine the sections as type=friend.

When Asterisk wants details for an outgoing call, it looks for a matching
section [whatever] with type=friend or type=peer, and takes the details
from that. When it wants details for an incoming call, it looks for a
matching section [whatever] with type=friend or type=user.

In the case of voiptalk, friend isn't useful unless you name your outgoing
section [08700nn] and use Dial(IAX2/08700nn/${EXTEN}). In that
case you could use type=friend, but I think that makes things less clear.

Hope this helps!

Cheers
Tony

> Cheers,
> 
> Mark
> 
> Tony Mountifield wrote:
> 
> >Mark Benson <[EMAIL PROTECTED]> wrote:
> >  
> >
> >>Tony Mountifield wrote:
> >>
> >>
> >>>Mark Benson <[EMAIL PROTECTED]> wrote:
> >>>  
> >>>
> Have a problem that I have been battling with for a few days now with 
> help from voiptalk.org support.but I thought someone here might have 
> seen this before.
> 
> I have an asterisk box running on a real non nat'ed ip address with an 
> incoming number from voiptalk.org on IAX2.
> 
> The problem I am seeing with or without firewall rules in place (port 
> 4569 udp open or all ports open ie firewall rules flushed) is rejected 
> connection attempt from xxx.xxx.xxx.xxx which is voiptalks IAX server...
> 
> 
> >>>Where is this message coming from? Asterisk? kernel IPtables? Also, where
> >>>is it appearing?  e.g. /var/log/messages, console, or Asterisk log file.
> >>>
> >>>What does your iax.conf look like?
> >>>  
> >>>
> >>The message is on the asterisk console - this is all I see even if iax2 
> >>debug is on and verbose is 30+
> >>
> >>
> >
> >OK, that says that it is Asterisk that is rejecting the connection, not
> >any firewall.
> >
> >  
> >
> >>iax.conf looks like this (more or less - comments removed)
> >>
> >>[general]
> >>bindport=4569
> >>
> >>allow=all   ; same as bandwidth=high
> >>disallow=lpc10
> >>
> >>jitterbuffer=no
> >>
> >>[voiptalk]
> >>type=peer
> >>username=
> >>secret=xx
> >>context=default
> >>host=iax.voiptalk.org
> >>
> >>[08700nn]
> >>type=peer
> >>username=08700nn
> >>context=default
> >>host=iax.voiptalk.org
> >>
> >>Last two items as per voiptalks' instructions (user and pass and 0870 no 
> >>removed for list)
> >>
> >>
> >
> >The [08700nn] section should be type=user, not type=peer. That is
> >almost certainly the cause of your problem. Also, some items are not
> >required, and will be ignored (e.g. [voiptalk] doesn't need context, and
> >[08700nn] doesn't need username or host).
> >
> >Here is my working setup:
> >
> >[08700nn]
> >type=user
> >notransfer=yes
> >context=voiptalk-incoming
> >
> >[voiptalk]
> >type=peer
> >username=
> >secret=xx
> >host=iax.voiptalk.org
> >notransfer=yes
> >;trunk=yes
> >qualify=yes
> >
> >
> >Hope this helps!
> >
> >Cheers
> >Tony
> >
> >  
> >
> 
> ___
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-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [Asterisk-Users] Intertex IX66 incoming IAX

[Florian Overkamp]

Hi, 

> -Original Message-
> Has anyone got incoming IAX to work on the above router.
> I can call out, but incoming calls are not reaching the * box.
> Has anyone got this working? Could they give me some 
> configuration hints.

I have that working and I didn't really do anything special to make it
happen. My asterisk does register to the gateway, so that might help a bit,
I haven't tried with port-redirection and such.

Grt,
Florian


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RE: [Asterisk-Users] howto answer a call in a queue

[Hecken, Guido]

Here you find a complete working pbxconfig for asterisk.
http://www.gwsnettech.de/work/astconfig.txt

This config is our production config and works quite well.
Perhaps there are some curious constructions in there, but they work for us
;-)
With the queue functions we have "only" these problems:
joinempty=no ; does not work
leavewhenempty = yes ; does not work
Incoming calls to the queue cannot be transfered with # transfer, even
though the queue command has the "tT" options...

Hope this helps

Guido



> -Ursprüngliche Nachricht-
> Von: Edgar de Leon [mailto:[EMAIL PROTECTED]
> Gesendet: Donnerstag, 3. Februar 2005 15:34
> An: Asterisk Users Mailing List - Non-Commercial Discussion
> Betreff: RE: [Asterisk-Users] howto answer a call in a queue
> 
> On the CLI everything seems to be ok, the call enters the queue plays the
> message, on the CLI, appear as a call entering the queue and then show a
> message saying wich agent is assigned to it! can you send me your config,
> maybe there is something im doing wrong,
> 
> thnx for all your help!!
> 
> Edgar
> 
> >> I tried everything you said, but its the same thing, when a call enters
> >> plays the sound and then is directly connected to one operator, on the
> >> operator phone only a beep i heard, what other thing can i try??
> > What's happening on the cli?
> > You should try to start asterisk with asterisk -vdc. Now you
> > should
> > see, what's going on.
> > What kind of phone do you use, perhaps you could use a softclient.
SJPhone
> > runs very stable for me.
> > Once more, do it as easy as possible, save your /etc/asterisk/*.* and
use
> > only files, you really need.
> >
> > Guido
> >
> >
> >
> >
> >>
> >> TIA
> >>
> >> Edgar
> >>
> >> > My suggestions:
> >> > Try first the easy (working) configuration then your best solution
> >> step
> > by
> >> > step.
> >> >
> >> > comment out leavewhenempty=yes ;it did not work in my system...
> >> > strategy = ringall ; seems to work
> >> > don't use groups in the first step
> >> >
> >> > ;Play an announcement as the first priority
> >> > exten => 76522,1,Playback(some_announce) ;even when using an empty
> >> file
> >> > exten => 76522,2,Queue(esculapio|tT|||300)
> >> > exten => 76522,3,Playback(some_announce_after_leaving_queue) ; if
> >> nobody
> >> > answers the call
> >> > exten => 76522,4,Hangup
> >> >
> >> > I had similiar problem in working with queues.
> >> >
> >> > Hope this helps a bit more...
> >> >
> >> > Guido Hecken
> >> >
> >> > -Ursprüngliche Nachricht-
> >> > Von: Edgar de Leon [mailto:[EMAIL PROTECTED]
> >> > Gesendet: Donnerstag, 3. Februar 2005 09:08
> >> > An: Asterisk Users Mailing List - Non-Commercial Discussion
> >> > Betreff: RE: [Asterisk-Users] howto answer a call in a queue
> >> >
> >> > Thanks for your help, here are my config for the queue,
> >> >
> >> > agents.conf
> >> >
> >> >
> >> > [agents]
> >> > musiconhold => random
> >> > autologoff=15
> >> > wrapuptime=5000
> >> > ackcall=yes
> >> > group=1
> >> > agent => 1001,3101,Edgar de Leon
> >> > agent => 1002,,Jorge Cabrera
> >> > agent => 1003,,Nati del Pozo
> >> > agent => 1004,,Emilio Perez
> >> > agent => 1005,,Diego Torres
> >> > agent => 1006,,Antonio Lopez
> >> > agent => 1007,,Luis Carlos
> >> > agent => 1008,,Luis Bonifacio
> >> > agent => 1009,,Javier Gonzalez
> >> >
> >> > queues.conf
> >> > [esculapio]
> >> > leavewhenempty = yes
> >> > music = random
> >> > strategy = fewestcalls
> >> > member => Agent/@1
> >> >
> >> > extensions.conf
> >> >
> >> > [ext-acd]
> >> > exten => 90,1,Answer
> >> > exten => 90,2,SetMusicOnHold(none)
> >> > exten => 90,3,Wait,1
> >> > exten => 90,4,AgentLogin
> >> >
> >> > ;Queue configuration
> >> > exten => 76522,1,Answer
> >> > exten => 76522,2,Wait,1
> >> > exten => 76522,3,Queue(esculapio|tT|||300)
> >> > exten => 76522,5,Hangup
> >> >
> >> > is my configuration correct?? im using the
> >> >
> >> > leavewhenempty = yes
> >> >
> >> > option, but when there are no agents the call still enters the queue,
> >> > thanks for your help
> >> >
> >> > TIA
> >> >
> >> > Edgar
> >> >
> >> >> Sometime ago, I wrote an example of a functional queue scenario.
> >> >> Perhaps you give it a try.
> >> >> http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue
> >> >>
> >> >> Btw, how is the queue command invoked in your extensions.conf?
> >> >> Post your relevant sections of queues.conf, agents.conf and
> >> >> extensions.conf.
> >> >>
> >> >> Guido Hecken
> >> >>
> >> >> -Ursprüngliche Nachricht-
> >> >> Von: Edgar de Leon [mailto:[EMAIL PROTECTED]
> >> >> Gesendet: Mittwoch, 2. Februar 2005 18:23
> >> >> An: Asterisk Users Mailing List - Non-Commercial Discussion
> >> >> Betreff: RE: [Asterisk-Users] howto answer a call in a queue
> >> >>
> >> >> Thanks for your answer, i got ackcall=yes but the call when enters
> >> only
> >> >> ring once in the agent phone and connect directly,
> >> >>
> >> >> agents.conf
> >> >>
> >> >> [age

RE: [Asterisk-Users] howto answer a call in a queue

[Edgar de Leon]

My implementation accepts fw calls, i use the # key to transfer the calls
and works very well, im using zultys softphone, im very happy with that
phone, thanx for all your help

Edgar

> Here you find a complete working pbxconfig for asterisk.
> http://www.gwsnettech.de/work/astconfig.txt
>
> This config is our production config and works quite well.
> Perhaps there are some curious constructions in there, but they work for
> us
> ;-)
> With the queue functions we have "only" these problems:
> joinempty=no ; does not work
> leavewhenempty = yes ; does not work
> Incoming calls to the queue cannot be transfered with # transfer, even
> though the queue command has the "tT" options...
>
> Hope this helps
>
> Guido
>
>
>
>> -Ursprüngliche Nachricht-
>> Von: Edgar de Leon [mailto:[EMAIL PROTECTED]
>> Gesendet: Donnerstag, 3. Februar 2005 15:34
>> An: Asterisk Users Mailing List - Non-Commercial Discussion
>> Betreff: RE: [Asterisk-Users] howto answer a call in a queue
>>
>> On the CLI everything seems to be ok, the call enters the queue plays
>> the
>> message, on the CLI, appear as a call entering the queue and then show a
>> message saying wich agent is assigned to it! can you send me your
>> config,
>> maybe there is something im doing wrong,
>>
>> thnx for all your help!!
>>
>> Edgar
>>
>> >> I tried everything you said, but its the same thing, when a call
>> enters
>> >> plays the sound and then is directly connected to one operator, on
>> the
>> >> operator phone only a beep i heard, what other thing can i try??
>> > What's happening on the cli?
>> > You should try to start asterisk with asterisk -vdc. Now you
>> > should
>> > see, what's going on.
>> > What kind of phone do you use, perhaps you could use a softclient.
> SJPhone
>> > runs very stable for me.
>> > Once more, do it as easy as possible, save your /etc/asterisk/*.* and
> use
>> > only files, you really need.
>> >
>> > Guido
>> >
>> >
>> >
>> >
>> >>
>> >> TIA
>> >>
>> >> Edgar
>> >>
>> >> > My suggestions:
>> >> > Try first the easy (working) configuration then your best solution
>> >> step
>> > by
>> >> > step.
>> >> >
>> >> > comment out leavewhenempty=yes ;it did not work in my system...
>> >> > strategy = ringall ; seems to work
>> >> > don't use groups in the first step
>> >> >
>> >> > ;Play an announcement as the first priority
>> >> > exten => 76522,1,Playback(some_announce) ;even when using an empty
>> >> file
>> >> > exten => 76522,2,Queue(esculapio|tT|||300)
>> >> > exten => 76522,3,Playback(some_announce_after_leaving_queue) ; if
>> >> nobody
>> >> > answers the call
>> >> > exten => 76522,4,Hangup
>> >> >
>> >> > I had similiar problem in working with queues.
>> >> >
>> >> > Hope this helps a bit more...
>> >> >
>> >> > Guido Hecken
>> >> >
>> >> > -Ursprüngliche Nachricht-
>> >> > Von: Edgar de Leon [mailto:[EMAIL PROTECTED]
>> >> > Gesendet: Donnerstag, 3. Februar 2005 09:08
>> >> > An: Asterisk Users Mailing List - Non-Commercial Discussion
>> >> > Betreff: RE: [Asterisk-Users] howto answer a call in a queue
>> >> >
>> >> > Thanks for your help, here are my config for the queue,
>> >> >
>> >> > agents.conf
>> >> >
>> >> >
>> >> > [agents]
>> >> > musiconhold => random
>> >> > autologoff=15
>> >> > wrapuptime=5000
>> >> > ackcall=yes
>> >> > group=1
>> >> > agent => 1001,3101,Edgar de Leon
>> >> > agent => 1002,,Jorge Cabrera
>> >> > agent => 1003,,Nati del Pozo
>> >> > agent => 1004,,Emilio Perez
>> >> > agent => 1005,,Diego Torres
>> >> > agent => 1006,,Antonio Lopez
>> >> > agent => 1007,,Luis Carlos
>> >> > agent => 1008,,Luis Bonifacio
>> >> > agent => 1009,,Javier Gonzalez
>> >> >
>> >> > queues.conf
>> >> > [esculapio]
>> >> > leavewhenempty = yes
>> >> > music = random
>> >> > strategy = fewestcalls
>> >> > member => Agent/@1
>> >> >
>> >> > extensions.conf
>> >> >
>> >> > [ext-acd]
>> >> > exten => 90,1,Answer
>> >> > exten => 90,2,SetMusicOnHold(none)
>> >> > exten => 90,3,Wait,1
>> >> > exten => 90,4,AgentLogin
>> >> >
>> >> > ;Queue configuration
>> >> > exten => 76522,1,Answer
>> >> > exten => 76522,2,Wait,1
>> >> > exten => 76522,3,Queue(esculapio|tT|||300)
>> >> > exten => 76522,5,Hangup
>> >> >
>> >> > is my configuration correct?? im using the
>> >> >
>> >> > leavewhenempty = yes
>> >> >
>> >> > option, but when there are no agents the call still enters the
>> queue,
>> >> > thanks for your help
>> >> >
>> >> > TIA
>> >> >
>> >> > Edgar
>> >> >
>> >> >> Sometime ago, I wrote an example of a functional queue scenario.
>> >> >> Perhaps you give it a try.
>> >> >> http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue
>> >> >>
>> >> >> Btw, how is the queue command invoked in your extensions.conf?
>> >> >> Post your relevant sections of queues.conf, agents.conf and
>> >> >> extensions.conf.
>> >> >>
>> >> >> Guido Hecken
>> >> >>
>> >> >> -Ursprüngliche Nachricht-
>> >> >> Von: Edgar de Leon [mailto:[EMAIL PROTECTED]
>> >> >> Gesendet: Mit

[Asterisk-Users] Bristuff and incoming call problems

[Remco Barende]

Hi list!
I have some strange problems with Asterisk 1.0.5-BRIstuffed-0.2.0-RC5.
Very regularly asterisk seems to lose connectivity with the ISDN line. If 
you try to call in you get the information tone that the number is not in 
use. Outbound calls do stil work however. Unloading the modules and 
reloading them and start/stop asterisk will solve the problem.

Another problem that occurs regularly : When you make an inbound call to 
asterisk the calling party does not get the tone that the phone is ringing 
on the receiving end. The line just seems completely dead untill the phone 
is picked up and you can hear the other party. Is this an asterisk / 
bristuff problem or something for the telco to sort out? Who should 
generate the ringing signal to the calling party?

Thanks!!!
Remco
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RE: [Asterisk-Users] Good 800 Number provider

[Trevor G. Hammonds]

Ed Greenberg wrote on Thursday, 3 February 2005 11:46 AM:

> Are there any recommendations for high quality providers that will
> assign a Toll Free number and deliver it over VOIP, while still
> allowing port-out if the service doesn't work out?  

I have had great luck with TXLink.  

http://txlink.net/voipsolutions.php


Sincerely,
Trevor Hammonds

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Re: [Asterisk-Users] Multi Office Configuration

[Duane]

Nathan Alberti wrote:
Does anyone have any examples of how they have configured Asterisk in a 
similar situation or better ideas that the ones above ?
5) use www.e164.org for your dns we already have a reasonably simple to 
use web interface which will in effect allow you to create a full mesh 
iax network on demand or fail over to pstn etc... examples can be found 
on both www.e164.org and www.asterisk.net.au...

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
"In the long run the pessimist may be proved right,
but the optimist has a better time on the trip."
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[Asterisk-Users] Swap Memory get used totally

[Angel Diaz]

Hi list,
 Time to time, my asterisk goes down.Verifying with TOP, I see the swap
memory of the computer get used totally but, I don't see what the process is
using it.
Hereis a copy wath I see doing top.
Does somebody have an idea ?

My asterisk version is >>>  Asterisk CVS-HEAD-08/18/04-22:30:24

Thanks
Angel.

 08:49:19  up  5:23,  1 user,  load average: 0.50, 0.70, 0.64
35 processes: 33 sleeping, 2 running, 0 zombie, 0 stopped
CPU states:  19.4% user  11.2% system   0.0% nice   0.0% iowait  69.4% idle
Mem:   222992k av,  191988k used,   31004k free,   0k shrd,   68604k
buff
120700k actv, 464k in_d,2384k in_c
Swap:  457844k av,1060k used,  456784k free   67764k
cached

  PID USER PRI  NI  SIZE  RSS SHARE STAT %CPU %MEM   TIME CPU COMMAND
 1238 root  15   0  6280 6212  1360 S10.5  2.7   3:49   0 asterisk
31788 root  16   0  1456 1456  1100 S 0.0  0.6   0:00   0 bash
 1200 mysql 16   0  1424 1424   896 S 0.0  0.6   0:02   0 mysqld
31855 root  15   0  1180 1180   860 R 0.0  0.5   0:00   0 top
  644 admin 15   0   948  948   600 S 0.0  0.4   0:00   0 bftpd
 1138 root  15   0   616  524   372 S 0.0  0.2   0:00   0 sshd
 1222 daemon15   0   176  176   120 S 0.0  0.0   0:00   0 atd
 1235 root  15   0   168  16896 S 0.0  0.0   0:00   0 bftpd
  992 root  15   0   188  156   112 S 0.0  0.0   0:00   0 syslogd
 1164 root  25   0   152  152 0 S 0.0  0.0   0:00   0
safe_mysqld
 1181 root  15   0   152  15288 S 0.0  0.0   0:00   0 crond
 1229 root  25   0   136  136 0 S 0.0  0.0   0:00   0 S99local
1 root  15   0   112   8456 S 0.0  0.0   0:04   0 init
  702 root  25   0   2884 0 S 0.0  0.0   0:00   0 rc
  996 root  15   0524 0 S 0.0  0.0   0:00   0 klogd
 1100 root  25   0524 0 S 0.0  0.0   0:00   0 apmd
 1152 root  24   0   1244 0 S 0.0  0.0   0:00   0 xinetd
2 root  15   0 00 0 SW0.0  0.0   0:00   0 keventd
3 root  15   0 00 0 SW0.0  0.0   0:00   0 kapmd
4 root  34  19 00 0 SWN   0.0  0.0   0:01   0
ksoftirqd_CPU0
9 root  25   0 00 0 SW0.0  0.0   0:00   0 bdflush
5 root  15   0 00 0 SW0.0  0.0   0:01   0 kswapd
6 root  15   0 00 0 SW0.0  0.0   0:00   0 kscand/DMA
7 root  15   0 00 0 SW0.0  0.0   0:02   0
kscand/Normal
8 root  15   0 00 0 SW0.0  0.0   0:00   0
kscand/HighMem
   10 root  15   0 00 0 SW0.0  0.0   0:00   0 kupdated
   11 root  23   0 00 0 SW0.0  0.0   0:00   0
mdrecoveryd
   15 root  15   0 00 0 SW0.0  0.0   0:01   0 kjournald
  617 root  15   0 00 0 SW0.0  0.0   0:00   0 kjournald
  942 root  15   0 00 0 SW0.0  0.0   0:00   0 eth1
 1014 rpc   23   0760 0 SW0.0  0.0   0:00   0 portmap
 1033 rpcuser   25   0800 0 SW0.0  0.0   0:00   0 rpc.statd



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Re: [Asterisk-Users] Installing ASTERIS@HOME, How to install on text mode same help? {Scanned}

[[EMAIL PROTECTED]]

I don't think CentOS supports K6II. Do you have an
Intel machine or a newer AMD?

--- Max <[EMAIL PROTECTED]> wrote:

> Hello~!
> 
> I see again this message on text mode installing 
> [EMAIL PROTECTED]:
> 
> "You are using unsupported hardware by CentOS, 
> press OK" if press OK reboot.
> 
> have a minor version, of [EMAIL PROTECTED] to minor
> hardware?
> 
> Max Rivera
> Brazil
> 
>   - Original Message - 
>   From: David Shaw 
>   To: Asterisk Users Mailing List - Non-Commercial
> Discussion 
>   Sent: Wednesday, February 02, 2005 12:57 PM
>   Subject: Re: [Asterisk-Users] Installing
> [EMAIL PROTECTED],How to install on text mode same help?
> {Scanned}
> 
> 
>   When it asked to install type "linux text" 
> without the "".
> 
>   But when I installed my [EMAIL PROTECTED] I believed
> it just installed..
> 
>   David
> - Original Message - 
> From: Max 
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion 
> Sent: Wednesday, February 02, 2005 2:40 AM
> Subject: [Asterisk-Users] Installing
> [EMAIL PROTECTED],How to install on text mode same help?
> {Scanned}
> 
> 
> Hello, Thanks for Help!
> 
> when try to install [EMAIL PROTECTED] powered by
> CEntOS
> normal boot, 3 minutes latter:
> 
> "You are using unsupported hardware by CentOS, 
> press OK" if press OK reboot.
> 
> I increment mor ram and CPU:
> 
> CPU K6II- 500Mhz
> 196Ram 
> HD 20GB 
> Lan cart 10/100Mb
> Fax modem genius (Lucent chipset)
> Fax Modem USR 33.66
> Sound OnBoard 
> Disk Driver 1.44
> 
> CD 52X
> 
> 
> How to install on text mode?
> 
> 
> regards!
> 
> Max Rivera
> Fprm Brazil.
> 
> -- 
> This message has been scanned for viruses and 
> dangerous content by MailScanner, and is 
> believed to be clean. 
> MailScanner thanks transtec Computers for their
> support. 
> Plase contact [EMAIL PROTECTED] if you
> have questions about this email. 
> 
> 
>

> 
> 
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Re: [Asterisk-Users] Swap Memory get used totally

[Bob Goddard]

On Friday 04 February 2005 11:45, Angel Diaz wrote:
> Hi list,
>  Time to time, my asterisk goes down.Verifying with TOP, I see the swap
> memory of the computer get used totally but, I don't see what the process
> is using it.
> Hereis a copy wath I see doing top.
> Does somebody have an idea ?
[...]

Consign top to the dustbin, use the ps command instead.

It's pointless asking us what uses up all the swap when the
screen you posted shows normal swap usage.
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Re: [Asterisk-Users] Swap Memory get used totally

[Peter Corlett]

Angel Diaz <[EMAIL PROTECTED]> wrote:
> Time to time, my asterisk goes down.Verifying with TOP, I see the
> swap memory of the computer get used totally but, I don't see what
> the process is using it. Hereis a copy wath I see doing top. Does
> somebody have an idea ?

I have no idea, because there's no problem with your swap space:

>  08:49:19  up  5:23,  1 user,  load average: 0.50, 0.70, 0.64
> 35 processes: 33 sleeping, 2 running, 0 zombie, 0 stopped
> CPU states:  19.4% user  11.2% system   0.0% nice   0.0% iowait  69.4% idle
> Mem:   222992k av,  191988k used,   31004k free,   0k shrd,   68604k buff
> 120700k actv, 464k in_d,2384k in_c
> Swap:  457844k av,1060k used,  456784k free   67764k 
> cached

You've just 1MB of your 447MB of swapspace. It looks perfectly normal
to me.

-- 
A man will joyfully pay a lawyer five hundred dollars for untying the knot that
he begrudged a clergyman fifty dollars for tying.
- Helen Rowland
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RE: [Asterisk-Users] Bristuff and incoming call problems

[Rob Scott]

I have exactly the same problem.
It was also the same with RC3.
It seems that after a couple of days of working fine, at some point
incoming calls fail but outgoing calls still work (or I would hear user
complaints earlier).

For the lack of ring problem, I do the following in extensions.conf:

[fromexternal]
exten => s,1,Ringing
exten => s,2,Wait,3
exten => s,3,Answer
exten => s,4,Wait,1
exten => s,5,Background(enter-ext-of-person)

So Asterisk singals a ringing tone for 3 seconds so that the caller's
phone has a chance to ring, then answers and plays the 'enter the
extension of the person you want to call' thing while at the same time
listening for digits.

I don't know if this is the right or expected approach but it works for
me.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Friday, February 04, 2005 12:16 PM
To: Asterisk Users List
Subject: [Asterisk-Users] Bristuff and incoming call problems

Hi list!

I have some strange problems with Asterisk 1.0.5-BRIstuffed-0.2.0-RC5.

Very regularly asterisk seems to lose connectivity with the ISDN line.
If you try to call in you get the information tone that the number is
not in use. Outbound calls do stil work however. Unloading the modules
and reloading them and start/stop asterisk will solve the problem.

Another problem that occurs regularly : When you make an inbound call to
asterisk the calling party does not get the tone that the phone is
ringing on the receiving end. The line just seems completely dead untill
the phone is picked up and you can hear the other party. Is this an
asterisk / bristuff problem or something for the telco to sort out? Who
should generate the ringing signal to the calling party?

Thanks!!!
Remco
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RE: [Asterisk-Users] Odd behaviour between Grandstream and Xlite

[Rob Scott]

I allow them to us any codec except speex (which seems to crash Asterisk
when used from an Xlite).

But it would be good if the user could choose their preferred codec
because with a softphone on a laptop sometimes you are on a connection
with good bandwidth to Asterisk and sometimes somewhere with terrible
bandwidth so you want to use a low bandwidth codec.
If Asterisk chooses for you then the codec choosing feature on the Xlite
is pointless.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, February 03, 2005 7:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Odd behaviour between Grandstream and
Xlite

> Whatever codec I choose in Xlite, when calling the Grandstream it 
> always uses the GSM codec even if it is greyed out.
> 
> Whatever codec I choose in Xlite, when getting called by the 
> Grandstream it always uses ulaw even if it is greyed out.
and what about the phone config in sip.conf ?  
what codec do you allow them to use ?

I think * doesn't care what codec is grayed out in X-lite, her use what
sip.conf tell him he can

hth
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Re: [Asterisk-Users] Bristuff and incoming call problems

[Peter Svensson]

On Fri, 4 Feb 2005, Remco Barende wrote:

> Very regularly asterisk seems to lose connectivity with the ISDN line. If 
> you try to call in you get the information tone that the number is not in 
> use. Outbound calls do stil work however. Unloading the modules and 
> reloading them and start/stop asterisk will solve the problem.

Can't help you there, we are PRI only.

> Another problem that occurs regularly : When you make an inbound call to 
> asterisk the calling party does not get the tone that the phone is ringing 
> on the receiving end. The line just seems completely dead untill the phone 
> is picked up and you can hear the other party. Is this an asterisk / 
> bristuff problem or something for the telco to sort out? Who should 
> generate the ringing signal to the calling party?

Depending on what country you are in, probably you. On isdn you have the 
option of providing audio to the reverse path (i.e. ringing, busy etc) 
before the call is set up. See the 'r' option to Dial(). Use this when 
placing a call to something which does not generate ringing toned by 
itself - i.e. isdn phones and some pbx:es.

Peter


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Re: [Asterisk-Users] ASTCC Apllication

[Rodrigo Benavides]

Que telefonos estas ocupando?

El problema puede ser la forma de enviar los DMFT (esta deberia ser por RTP 
(RFC 2833) o por Sip info)

Saludos 

Rodrigo

El Vie 04 Feb 2005 06:49, Daniel Eboa escribió:
> Hello,
>
> I have some problem using ASTCC application. I've installed the
> application and everything works well. I've created card numbers, routes
> trunk and others. When I dial the desired number (77) in my case, I'm
> prompted to enter my card number. All goes well till I'm prompted to
> enter the destination number. When I enter a destination number, the
> system says it's not a recognized number and the call doesn't go
> through. Can any one help me out with this issue? Is there a file where
> I can define extensions like in extensions.conf?
>
>
>
> Thanks.
>
>
>
> Daniel.

-- 
Rodrigo Benavides F.
Sur Comunicaciones S.A.
Santiago de Chile
56-2-3712330
[EMAIL PROTECTED]
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[Asterisk-Users] Microsoft RTC Client SDK with Asterisk

[Jeremy Davis]

I'm using the the Microsoft Real-Time Communications Client API SDK 
using Visual Studio 6 and . NET 2003 SE to make SIP calls. Using the 
examples provided I can make unregistered SIP calls fine, however I am 
having trouble registering with Asterisk.

I have to produce an XML Profile to use when registering with a 
registrar. The one I use is...


   
   
   
   
   
   

Anyway I fail to create my profile. In fact Asterisk doesn't even seem 
to get anything at all from my test program.

Has anyone any experience using the Microsoft RTC Client that could 
provide an example please? TIA

Jerry
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[Asterisk-Users] Re: Where are chan_capi bug reports and bugfixes sent?

[Sergio]

b) if you have kernel 2.6, use mISDN kernel patches and chan_mISDN, 
that is, seems, well supported and developed (and works, with a 
compilation flag, with asterisk stable and asterisk head as well):

no mISDN driver for eicon diva cards so it's hard to get it working for me.
Luis Vazquez wrote:
I went to Asterisk bugtracker but I didn't find a capi (or related) 
section. I also looked at Junghanns.net and I didn't find an asterisk 
capi users mailing list or a way to report bugfixes to chan_capi.

I also patched an echo cancel bug over eicon diva and busy signals pakets
So feel free to contact me to share ur patches.
We could find a place share our chan_capi issues. I also mailed the 
chan_capi developer but no reponse from him.

Sergio
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[Asterisk-Users] Re: Chan_Capi initial deadlock

[Sergio]

I had applied the patch and it got much better. Now I only have
problems every two days
   

563 usleep(1);
 

1 is too high you can safely lower it to sleep(1) there's a while 
over there
otherwise it will lock the channel for 10 seconds.
that's code from chan_capi fax patch right?

Sergio
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RE: [Asterisk-Users] Bristuff and incoming call problems

[Remco Barende]

Thanks for the replies to my cry for help! :)
The weird thing is that sometimes tyhe caller does hear the phone ringing, 
and sometimes the line is "dead".

I will try your workaround, will it also work without playing the message 
for an extension? I use it at home and it sounds a bit silly :)

Are these bugs known at Junghanns?
On Fri, 4 Feb 2005, Rob Scott wrote:
I have exactly the same problem.
It was also the same with RC3.
It seems that after a couple of days of working fine, at some point
incoming calls fail but outgoing calls still work (or I would hear user
complaints earlier).
For the lack of ring problem, I do the following in extensions.conf:
[fromexternal]
exten => s,1,Ringing
exten => s,2,Wait,3
exten => s,3,Answer
exten => s,4,Wait,1
exten => s,5,Background(enter-ext-of-person)
So Asterisk singals a ringing tone for 3 seconds so that the caller's
phone has a chance to ring, then answers and plays the 'enter the
extension of the person you want to call' thing while at the same time
listening for digits.
I don't know if this is the right or expected approach but it works for
me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Friday, February 04, 2005 12:16 PM
To: Asterisk Users List
Subject: [Asterisk-Users] Bristuff and incoming call problems
Hi list!
I have some strange problems with Asterisk 1.0.5-BRIstuffed-0.2.0-RC5.
Very regularly asterisk seems to lose connectivity with the ISDN line.
If you try to call in you get the information tone that the number is
not in use. Outbound calls do stil work however. Unloading the modules
and reloading them and start/stop asterisk will solve the problem.
Another problem that occurs regularly : When you make an inbound call to
asterisk the calling party does not get the tone that the phone is
ringing on the receiving end. The line just seems completely dead untill
the phone is picked up and you can hear the other party. Is this an
asterisk / bristuff problem or something for the telco to sort out? Who
should generate the ringing signal to the calling party?
Thanks!!!
Remco
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[Asterisk-Users] T.38 bounty

[Roy Sigurd Karlsbakk]

hi
there are some comments here, 
http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38%20Bounty, 
that people that have earlier offered high bounties for T.38 in 
asterisk. Please add up, so the one that one day manages to add good 
T.38 support may get something back for it :)

roy
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Re: [Asterisk-Users] Q: How to get the preset callerid from a CLID-no-screen E1-PRI

[Frank Sautter]

Peter Svensson wrote:
On Fri, 4 Feb 2005, Frank Sautter wrote:
RDNIS is empty.
So the operator sets an incomplete callerid? Sounds like a 
misconfiguration at the operators end.

Do a "pri intense debug span XXX" on one of the calls and post the log 
of the SETUP to CONNECT_ACK messages.
< Protocol Discriminator: Q.931 (8)  len=55
< Call Ref: len= 2 (reference 4122/0x101A) (Originator)
< Message type: SETUP (5)
< [04 03 80 90 a3]
< Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
<  Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)
<  Ext: 1  User information layer 1: A-Law (35)
< [18 03 a9 83 81]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0
 Are you sure that the RDNIS is empty?
yes, here is a debug output of this call where i put out all variables i 
get
-- Executing NoOp("Zap/1-1", "-user2user  -c2ton 65 -csub  -ani2 00 
-cton 33 -ctns 0 -cani2 0 -cnum 070317145 -id 070317145 -rdnis  -pres 3")


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RE: [Asterisk-Users] Asterisk crashes from time to time

[Hecken, Guido]

> > (There are other debug modes, but not sure I'd use those to catch a
> > production problem. The one's I know about are primarily intended for
> > development debugging. Other folks might contribute hints here.)
> This reeks of a deadlock,
> http://voip-info.org/wiki-Asterisk+deadlock
> see this
> HowTo Debug a DeadLock in Asterisk
> i wrote up eons ago on the wiki
> http://voip-info.org/wiki-Asterisk+debugging

Thanks for your informations, I will "try" to follow the instructions on
debugging asterisk.
Since I'm not a programmer, I think I will get some fun with it ;-)

Guido

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[Asterisk-Users] Re: different IAX ports for different contexts

[Doug Meredith]

"dean collins" <[EMAIL PROTECTED]> wrote:

>My question is this, can you have different ports for different contexts
>within IAX?

If I understand correctly, you want to use different *local* ports for
different contexts.  I don't think you can do this.  You could run
more than one copy of Asterisk on the same machine, and configure each
one to use a different port.
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

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RE: [Asterisk-Users] OT: How to "own" a telephone number?

[David Brodbeck]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]

> Is providing the ability to assign numbers to people instead of to 
> locations really that hard?  Is it really so much easier for Internet 
> domains to do it?  Or is this just an oligarchy at work?  :)

A phone number is more analogous to an IP address than a domain name.  If
you move, you'll have a different ISP, and you won't get to keep your old IP
address.  Your domain name, however, can be pointed to any IP address you
like.  It's that extra layer of indirection -- the domain being resolved to
an IP -- that lets Internet domains be moved so easily.

It's also partly a historical issue.  The phone system is layed out
geographically, because in the days of mechanical switches that was the only
reasonable way to do it.  Each area code represents a certain area of the
country, and each exchange (the first three digits of the local number)
represents a particular central office.  If you're outside the area covered
by that central office, there's no way to get a direct line run to you
(unless you use forwarding, or something like VOIP.)  Billing is based on
this, too.  If people could move numbers around willy-nilly, you'd never
know if you were making a long-distance call or not.

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RE: [Asterisk-Users] Re: different IAX ports for different contexts

[Sandor R. Repas]

Vigasztalj meg, hogy mi jobbak vagyunk.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Meredith
Sent: Friday, February 04, 2005 2:56 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: different IAX ports for different contexts


"dean collins" <[EMAIL PROTECTED]> wrote:

>My question is this, can you have different ports for different 
>contexts within IAX?

If I understand correctly, you want to use different *local* ports for
different contexts.  I don't think you can do this.  You could run more
than one copy of Asterisk on the same machine, and configure each one to
use a different port.
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

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Re: [Asterisk-Users] Where are chan_capi bug reports and bugfixes sent?

[Luis Vazquez]

Thank you for your very instructive response!
I also use chan_capi with Eicon Diva Server, and the problem I had was 
not critic, but related with some  ERROR's  reported by chan_capi when 
the it receives not handled (subcommand=CAPI_CONF command=CAPI_LISTEN)  
messages from the telco.
I fixed this by looking at the handling of this message type in the 
capisuite proyect and adding the code snippet in the proper case entry.
I didn't know of the patch on leviogo.de so I will take a look in case 
the problem I found is fixed in this code.

On the other hand, I am currently using the chan_capi to  handle 6 (3BRI 
x2) pstn lines at the company I'm working in, and the only serious 
problem I'm having is with some phantom (bursts of) dtmf  tones the 
users hear some times in the middle of a conversation.
Have you (or any other capi user) experienced similar problems? There is 
any known fix for this?

Thanks a lot and best regards
Luis
Patrick wrote:
Hi Luis,
I think the best way is to make a diff against the latest chan_capi 
which seems to be version 0.3.5 and email it to kapejod. You may 
probably want to use something like : diff -uNr 
   > 
chan_capi_patch.txt

You can find his email address here: 
http://www.junghanns.net/asterisk/page1.html

Are you aware that there is a patch for chan_capi that fixes some 
issues? You can find it here:
http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
The note about this patch on voip-info.org says:
NOTE: If you are using chan_capi 0.3.5 with asterisk cvs-head (as of 
November 2004) you need to apply a small patch here for it to work. 
This patch also enables sending and receiving of faxes with active 
ISDN cards.

I would appreciate it if you could email me the diff because I use 
chan_capi-0.3.5 with an Eicon Diva Server card at home.

Thanks for finding the bugs.
Regards,
Patrick
Luis Vazquez wrote:
Hello all
I found a bug in the chan_capi driver (really a not implemented 
message handling and then a false error condition) and I guess I have 
wrote a patch to fix it (basically I searched the internet for other 
capi open source implementation an borrowed the code snippet) but I 
don't know where to send the report and bugfix.
I also found some miss-behaviours that I would like to share with 
other asterisk+chan_capi users.
I went to Asterisk bugtracker but I didn't find a capi (or related) 
section. I also looked at Junghanns.net and I didn't find an asterisk 
capi users mailing list or a way to report bugfixes to chan_capi.

Does anybody knows the best way to submit the report so it's 
available to anyone?

Thanks a lot
Luis
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[Asterisk-Users] gsm audio files

[Edgar de Leon]

Hello, anyone knows if exist the audio files in spanish??

or how can i record the voice in gsm extension???

can i play for some announce a random file??

TIA

Edgar
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RE: [Asterisk-Users] ASTCC Apllication

[Karl H. Putz]

-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Daniel Eboa
>Sent: Friday, February 04, 2005 4:50 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [Asterisk-Users] ASTCC Apllication
>
>
>Hello,
>I have some problem using ASTCC application. I've installed the application
and everything works
>well. I've created card numbers, routes trunk and others. When I dial the
desired number (77) in
>my case, I'm prompted to enter my card number. All goes well till I'm
prompted to enter the
>destination number. When I enter a destination number, the system says it's
not a recognized
>number and the call doesn't go through. Can any one help me out with this
issue? Is there a file
>where I can define extensions like in extensions.conf?

Daniel,

It sounds like the problem is the pattern you are trying to use in the
"routes" table.  The pattern should be a REGEX for matching the dialed
number to the appropriate cost for that call.  Take a look at
http://dev.mysql.com/doc/mysql/en/pattern-matching.html for more specifics
on MySQL REGEX matching.

In the US, for example, I would use the pattern: '^1312' to match for calls
to Chicago or '^01149' for calls to Germany.  You can also match for city
codes or especially Cellular "exchanges" in specific countries where the
termination costs are much higher than land-line termination.

The SQL statement in astcc returns all the matched patterns with the
longest, most specific match first and uses only that first match in its
processing.  So you could also use the pattern: '.' to match any dialed
number not already matched as a default BUT BE SURE to set that cost high
enough to cover yourself.

Good luck!


Karl Putz

>
>
>Thanks.
>
>Daniel.




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Re: [Asterisk-Users] MWI with IAX

[Andrew Kohlsmith]

On February 4, 2005 01:23 am, [EMAIL PROTECTED] wrote:
> registration acknoledge.
> - If you have messages, it returns 65535
> - if you have none, it returns 0
> - if you don't have a voicemail box, it returns -1

... it's sending a dword for this?  -1 and 65535 are the same thing in 16 
bits.

-A.
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RE: [Asterisk-Users] ASTCC Application

[Kelly Griffin]

You must have routes and trunks setup for
this to work properly.  If you enter 479-464-8998 and you do not have a
route for 1 or 1479, then it does not know where to send the call.  Make
sure your have your VoIP trunks and calling routes configured properly.



---
Kelly D Griffin
Network Engineer
Tantella Wireless
http://tantella.com
800.636.0306 Voice
479.464.8998 Fax



-Original Message-
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Daniel Eboa
Sent: Friday, February 04, 2005
3:50 AM
To: Asterisk Users
 Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] ASTCC
Apllication

 

Hello,

I have some problem using
ASTCC application. I’ve installed the application and everything works
well. I’ve created card numbers, routes trunk and others. When I dial the
desired number (77) in my case, I’m prompted to enter my card number. All
goes well till I’m prompted to enter the destination number. When I enter
a destination number, the system says it’s not a recognized number and
the call doesn’t go through. Can any one help me out with this issue? Is
there a file where I can define extensions like in extensions.conf? 

 

Thanks.

 

Daniel.

 






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[Asterisk-Users] Call forwarding

[Adam Robins]

I've written a macro that allows users to dynamically change their call
forwarding destination.  The purpose is to set up a "follow me" process
where a user can get calls on their cell, at home, etc., based on the
forwarding number they enter.  Using the CFIM database, I have the setup
portion working great.  Now, I want to actually use that information to
forward a call.  
 
Here is my issue:  The forwarded number saved in CFIM could be another
extension, a local number or an LD number, each of which would be dialed
using a different technology (internal, SIP-provider, Zap, etc.).  I
want to avoid having to check the number and code all of the logic for
each method - because I already have all of this set up in the dialplan
for callers who would have dialed this forwarded number directly.
 
What I would like to do is take the variable containing the number
retrieved from CFIM, place it on the stack as the called number, and
have it reenter the dial plan, similar to the WAITEXTEN command.
 
Any ideas are appreciated!
 
For those interested, here is the "Forwarding Setup" macro:
 
;
; Call forwarding Macro
;
[macro-forwarding]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,DigitTimeout(3)
exten => s,4,ResponseTimeout(10)
exten => s,5,Read(fwext,fw-extension,2); ask
extension (2 digits)
exten => s,6,Authenticate(/etc/asterisk/authFWD)   ; only
authorized individuals
exten => s,7,Playback(fw-extension); repeat back
extension
exten => s,8,SayNumber(${fwext},f)
exten => s,9,DBget(fwnum=CFIM/${fwext}); check if
already forwarded
 
; ext is forwarded
exten => s,10,Playback(fw-is-forwarded-to) ; play
forwarded number from database
exten => s,11,SayDigits(${fwnum})
exten => s,12,Read(resp,fw-cancel-1-change-2,1); 1 to cancel
fwd, 2 to change #
exten => s,13,GotoIf($[${resp} = 1]?17:14) ; 1 entered,
goto delete
exten => s,14,GotoIf($[${resp} = 2]?111:15); 2 entered,
jump to change number
exten => s,15,Playback(fw-invalid-response); invalid
response, loop back
exten => s,16,Goto(s,12)
exten => s,17,DBdel(CFIM/${fwext}) ; delete
entry from database
exten => s,18,Playback(fw-call-fwd-canceled)   ; give status
& end call
exten => s,19,Playback(fw-goodbye)
exten => s,20,Hangup
 
; ext is not forwarded
exten => s,110,Playback(fw-is-not-currently-forwarded) ; say number
is not forwarded
exten => s,111,Playback(fw-enter-new-forwarding-number); ask for new
number
exten => s,112,Read(fwnum,fw-press-pound-when-finished); accept new
number, since variable length, ask for #
exten => s,113,GotoIf($[${LEN(${fwnum})} < 2]?114:116) ; if len < 2
then bad number
exten => s,114,Playback(fw-invalid-response)
exten => s,115,Goto(s,111)
exten => s,116,Playback(fw-you-entered); repeat back
number
exten => s,117,SayDigits(${fwnum})
exten => s,118,Read(resp,fw-if-corr-press-1-otherwise-2,1) ; confirm 1
if correct, 2 if not
exten => s,119,GotoIf($[${resp} = 1]?120:111)  ; if 1,
proceed and update db, else loop back
exten => s,120,DBdel(CFIM/${fwext}); delete db
exten => s,121,DBput(CFIM/${fwext}=${fwnum})   ; add new db
entry
exten => s,122,Playback(fw-ext-is-forwarded)   ; give status
& end call
exten => s,123,Playback(fw-goodbye)
exten => s,124,Hangup

 

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Re: [Asterisk-Users] why asterisk and ser

[Matthew Boehm]

http://drmac.homeunix.net/images/ser_asterisk.jpg

- Original Message - 
From: "Altus Snyman" <[EMAIL PROTECTED]>
To: "asterisk" 
Sent: Friday, February 04, 2005 2:35 AM
Subject: [Asterisk-Users] why asterisk and ser


> Good day all
> Why would u use asterisk and ser together and what is the big
> difference?
> Thanks
> altus
> 
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Re: [Asterisk-Users] T.38 bounty

[Matthew Boehm]

Since t38 is seperate from SIP, you basically need a chan_t38 right?

-Matthew

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

Sent: Friday, February 04, 2005 7:45 AM
Subject: [Asterisk-Users] T.38 bounty


> hi
>
> there are some comments here,
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38%20Bounty,
> that people that have earlier offered high bounties for T.38 in
> asterisk. Please add up, so the one that one day manages to add good
> T.38 support may get something back for it :)
>
> roy
>
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Re: [Asterisk-Users] T.38 bounty

[Robert Jackson]

Matthew Boehm wrote:
Since t38 is seperate from SIP, you basically need a chan_t38 right?
-Matthew
That is my understanding.
Robert Jackson
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RE: [Asterisk-Users] ASTCC Apllication

[Karl H. Putz]

>The SQL statement in astcc returns all the matched patterns with the
>longest, most specific match first and uses only that first match in its
>processing.  So you could also use the pattern: '.' to match any dialed
>number not already matched as a default BUT BE SURE to set that cost high
>enough to cover yourself.
>
>
Sorry all for replying to my own posting but I realized that my advise on
the above "default" pattern was a BAD idea.  The main issue is that it does
not allow for blocking any routes especially to 1-900 or other pay
exchanges.


Karl Putz



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[Asterisk-Users] How to Create customized audio file to use with ASTCC??

[Daniel Eboa]

Hello all,

Can anyone help me out with this issue ?? I got
ASTCC running, but the audios doesn’t match my needs (currency, etc.). is
there any way to create my own audios and replace the current one??

 

Thanks.

 

Daniel.

 

 






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RE: [Asterisk-Users] ASTCC Apllication

[Daniel Eboa]

Thanks a lot. Now I understand and it's working.

Regards.

Daniel.



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl H. Putz
Sent: vendredi 4 février 2005 15:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] ASTCC Apllication

-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Daniel Eboa
>Sent: Friday, February 04, 2005 4:50 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [Asterisk-Users] ASTCC Apllication
>
>
>Hello,
>I have some problem using ASTCC application. I've installed the application
and everything works
>well. I've created card numbers, routes trunk and others. When I dial the
desired number (77) in
>my case, I'm prompted to enter my card number. All goes well till I'm
prompted to enter the
>destination number. When I enter a destination number, the system says it's
not a recognized
>number and the call doesn't go through. Can any one help me out with this
issue? Is there a file
>where I can define extensions like in extensions.conf?

Daniel,

It sounds like the problem is the pattern you are trying to use in the
"routes" table.  The pattern should be a REGEX for matching the dialed
number to the appropriate cost for that call.  Take a look at
http://dev.mysql.com/doc/mysql/en/pattern-matching.html for more specifics
on MySQL REGEX matching.

In the US, for example, I would use the pattern: '^1312' to match for calls
to Chicago or '^01149' for calls to Germany.  You can also match for city
codes or especially Cellular "exchanges" in specific countries where the
termination costs are much higher than land-line termination.

The SQL statement in astcc returns all the matched patterns with the
longest, most specific match first and uses only that first match in its
processing.  So you could also use the pattern: '.' to match any dialed
number not already matched as a default BUT BE SURE to set that cost high
enough to cover yourself.

Good luck!


Karl Putz

>
>
>Thanks.
>
>Daniel.




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Re: [Asterisk-Users] T.38 bounty

[Steve Underwood]

Do we have a chan_rtp? T.38 is to FAX as RTP is to audio. In fact, the 
latest T.38 spec. now at last does what the original should have done, 
and specifies a T.38 over RTP option.

chan_h323 and chan_sip need to be modified to accomodate T.38.
Regards,
Steve
Matthew Boehm wrote:
Since t38 is seperate from SIP, you basically need a chan_t38 right?
-Matthew
- Original Message - 
From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Friday, February 04, 2005 7:45 AM
Subject: [Asterisk-Users] T.38 bounty

 

hi
there are some comments here,
http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38%20Bounty,
that people that have earlier offered high bounties for T.38 in
asterisk. Please add up, so the one that one day manages to add good
T.38 support may get something back for it :)
roy
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Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

[Mark Eissler]

I'm quite happy with iax.cc (Sixtel). I don't have DIDs with them but 
use them for outbound and have no complaints. Whenever I've contacted 
support I've received a reply the same day. Perhaps they prioritize 
their support email based on whether or not you have an account with 
them? They match up inbound support messages with your email address to 
check if its associated with an account.

Anyhow, I might obtain DIDs from them sometime soon if Voicepulse 
doesn't get their act together and gets inbound DTMF working properly 
over IAX. It's been almost two months since I first reported the 
problem and they just reply that they have to perform several software 
upgrades over a 2-4 week period. Jeez. I can't recommend Voicepulse 
either right now as I have no intention of switching back to SIP (from 
IAX) for termination.

The funny thing is that I have ended up using FWD the most because of 
their toll free gateway. I'm constantly amazed at the clarity of those 
calls. But then again I'm constantly amazed at the clarity of any of my 
calls through Asterisk vs. say my residential phone services via Vonage 
and Broadvox.

-mark
On Feb 3, 2005, at 10:38 PM, Brian Dingman wrote:
I took them up on their offer for a refund. IMHO they shouldn't offer
* service at all. Even outgoing calls aren't handled properly. Lots of
making progress - no answer results.
Others have suggested iax.cc. However, they haven't repsonded to my
email (over 2 days now) and I can't get through to them over the phone
or IM. Not very promising.
All I want is a toll free DID that works on * and isn't too expensive.
Any suggestions for a provider? I don't even care if it can be ported
away!
On Thu, 3 Feb 2005 10:12:02 -0500, Mark Eissler <[EMAIL PROTECTED]> 
wrote:
Based on the support and management responses that have been posted to
this list it doesn't sound to me (at least) like LiveVoip really wants
business from * users anyhow. They blame a lot of problems on * and 
are
quick to offer a refund. There are plenty of DID providers that are
more asterisk-friendly.

-mark
On Feb 2, 2005, at 1:25 PM, Brian Dingman wrote:
Finally got a reply from LV support. Not what I was hoping for.
Hopefully they will file a bug with Digium since they investigated 
the
issue not holding my breath.

Since this is such basic * functionality that they can't seem to
accomplish I would think twice before aquiring DID's from them.
 LiveVoip Support
Our people have looked into this matter over the past few days. They
tell me
that it is a problem with Asterisk.
We are not going to be able to help you with this. If you would like 
a
refund so that you can migrate to another
service provider we will be happy to do so. With each rev. of 
Asterisk
more
and more improvements are made.
At some point these issues may resolve but, for the time being it is
not a
problem we can help you with.


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--
Mark Eissler, [EMAIL PROTECTED]
Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com
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[Asterisk-Users] X-lite to Cisco ATA - no RTP

[Keith Burns]

Title: X-lite to Cisco ATA - no RTP






Hi there,

I have X-lite and a Cisco ATA on the same hub (i.e. no NAT, no ACLs) as my Asterisk box.

Ethereal shows normal SIP signaling when I call from X-lite to the ATA.

Ethereal also shows RTP is passed from X-lite to Asterisk, and RTP is passed from the ATA to Asterisk, but no RTP from Asterisk to either device.

(Note that Ethereal does show the SIP signaling packets originating from Asterisk, so nothing funky with my Ethereal filter either)

Has anyone run into anything similar? Any pointers? I set up all the extensions using AMP.

If you need specific configs, I am happy to provide.

Cheers

Keith.


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RE: [Asterisk-Users] Bristuff and incoming call problems

[Rob Scott]

Sure you can put whatever you like in the answering config.
That is just mine because after dialing an incoming number the caller
dials an extension number that Asterisk uses to complete the call.

Junhanns does post on this group occasionally so I guess he watches it
but I haven't so far seen any useful messages on how to solve this
problem.

A behaviour I did notice once was that if you dialed in and then waited
about 4 seconds then you did eventually get a ringing tone, which
suggests that it was connecting but waiting for some timeout before
following the context code; i.e. it was working but had a timeout or was
working extremely slowly. Next time to thing behaves badly I will check
if this is still the behaviour.

If it doesn't get fixed then I will probably use a script that stops
asterisk, reloads the modules, and starts asterisk again and runs it at
say 5am every morning. Not ideal but what can you do? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Friday, February 04, 2005 2:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Bristuff and incoming call problems

Thanks for the replies to my cry for help! :)

The weird thing is that sometimes tyhe caller does hear the phone
ringing, and sometimes the line is "dead".

I will try your workaround, will it also work without playing the
message for an extension? I use it at home and it sounds a bit silly :)

Are these bugs known at Junghanns?


On Fri, 4 Feb 2005, Rob Scott wrote:

> I have exactly the same problem.
> It was also the same with RC3.
> It seems that after a couple of days of working fine, at some point 
> incoming calls fail but outgoing calls still work (or I would hear 
> user complaints earlier).
>
> For the lack of ring problem, I do the following in extensions.conf:
>
> [fromexternal]
> exten => s,1,Ringing
> exten => s,2,Wait,3
> exten => s,3,Answer
> exten => s,4,Wait,1
> exten => s,5,Background(enter-ext-of-person)
>
> So Asterisk singals a ringing tone for 3 seconds so that the caller's 
> phone has a chance to ring, then answers and plays the 'enter the 
> extension of the person you want to call' thing while at the same time

> listening for digits.
>
> I don't know if this is the right or expected approach but it works 
> for me.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Remco 
> Barende
> Sent: Friday, February 04, 2005 12:16 PM
> To: Asterisk Users List
> Subject: [Asterisk-Users] Bristuff and incoming call problems
>
> Hi list!
>
> I have some strange problems with Asterisk 1.0.5-BRIstuffed-0.2.0-RC5.
>
> Very regularly asterisk seems to lose connectivity with the ISDN line.
> If you try to call in you get the information tone that the number is 
> not in use. Outbound calls do stil work however. Unloading the modules

> and reloading them and start/stop asterisk will solve the problem.
>
> Another problem that occurs regularly : When you make an inbound call 
> to asterisk the calling party does not get the tone that the phone is 
> ringing on the receiving end. The line just seems completely dead 
> untill the phone is picked up and you can hear the other party. Is 
> this an asterisk / bristuff problem or something for the telco to sort

> out? Who should generate the ringing signal to the calling party?
>
> Thanks!!!
> Remco
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Re: [Asterisk-Users] Call forwarding

[Kevin P. Fleming]

Adam Robins wrote:
What I would like to do is take the variable containing the number
retrieved from CFIM, place it on the stack as the called number, and
have it reenter the dial plan, similar to the WAITEXTEN command.
Easy-peasy. That's what Local channels are for.
If you have retrieved the CFIM value into a variable called CFIM, and 
the user's phone would normally dial via a context called 
"customer-dial", then:

exten => ...,...,Dial(Local/[EMAIL PROTECTED])
This will process the call exactly the same way as if the phone user had 
dialed that number from their phone.
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[Asterisk-Users] No Playback() when Digicom TE110P enabled

[Gareth Blades]

I have a Digicom TE110p card installed in our exchange. I have compiled
and installed libpri, zaptel and recompiled and installed asterisk.
I have configured udev as I am running Fedora Core 3.

The problem that I have is that when zaptel is not running everything
works fine. However when I start zaptel (service start zaptel) then I
can make normal calls ok but the 'Playback()' function,voicemail etc...
dont work and all I get on the IAX client is silence.

In /etc/sysconfig/zaptel I have:-
TELEPHONY=yes
MODULES="$MODULES wcte11xp" # TE110P - Single Span T1/E1 Card

Any idea what is going wrong?

/etc/asterisk/zapata.conf has not been altered (card is not currently
connected to anything).

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[Asterisk-Users] echo's + cheap phones

[Giovanni Powell]

Is it possible that cheap phones (Budgetone) cause echo's? 

I had a digium X100p and i managed to get rid of all the echo problems
i was having. Recently i got a Voicetronix Openswitch12, and getting
terrible noises when i use an IP phone (budgetone) to call analog
phones or PSTN.

I have tried all the possible things (rxgain, txgain, echocancel, i
even changed the codec to g711)

Is it possible that the IP phones itself is causing this.
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[Asterisk-Users] HP ProLiant server for Asterisk

[Dana Olson]

I'm looking at ordering a server from HP. I checked around on Google
and found in the Wiki that the ProLiant DL380 is supposed to be known
to work with *.

I'm going to get a price quote on the following setup:

HP ProLiant DL380 G4 Server w/ the following options:
Intel Xeon 3.20GHz/1MB
2GB REG PC2-3200 (2 X 1GB)
HP ProLiant Battery Backed Write Cache Enabler for SA6i
RAID 1 drive set
HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive
HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive
Hot Plug Redundant Power Supply Module
HP Redundant Fan Option Kit (3 fans)
1.44MB Floppy Disk Drive
Slimline 24X CD-ROM

Can anyone comment any further on this system? Do you think I would
make a wise choice to order this? I will be putting a 4-port Digium TE
card in it.

--
Dana
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[Asterisk-Users] manager api - Async:True?

[Ken Godee]

Asterisk 1.0.3 / TE410 / ISDN/PRI Zap channels
As I understand it using the "Async: True" in an
originate action is supposed do a "Fast Originate"
"originate a call from a channel to an extension without waiting
for call to complete".
I'm finding no difference using Async or not, calls
always wait for completion before connecting to extensions.
Don't know if I'm missing something or if this
just doesn't work when using ISDN channels and
the ISDN signaling is overriding completion?
I'm trying to do exactly what this feature is
meant to do, connect channel->exten before completion.
Any ideas?
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RE: [Asterisk-Users] HP ProLiant server for Asterisk

[Edge Bisset]

Hi Dana

The DL380 G4 may not be the best choice; there has been a lot of talk on
the forum about problems with the DL380 G4 & the TE410P. 

See this thread:
http://lists.digium.com/pipermail/asterisk-users/2005-January/081544.htm
l

Cheers,
Edge.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dana Olson
Sent: Friday, February 04, 2005 5:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] HP ProLiant server for Asterisk


I'm looking at ordering a server from HP. I checked around on Google and
found in the Wiki that the ProLiant DL380 is supposed to be known to
work with *.

I'm going to get a price quote on the following setup:

HP ProLiant DL380 G4 Server w/ the following options:
Intel Xeon 3.20GHz/1MB
2GB REG PC2-3200 (2 X 1GB)
HP ProLiant Battery Backed Write Cache Enabler for SA6i
RAID 1 drive set
HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive
HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive
Hot Plug Redundant Power Supply Module HP Redundant Fan Option Kit (3
fans) 1.44MB Floppy Disk Drive Slimline 24X CD-ROM

Can anyone comment any further on this system? Do you think I would make
a wise choice to order this? I will be putting a 4-port Digium TE card
in it.

--
Dana
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Re: [Asterisk-Users] echo's + cheap phones

[Jon Radon]

My Budgetone had a really bad echo no matter what.  Switching to my
Sipura or Polycom the problem was no wear near as bad.


On Fri, 4 Feb 2005 10:12:41 -0500, Giovanni Powell
<[EMAIL PROTECTED]> wrote:
> Is it possible that cheap phones (Budgetone) cause echo's?
> 
> I had a digium X100p and i managed to get rid of all the echo problems
> i was having. Recently i got a Voicetronix Openswitch12, and getting
> terrible noises when i use an IP phone (budgetone) to call analog
> phones or PSTN.
> 
> I have tried all the possible things (rxgain, txgain, echocancel, i
> even changed the codec to g711)
> 
> Is it possible that the IP phones itself is causing this.
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 


-- 
Is it something someone said, was it something someone said?
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Re: [Asterisk-Users] Re: Chan_Capi initial deadlock

[Carl Sempla]

On Friday, 04 February, 2005 14:16 : Sergio <[EMAIL PROTECTED]> wrote:

>>> I had applied the patch and it got much better. Now I only have
>>> problems every two days
>>>
>>>
>>>
>> 563 usleep(1);
>>
>>
>>
> 1 is too high you can safely lower it to sleep(1) there's a while
> over there
> otherwise it will lock the channel for 10 seconds.
> that's code from chan_capi fax patch right?

Nop, it's in the original code. Don't forget that usleep use microseconds,
not milli, so this line doesn't wait 10s.

-- 
Carl

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Re: [Asterisk-Users] Q: How to get the preset callerid from a CLID-no-screen E1-PRI

[Peter Svensson]

On Fri, 4 Feb 2005, Frank Sautter wrote:

> < Message type: SETUP (5)
[snip]
> < [6c 0c 21 80 31 37 32 39 38 37 36 35 34 33]
> < Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI: 
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> <   Presentation: Presentation permitted, user 
> number not screened (0) '1729876543' ]
> < [6c 0a 21 83 37 30 33 31 37 31 34 35]
> < Calling Number (len=12) [ Ext: 0  TON: National Number (2)  NPI: 
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> <   Presentation: Presentation allowed of 
> network provided number (3) '70317145' ]
[snip]
> 
> as you can see there are two calling numbers sent: 1729876543 (CLI-no 
> screen - which should be the callerid) and a second one 70317145 
> (Network provided number).

This is rather weird? What network do you receive this from? Neither 
ITU q.931 nor ETSI EN 300 403-1 (EiroISDN definition) lists the "Calling 
Number" IE among those that may be repeated. 

I am quite certain that libpri does not handle this. The last one will 
overwrite the earlier calling numbers. Some hacking of libpri is probably 
needed to handle this. To handle it cleanly a more complex interface 
between chan_zap and libpri may be needed.

Peter


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Re: [Asterisk-Users] HP ProLiant server for Asterisk

[Dana Olson]

Thanks, I've been reading that. I'm pretty sure that HP is our only
option here, so maybe I'll try getting quotes on the other models.
--
Dana


On Fri, 4 Feb 2005 17:31:26 +0200, Edge Bisset <[EMAIL PROTECTED]> wrote:
> Hi Dana
> 
> The DL380 G4 may not be the best choice; there has been a lot of talk on
> the forum about problems with the DL380 G4 & the TE410P.
> 
> See this thread:
> http://lists.digium.com/pipermail/asterisk-users/2005-January/081544.htm
> l
> 
> Cheers,
> Edge.
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Dana Olson
> Sent: Friday, February 04, 2005 5:20 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] HP ProLiant server for Asterisk
> 
> I'm looking at ordering a server from HP. I checked around on Google and
> found in the Wiki that the ProLiant DL380 is supposed to be known to
> work with *.
> 
> I'm going to get a price quote on the following setup:
> 
> HP ProLiant DL380 G4 Server w/ the following options:
> Intel Xeon 3.20GHz/1MB
> 2GB REG PC2-3200 (2 X 1GB)
> HP ProLiant Battery Backed Write Cache Enabler for SA6i
> RAID 1 drive set
> HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive
> HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive
> Hot Plug Redundant Power Supply Module HP Redundant Fan Option Kit (3
> fans) 1.44MB Floppy Disk Drive Slimline 24X CD-ROM
> 
> Can anyone comment any further on this system? Do you think I would make
> a wise choice to order this? I will be putting a 4-port Digium TE card
> in it.
> 
> --
> Dana
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[Asterisk-Users] Specify a codec in dial plan?

[Brian McCrary]

Hello,

Earlier I was having problems getting g729 pass-thru to work correctly.
I was able to fix this by putting allow=g729 as the first allow
statement directly under the configuration of the user.  I also put
allow=ulaw in there below it.  It seems the second "allow" statement is
totally ignored, as asterisk only negotiates codecs based on the first
statement.

It looks as if asterisk only does codec priority based on the [general]
part of the config, but I haven't seen it makes any difference as far as
negotiation goes.  The first codec listed is the only codec allowed,
period. 

So, therefore, I am wondering if there is a way to force a codec to be
used in a dial plan, so when voicemail is dialed, for example, I can
force the ulaw codec instead of g729.  Both codec are already listen
under the user part of the config.  Yes, I know g729 licenses are cheap,
but they aren't available for my platform.

I've seen this question pop up from time to time but never have seen any
replies to it, so if you have a setup using two codecs I'd appreciate
hearing from you!

Thanks,

Brian
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[Asterisk-Users] Digitnetworks Analog Card on Dell 2800 with Asterisk

[Steve Blair]

Hello:
  I had been using a Digit Networks analog Card on Dell 1850 running
RH for about a year. I recently purchased a Dell PE 2800 with RH. When
I install the analog interface card the 2800 won't boot. There isn't any
diagnostic message either. Does anyone have experience with this?
Any idea how to make this card work in the 2800?
Thanks,Steve
--
 
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104  

voice: 215-573-8396 

  215-746-8001
fax: 215-898-9348

sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] OT: How to "own" a telephone number?

[Andrew Thompson]

David Brodbeck wrote:
Is providing the ability to assign numbers to people instead of to 
locations really that hard?  Is it really so much easier for Internet 
domains to do it?  Or is this just an oligarchy at work?  :)

A phone number is more analogous to an IP address than a domain name.  If
you move, you'll have a different ISP, and you won't get to keep your old IP
address.  
For most end users, this is a correct statement.
Hosting companies, and other businesses with significant Internet 
presences can request a block of IPs directly from ARIN or their 
regional equivalent. Once assigned, the company can buy Internet access 
from any and multiple carriers who all point the inbound traffic to the 
companies assigned IPs. Among other things, this allows for 
semi-intelligent recovery if a carrier's network goes offline, by 
routing packets through another carrier's network.

Billing is based on
this, too.  If people could move numbers around willy-nilly, you'd never
know if you were making a long-distance call or not.
This is very true, of our current area coding schema. I don't know how 
it could be made fair without perhaps a nationwide maximum charge for 
dialing 700 numbers or something like that. But it still ought to be a 
local call(very, very cheap or free) next door to my neighbors.

--
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
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[Asterisk-Users] IAX2 register Refresh

[Liaan vd Merwe]

Hi all
I been looking into the whole code strugture of 
chan_iax and i see there is a option to specify the refresh rate of 
registrations: But there is no code to actually load this from the config 
file
thus i changed the setting in chan_so.h, and 
recompiled. But still my refresh rate is 60 sec.
 
I need to get this down to 15 sec (nat /pat 
firewall issue)
 
any ideas?
thanks
Liaan
 

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RE: [Asterisk-Users] Q: How to get the preset callerid from aCLID-no-screen E1-PRI

[Paul Brock]

>This is rather weird? What network do you receive this from? Neither 
>ITU q.931 nor ETSI EN 300 403-1 (EiroISDN definition) lists the "Calling 
>Number" IE among those that may be repeated. 

q.931 (and q.931e) traces include both "called number" and "calling number",
on all Uk variants. Due to the European nature of q931, I would guess this
to be the same across europe...

Paul

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RE: [Asterisk-Users] gsm audio files

[asterisk]

You have to use the app record attached to some extension, save your voice
to a file with the desireable format and you then can hear your voice with
another extension and if you like it, yo can rename your file in the dir
/var/lib/asterisk/souds and use it as any other of the preinstalled asterisk
sounds!!!

Have fun!

exten => 205,1,Wait(2)
exten => 205,2,Record(/tmp/asterisk-recording:gsm)
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/asterisk-recording)
exten => 205,5,Wait(2)
exten => 205,6,Hangup

exten => 206,1,Playback(/tmp/asterisk-recording)
exten => 206,2,Hangup()

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Edgar de Leon
Enviado el: Viernes, 04 de Febrero de 2005 07:44 a.m.
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] gsm audio files

Hello, anyone knows if exist the audio files in spanish??

or how can i record the voice in gsm extension???

can i play for some announce a random file??

TIA

Edgar
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[Asterisk-Users] PCMCIA card

[Calin Serbanescu]

Hello,

Are there any T1/E1 PCMCIA cards available on the market supported by
zaptel drivers and asterisk ? 

I need to make some demos at my clients with asterisk and it's a pain to
move around with a midi-tower computer just for that.

Thanks,
Calin.

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Re: [Asterisk-Users] IAX2 register Refresh

[Andrew Kohlsmith]

On February 4, 2005 10:52 am, Liaan vd Merwe wrote:
> I been looking into the whole code strugture of chan_iax and i see there is
> a option to specify the refresh rate of registrations: But there is no code
> to actually load this from the config file thus i changed the setting in
> chan_so.h, and recompiled. But still my refresh rate is 60 sec.
>
> I need to get this down to 15 sec (nat /pat firewall issue)

Fix the NAT/PAT device?  15s timeout is insane, most things have many, many 
minutes for timeout.

-A.
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Re: [Asterisk-Users] PCMCIA card

[Bob Goddard]

On Friday 04 February 2005 16:05, Calin Serbanescu wrote:
> Hello,
>
> Are there any T1/E1 PCMCIA cards available on the market supported by
> zaptel drivers and asterisk ?
>
> I need to make some demos at my clients with asterisk and it's a pain to
> move around with a midi-tower computer just for that.

PCMCIA cards are 16-bit, I doubt whether you will find one which works.

Use a mini-ITX instead.


B
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Re: [Asterisk-Users] New Asterisk user with a goal

[Andrew Thompson]

Ryan Coates wrote:
at present I am trying to test with some software phones and an
asterisk box on a virtual network (no nat, no firewalls) to see if we
can get anything working (client 1 calling client 2), before we splash
out a fair ammount on some decent IP Phones, but am not having much
joy
if anyone could give me some help/advice on the matter I would be greatful
if you need any more details please do not hesitate to ask, ill try to
answer whatever I can
Why don't you tell us exactly what you're trying, and what is and is not 
working? Are you getting error messages, or does your machine fall over 
and vomit all over itself?

--
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
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Re: [Asterisk-Users] T.38 bounty

[Matthew Boehm]

So..need to modify the chan_sip and chan_h323 to recognize the T38 audio?

That makes it sound more like T38 is a codec like 711/729 rather than a
protocol like SIP/H323; is that right?

The feature page for the new Sipura-2100 says this:

  "Fax - G.711 Pass-Through or Real Time Fax over IP via T.38"

So is that actually read as "Real Time Fax over IP via T.38 using SIP"?

This is where my confusion starts; I don't know if T38 is a protocol or a
codec...

-Matthew

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

Sent: Friday, February 04, 2005 8:41 AM
Subject: Re: [Asterisk-Users] T.38 bounty


> Do we have a chan_rtp? T.38 is to FAX as RTP is to audio. In fact, the
> latest T.38 spec. now at last does what the original should have done,
> and specifies a T.38 over RTP option.
>
> chan_h323 and chan_sip need to be modified to accomodate T.38.
>
> Regards,
> Steve
>
>
> Matthew Boehm wrote:
>
> >Since t38 is seperate from SIP, you basically need a chan_t38 right?
> >
> >-Matthew
> >
> >- Original Message - 
> >From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
> >To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >
> >Sent: Friday, February 04, 2005 7:45 AM
> >Subject: [Asterisk-Users] T.38 bounty
> >
> >
> >
> >
> >>hi
> >>
> >>there are some comments here,
> >>http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38%20Bounty,
> >>that people that have earlier offered high bounties for T.38 in
> >>asterisk. Please add up, so the one that one day manages to add good
> >>T.38 support may get something back for it :)
> >>
> >>roy
> >>
> >>___
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> >>
> >
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> >
> >
>
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[Asterisk-Users] zapata.conf ERROR?????? please help

[asterisk]

Hi asterisk GURUs I have an issue with a call
forwarding setup, I have the next zapata conf:

 

-zapata.conf-

 

[channels]

context=incoming

 

signalling=fxs_ks

usecallerid=yes

hidecallerid=no

echocancel=yes

echocancelwhenbridged=yes

rxgain=0.0

txgain=0.0

 

channel=1

 

-zapata.conf-

 

And in extensions.cof I have a simple forward setup
where $CASA define all my voip extension installed at my home, and the problem
is that if I pick up the call in the PSTN connected phone, each 10 seconds asterisk
keeps ringing and ringing the $CASA extensions, and even with an outgoing call
if I took the PSTN phone to make an outgoing call, asterisk keeps ringing and
ringing the $CASA extensions, this is really annoying, I don’t know if I
have a miscofigurations or is a software/hardware problem. I’m running
asterisk v.1.0.3 and I have a X100P card.

 

-extensions.conf-

 

[incoming]

exten => s,1,Dial(${CASA},10)

 

-extensions.conf-

 

Thanks in advance!






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Re: [Asterisk-Users] PCMCIA card

[Andrew Kohlsmith]

On February 4, 2005 11:10 am, Bob Goddard wrote:
> PCMCIA cards are 16-bit, I doubt whether you will find one which works.

CardBus is 32-bit and many PCI devices fit into a PCMICA slot.  Unfortunately 
I agree with you on the other points -- either use a PCMCIA-PCI expander 
(pricey) or a small mini-itx or shuttle-type PC.

Also remember that the original tormentia T1 card *was* ISA.  :-)

-A.
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Re: [Asterisk-Users] Good 800 Number provider

[Michael Graves]

On Thu, 03 Feb 2005 11:45:33 -0800, Ed Greenberg wrote:

>
>
>--On Thursday, February 03, 2005 2:20 PM -0500 Andrew Thompson 
><[EMAIL PROTECTED]> wrote:
>
>> What you are seeing with these bargain providers is they have a clause in
>> their contract that says they own the number, not you. It is a lock, and
>> it ought to be illegal, but sadly, it's probably not. If you choose one
>> of these companies that doesn't allow you to "port" or "resporg" your
>> number out, that's your decision.  Just ask when you get the toll-free if
>> they do allow resporg's out, and have them show you the wording in their
>> contract that confirms it.
>
>
>Are there any recommendations for high quality providers that will assign a 
>Toll Free number and deliver it over VOIP, while still allowing port-out if 
>the service doesn't work out?
>

I've been very pleased with Clearpath (www.clearpath1.com) Been using
them for about a year.

Michael
--
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Sr. Product Specialist  www.pixelpower.com
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[Asterisk-Users] Callerid problems with 1.0.5

[BiGReDSaL]

I just upgraded my two asterisk boxes to 1.0.5 stable and I've noticed
that callerid is not functioning properly. My setup looks like this:

SIP Phone <--> SER <--> Asterisk <--> Asterisk <---> PSTN

No iax is being used at this time.

The problem can be best described by the following scenarios:

1.) SIP to PSTN call: When a SIP phone calls a PSTN bound number, the
callerid displayed on the PSTN phone is the number of the PSTN phone
instead of the SIP phone's number.

2.) PSTN to SIP call: When a PSTN phone calls a SIP Phone number, the
callerid displayed on the SIP phone is the number of the SIP phone instead
of the PSTN phone's number.
For both scenarios - ${CALLERID}, ${EXTEN}, and ${CALLERIDNUM} all have the number of the called phone for ZAP to SIP, SIP to ZAP, and SIP to SIP. I have noticed that explicitly declaring SetCallerID(${CALLERID}) before my dial seems to fixe this issue for only the ZAP to SIP piece. In the next Asterisk where a SIP to SIP relay is occurring ${CALLERID} ends up matchign ${EXTEN} again.
This is causing some havoc with users calling cell phone from SIP phones.
Some users are being dumped into certain company's cell phone voicemail
because the callerid is keyed to the called phone's number.

Has anyone else experienced this problem with 1.0.5 stable? I checked the
bugs.digum.com page and found a similar bug with regard to the call being
delivered to the manager API. Also, I searched the configs and I did not
see any new settings related to callerid. If this is a simple
configuration change introduced into version 1.0.5, any info would be
greatly appreciated.

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RE: [Asterisk-Users] New Asterisk user with a goal

[Keith Burns]

Hi Ryan,

Assuming you have looked at the WIKI at www.voip-info.org, there is some
good info there. Not sure of your background, mine is mainly VoIP,
telephony and networking, and definitely not strong on Linux, so if you
would like to email me directly offline ([EMAIL PROTECTED]) with
some of your issues, as long as they are VoIP and IP, I may be able to
help.

On the Linux stuff, I can only share with you my experience which is
limited at best :)



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ryan Coates
> Sent: Friday, February 04, 2005 2:09 AM
> To: Asterisk-Users@lists.digium.com
> Subject: [Asterisk-Users] New Asterisk user with a goal
> 
> Hi All, I am rather new to the asterisk world, and new to VoIP in
> general, my question seems rather simple compared to some of the
> topics under discussion here :)
> 
> I have done quite a bit of reading and fiddling trying to get a system
> set up, to no avail yet
> basically myself and a friend (both behind NAT and Firewalls, but able
> to set up the firewall rules/port mappings ourself) are interesting in
> setting up a PBX each, initially just for IP based calls over the net,
> so that we can call each other for free
> eventually we will look at plugging this into the POTS system and
> repacing all our phones with  IP phones and rouitng the calls
> appropriately depending on destination
> 
> at present I am trying to test with some software phones and an
> asterisk box on a virtual network (no nat, no firewalls) to see if we
> can get anything working (client 1 calling client 2), before we splash
> out a fair ammount on some decent IP Phones, but am not having much
> joy
> 
> if anyone could give me some help/advice on the matter I would be
greatful
> 
> if you need any more details please do not hesitate to ask, ill try to
> answer whatever I can
> --
> Regards,
> 
> Ryan "Phoenix" Coates
> [EMAIL PROTECTED]
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Re: [Asterisk-Users] T.38 bounty

[Steve Underwood]

SIP and H.323 are signalling. They need to know how to signal the use of 
T.38, so they must be modified.

The T.38 spec defines a streaming protocol over UDP, equivalent to RTP. 
The latest version also allows the use of RTP. T.38 can also use a 
non-quite-streaming TCP transport.

T.38 also defines a codec like way to transport the FAX information as 
digital data (i.e. not sending the modem tones as digitised audio). 
However, unlike most codecs the two directions are not independent. T.38 
processing needs to incorporate much of a T.30 engine, and deal with the 
information in both directions.

Regards,
Steve
Matthew Boehm wrote:
So..need to modify the chan_sip and chan_h323 to recognize the T38 audio?
That makes it sound more like T38 is a codec like 711/729 rather than a
protocol like SIP/H323; is that right?
The feature page for the new Sipura-2100 says this:
 "Fax - G.711 Pass-Through or Real Time Fax over IP via T.38"
So is that actually read as "Real Time Fax over IP via T.38 using SIP"?
This is where my confusion starts; I don't know if T38 is a protocol or a
codec...
-Matthew
- Original Message - 
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Friday, February 04, 2005 8:41 AM
Subject: Re: [Asterisk-Users] T.38 bounty

 

Do we have a chan_rtp? T.38 is to FAX as RTP is to audio. In fact, the
latest T.38 spec. now at last does what the original should have done,
and specifies a T.38 over RTP option.
chan_h323 and chan_sip need to be modified to accomodate T.38.
Regards,
Steve
Matthew Boehm wrote:
   

Since t38 is seperate from SIP, you basically need a chan_t38 right?
-Matthew
- Original Message - 
From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Friday, February 04, 2005 7:45 AM
Subject: [Asterisk-Users] T.38 bounty


 

hi
there are some comments here,
http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38%20Bounty,
that people that have earlier offered high bounties for T.38 in
asterisk. Please add up, so the one that one day manages to add good
T.38 support may get something back for it :)
roy
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[Asterisk-Users] *, BeroNet BN4S0 and misdn - problems

[Andreas Czerniak]

Hi,
i use an BN4S0 with misdn an asterisk on Linux 2.6.9. The hfcmulti module 
is loaded with option:

type=0x04 protocol=0x2,0x2,0x22,0x2 layermask=0xf,0xf,0xf,0xf
and the fourth port is connected to an ISDN PTMP (MSN) port.
Call to #72 from S0 (BN port 4) are not accepted from asterisk but why ?
Can anyone give me a hint ??

misdn debug messages follows:
lib: NEW_CR Ind with l3id:80001
new_process: New L3Id: 80001
I IND :SETUPpid:0   mode:TE addr:0  port:4
--> dad: 72 oad 177xxx channel 1 port 4
Locking Config Mutex
UnLocking Config Mutex
Locking Config Mutex
UnLocking Config Mutex
--> Bearer: Audio
* NEW CHANNEL dad:72 oad:177xxx ctx:bat-outside-isdn
* Queuing chan 0x81a406
* List is empty so new chan is Listroot
I SEND:SETUP_ACKNOWLEDGEport:4  pid:0   mode:TE addr:0
--> dad: 72 oad 177xxx channel 1 port 4
$$$ Setting up bc with stid :1104
--> Got Adr 51400104
--> Channel is 1
* Starting Ast ctx:bat-outside-isdn dad:72 oad:177xxx
GOT SETUP OK: port 4
   -- Executing Wait("mISDN/4/177xxx", "1") in new stack
I IND :RELEASE  pid:0   mode:TE addr:51400104   port:4
--> dad: 72 oad 177xxx channel 1 port 4
--> cause 101
* RELEASING CHANNEL pid:0 ctx:bat-outside-isdn dad:72 oad:177xxx state: 
DIALING
* --> State Down
* --> In State Calling|Dialing
* --> Queue Hangup
* Dequeuing chan 0x81a4060 from List 0x40da687c
* Its the first Chan
lib: RELEASE_CR Ind with l3id:80001
lib: CLEANING UP l3id: 80001
empty chan 1
Idx: 0 stack->cchan: 0 Chan 1
Idx: 1 stack->cchan: 0 Chan 2
Feb  4 17:00:55 DEBUG[30462]: chan_sip.c:831 __sip_ack: Stopping 
retransmission on '[EMAIL PROTECTED]' of 
Request 102: Found
$$$ Cleaning up bc with stid :1104
Duming again:
Idx: 0 stack->cchan: 0 Chan 1
Idx: 1 stack->cchan: 0 Chan 2
Dumped
I IND :CLEAN_UP pid:0   mode:TE addr:51400104   port:4
--> dad:  oad  channel 0 port 4
$$$ find_chan: No channel found for oad: dad:
$$$ MGMT FRAME: prim f2481 addr 50400104 dinfo 0
 == Spawn extension (bat-outside-isdn, 72, 1) exited non-zero on 
'mISDN/4/177xxx'
   -- Executing Hangup("mISDN/4/177xxx", "") in new stack
 == Spawn extension (bat-outside-isdn, h, 1) exited non-zero on 
'mISDN/4/177xxx'

part of misdn.conf:
[batports]
ports=4
language=de
context=bat-outside-isdn
msns=72
part of extensions.conf:
[bat-outside-isdn]
 exten => 72,1,Wait(1)
 exten => 72,2,Dial(SIP/72,30,t)
Regards,
Andreas.
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Re: [Asterisk-Users] PCMCIA card

[Steve Underwood]

Hi Calin,
There are several PCMCIA or PC Card E1/T1 interfaces. They are mostly 
made to turn laptops into portable test equipment. I don't know one with 
Asterisk drivers, though. Examples are:

http://www.solinet.com/solinet_notebook_packages.htm
http://www.utelsystems.com/instruments/hardware/index.php
http://www.odints.com/ftp/pub/pbriefs/Thor-PCMCIA-ProductBrief.pdf
Regards,
Steve
Calin Serbanescu wrote:
Hello,
Are there any T1/E1 PCMCIA cards available on the market supported by
zaptel drivers and asterisk ? 

I need to make some demos at my clients with asterisk and it's a pain to
move around with a midi-tower computer just for that.
Thanks,
Calin.
 

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RE: [Asterisk-Users] Callerid problems with 1.0.5

[mattf]

http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0003490

apply the patch:  app_dial_CID_nodelete.patch  
and the deleting of the original callerid will stop in v1.0.5.

Also in CVS_HEAD preserving original callerid has been given a flag 'o' in
the dial string.


MATT---


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Friday, February 04, 2005 11:24 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Callerid problems with 1.0.5


I just upgraded my two asterisk boxes to 1.0.5 stable and I've noticed
that callerid is not functioning properly. My setup looks like this:

SIP Phone <--> SER <--> Asterisk <--> Asterisk <---> PSTN

No iax is being used at this time.

The problem can be best described by the following scenarios:

1.) SIP to PSTN call: When a SIP phone calls a PSTN bound number, the
callerid displayed on the PSTN phone is the number of the PSTN phone
instead of the SIP phone's number.

2.) PSTN to SIP call: When a PSTN phone calls a SIP Phone number, the
callerid displayed on the SIP phone is the number of the SIP phone instead
of the PSTN phone's number.

For both scenarios - ${CALLERID}, ${EXTEN}, and ${CALLERIDNUM} all have the
number of the called phone for ZAP to SIP, SIP to ZAP, and SIP to SIP. I
have noticed that explicitly declaring SetCallerID(${CALLERID}) before my
dial seems to fixe this issue for only the ZAP to SIP piece. In the next
Asterisk where a SIP to SIP relay is occurring ${CALLERID} ends up matchign
${EXTEN} again.
This is causing some havoc with users calling cell phone from SIP phones.
Some users are being dumped into certain company's cell phone voicemail
because the callerid is keyed to the called phone's number.

Has anyone else experienced this problem with 1.0.5 stable? I checked the
bugs.digum.com page and found a similar bug with regard to the call being
delivered to the manager API. Also, I searched the configs and I did not
see any new settings related to callerid. If this is a simple
configuration change introduced into version 1.0.5, any info would be
greatly appreciated.
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Re: [Asterisk-Users] OT: How to "own" a telephone number?

[Jeremy Kitchen]

On Friday 04 February 2005 08:00 am, David Brodbeck wrote:
> > Is providing the ability to assign numbers to people instead of to
> > locations really that hard?  Is it really so much easier for Internet
> > domains to do it?  Or is this just an oligarchy at work?  :)

[snip]

> Billing is based on
> this, too.  If people could move numbers around willy-nilly, you'd never
> know if you were making a long-distance call or not.

so many phone companies are offering free long distance anymore to compete 
with cell phones (and of course, most, if not all cell phones are free long 
distance) that I think we'll get to a point where anyone in the US will be 
able to call anyone in the US without paying extra.

Then there are always services like Vonage... :)

I plan to get an asterisk server set up and bring in vonage (or similar) and 
FWD for carrier service.  Should be a nice fun little project.

-Jeremy

-- 
Jeremy Kitchen ++ Systems Administrator ++ Inter7 Internet Technologies, Inc.
  [EMAIL PROTECTED] ++ www.inter7.com ++ 866.528.3530 ++ 815.776.9465 int'l
  kitchen @ #qmail #gentoo on EFnet IRC ++ scriptkitchen.com/qmail
 GnuPG Key ID: 481BF7E2 ++ jabber:[EMAIL PROTECTED]


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Re: [Asterisk-Users] IAX2 register Refresh

[Liaan vd Merwe]

Ha. this is kerio winroute pro
but got it sorted.. you need to specify it in the
"target" server.. not 
regisee.
working like a charm..
ps: the default is 60 sec mine running 14sec now.

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

Sent: Friday, February 04, 2005 6:09 PM
Subject: Re: [Asterisk-Users] IAX2 register Refresh


> On February 4, 2005 10:52 am, Liaan vd Merwe wrote:
>> I been looking into the whole code strugture of
chan_iax and i see there 
>> is
>> a option to specify the refresh rate of
registrations: But there is no 
>> code
>> to actually load this from the config file thus i
changed the setting in
>> chan_so.h, and recompiled. But still my refresh
rate is 60 sec.
>>
>> I need to get this down to 15 sec (nat /pat
firewall issue)
>
> Fix the NAT/PAT device?  15s timeout is insane, most
things have many, 
> many
> minutes for timeout.
>
> -A.
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[Asterisk-Users] BRI in the US?

[Michael Graves]

OK, I asked this about a week back and met with no repsonse at all. But
perhaps its worth trying again. 

Does anyone on-list have * running BRI to their local telco? I'm
considering this as an alternative to my TDM400p card.

Michael

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Re: [Asterisk-Users] OT: How to "own" a telephone number?

[Peter Svensson]

On Fri, 4 Feb 2005, Jeremy Kitchen wrote:

> so many phone companies are offering free long distance anymore to compete 
> with cell phones (and of course, most, if not all cell phones are free long 
> distance) that I think we'll get to a point where anyone in the US will be 
> able to call anyone in the US without paying extra.

Some countries like Sweden have dropped the long distance concept 
altogether. All PSTN calls are at the local rate (1-3 cents / minute 
depending on the carrier).

Peter


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Re: [Asterisk-Users] Help with chan_h323

[Bruno Hertz]

On Fri, 2005-02-04 at 15:54 +0900, Andrew Kochetkoff wrote:

> EndedByTransportFail

You should give more details, all there can be seen from your log is
that the call is dropped due to EndedByTransportFail, which can have
various reasons.

Apparently, this is a LAN call with no NAT involved, right? But what
client (software) is involved in the call, how does your oh323 setup
look like? What are your asterisk, chan_oh323, openh323 versions?

Regards, Bruno.



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RE: [Asterisk-Users] Q: How to get the preset callerid from aCLID-no-screen E1-PRI

[Peter Svensson]

On Fri, 4 Feb 2005, Paul Brock wrote:

> >This is rather weird? What network do you receive this from? Neither 
> >ITU q.931 nor ETSI EN 300 403-1 (EiroISDN definition) lists the "Calling 
> >Number" IE among those that may be repeated. 
> 
> q.931 (and q.931e) traces include both "called number" and "calling number",
> on all Uk variants. Due to the European nature of q931, I would guess this
> to be the same across europe...

So? 

The trace included two _calling number_ IEs. Actually it included two 
calling numbers, one called number nad all the ususal IEs one would 
expext. The interesting part was the two calling numbers. 

Also, q.931 is the ITU specification. It is modified by ETSI EN 300 403-1 
when used in Europe and then further by national standards. E.g. in Sweden 
on Telia's (the national incumbent's) network the ETSI directive is 
further specified by the specification LZBA 506 403/0.

Peter


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Re: [Asterisk-Users] BRI in the US?

[Jonathan Moore]

Not sure if this is a helpful answer, but we have looked and haven't come across
anything yet for the US. Curious to see if you get any other responses.

-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Michael Graves <[EMAIL PROTECTED]>:

> OK, I asked this about a week back and met with no repsonse at all. But
> perhaps its worth trying again.
>
> Does anyone on-list have * running BRI to their local telco? I'm
> considering this as an alternative to my TDM400p card.
>
> Michael
>
> --
> Michael Graves   [EMAIL PROTECTED]
> Sr. Product Specialist  www.pixelpower.com
> Pixel Power Inc. [EMAIL PROTECTED]
>
> o713-861-4005
> o800-905-6412
> c713-201-1262
>
>
>
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Visit Winfield Public Schools at http://usd465.com
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Re: [Asterisk-Users] IAX2 register Refresh

[Liaan vd Merwe]

Ha. this is kerio winroute pro
but got it sorted.. you need to specify it in the
"target" server.. not
regisee.
working like a charm..
ps: the default is 60 sec mine running 14sec now.

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

Sent: Friday, February 04, 2005 6:09 PM
Subject: Re: [Asterisk-Users] IAX2 register Refresh


> On February 4, 2005 10:52 am, Liaan vd Merwe wrote:
>> I been looking into the whole code strugture of
chan_iax and i see there 
>> is
>> a option to specify the refresh rate of
registrations: But there is no 
>> code
>> to actually load this from the config file thus i
changed the setting in
>> chan_so.h, and recompiled. But still my refresh
rate is 60 sec.
>>
>> I need to get this down to 15 sec (nat /pat
firewall issue)
>
> Fix the NAT/PAT device?  15s timeout is insane, most
things have many, 
> many
> minutes for timeout.
>
> -A.
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Re: [Asterisk-Users] T.38 bounty

[Kevin P. Fleming]

Steve Underwood wrote:
The T.38 spec defines a streaming protocol over UDP, equivalent to RTP. 
The latest version also allows the use of RTP. T.38 can also use a 
non-quite-streaming TCP transport.
How likely is it that any equipment is going to support T.38-over-RTP 
soon, though? Since it's so new, I'd suspect that most equipment will 
only do SIP with T.38 as the media path...
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