date:20040906

On Tue, 7 Sep 2004, Oliver Breidenbach wrote:

> Update:
> 
> editing "channels/chan_zap.c" and setting "#define DEFAULT_CIDRINGS 2" 
> and recompile seems to have fixed the problem although it still shows 
> only the higher 6 numbers in the CLI console...
> 
> Very, very, very esoteric.

I still think your pstn provider has configured the link to send the last 
two digits as overlap digits instead of using enbloc dialing (sending the 
whole number at once after dialing is completed). Did you try enabling 
overlap dialing on that line ("overlapdial=yes" in zapata.conf)?

The "Called Number" was not marked with "Sending Complete" by the pstn so 
there may be overlap digits sent as information elements. By setting 
DEFAULT_CIDRINGS I think you forced asterisk to wait a bit longer. Setting 
"overlapdial=yes" should make it wait until the pstn provider sends 
"Sending Complete".

For some reason asterisk is sending "Call Proceeding" which tells the 
other end of the isdn connection that no more digits is needed. It has 
enough information to process the call. Are you sure there are no rules 
that could match only the first 6 digits? If not i really do not see what 
is going wrong. Asterisk should accumulate digits until "Sending Complete" 
or a match is found in overlap mode.

If you run a "pri intense debug" on your modified asterisk with 
DEFAULT_CIDRINGS, do you see more digits coming in in information 
elements? 

If you fail to get asterisk to work correctly you could ask your pstn 
provider to configure the isdn link to send the called number indication 
enbloc. They should be able to configure that.

Peter

> 
> Cheers,
> 
> Oliver.
> 
> 
 On 07.09.2004, at 01:30, Oliver Breidenbach wrote:
> 
> > Peter,
> >
> > thanks for trying to help.
> >
> > I've enabled overlap dialing with no effect.
> >
> > "pri intense debug span 1" gives this output:
> >
> > < Informational frame:
> > < SAPI: 00  C/R: 1 EA: 0
> > <  TEI: 000EA: 1
> > < N(S): 102   0: 0
> > < N(R): 105   P: 0
> > < 41 bytes of data
> > -- ACKing all packets from 104 to (but not including) 105
> > -- Since there was nothing left, stopping T200 counter
> > -- Stopping T203 counter since we got an ACK
> > -- Nothing left, starting T203 counter
> > < Protocol Discriminator: Q.931 (8)  len=41
> > < Call Ref: len= 2 (reference 102/0x66) (Originator)
> > < Message type: SETUP (5)
> > < [04 03 90 90 a3]
> > < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
> > capability: 3.1kHz audio (16)
> > <  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
> >  > <   Ext: 1  Coding: 0   Number Specified   Channel 
> > Type: 3
> > <   Ext: 1  Channel: 1 ]
> > < [1e 02 80 83]
> > < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard 
> > (0) 0: 0   Location: User (0)
> > <   Ext: 1  Progress Description: Calling 
> > equipment is non-ISDN. (3) ]
> > < [6c 0b 21 83 38 39 38 39 34 33 39 39 30]
> > < Calling Number (len=13) [ Ext: 0  TON: National Number (2)  NPI: 
> > ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> > <   Presentation: Presentation allowed of 
> > network provided number (3) '123456789' ]
> >
> > =^
> > I changed this number to protect privacy. But it is 9 digits and it is 
> > correct.
> > =
> >
> > < [70 07 c1 38 34 30 30 35 33]
> > < Called Number (len= 9) [ Ext: 1  TON: Subscriber Number (4)  NPI: 
> > ISDN/Telephony Numbering Plan (E.164/E.163) (1) '876543' ]
> >
> > =^
> > I also changed this. len=9 should mean 9 digits, right? the number i 
> > dialed is "87654321", the "21" is missing.
> > =
> >
> > Sending Receiver Ready (103)
> > > [ 02 01 01 ce ]
> > > Supervisory frame:
> > > SAPI: 00  C/R: 1 EA: 0
> > >  TEI: 000EA: 1
> > > Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> > > N(R): 103 P/F: 0
> > > 0 bytes of data
> > -- Restarting T203 counter
> > -- Restarting T203 counter
> > > [ 00 01 d2 ce 08 02 80 66 02 18 03 a9 83 81 ]
> > > Informational frame:
> > > SAPI: 00  C/R: 0 EA: 0
> > >  TEI: 000EA: 1
> > > N(S): 105   0: 0
> > > N(R): 103   P: 0
> > > 10 bytes of data
> > -- Restarting T203 counter
> > Stopping T_203 timer
> > Starting T_200 timer
> > > Protocol Discriminator: Q.931 (8)  len=10
> > > Call Ref: len= 2 (reference 32870/0x8066) (Terminator)
> > > Message type: CALL PROCEEDING (2)
> > > [18 03 a9 83 81]
> > > Channel ID (len= 5) [ Ext: 1  IntID: Impli

Re: [Asterisk-Users] Zaptel 'Under the Hood' Project

2004-09-06 Thread Jamie Carl

Victor Rini wrote:
Hello,
After poking and prodding at Asterisk and Zaptel for over a couple 
years now, I've dedicated some time to actually reading the code and 
trying to figure it out.

It's been fascinating. With the driver source on one part of the 
screen and a pdf of "Linux Device Drivers" on another part I've 
aquainted myself with device driver programming and the interesting 
hardware on the wildcards. I've always thought Asterisk and Zaptel 
were two of the coolest FOSS projects around and now that I've
spelunked through the code a little bit I'm curious:

Has anyone ever wrote a zaptel "under the hood" type of document, 
discussing how the pseudo tdm bus works, the zaptel hardware, etc? If 
so, please point me there.

If not, I'd like to take a stab at compiling a paper or article about 
zaptel for a general audience, technically inclined but not hard core 
technical, i.e. people like me who
have used asterisk but always wondered how it worked down to the 
hardware, spans, channels, chunks, samples level. Some help from the 
community of course would
be great, perhaps through using a blog or wiki.

Once the zaptel "dragon" is dispatched, I'd then focus on Asterisk.
What do you all think?
Regards,
Victor
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I can't speak for anyone else, but I sure as hell would be interested in 
such a document.  I don't think it would even just be for the 
"technically inclined but not hard core technical" guys either.  I 
consider myself pretty "hard core" but I just don't have the time to sit 
down and learn about how it all works on the inside.  There's just too 
many other projects that need to be done.  So in my opinion, a document 
that just lays it out in plain english would save me a heck load of time 
and allow me to learn about something that I unfortunately just don't 
have the time (or motivation) to figure out for myself and therefore 
probably wouldn't end up learning about otherwise. :)

My 2c.
Jamie
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Re: [Asterisk-Users] offtopic - channel banks

2004-09-06 Thread Chris A. Icide

Ilia,
Okay, we are stretching the limits of my official technical knowledge and 
getting into the realm of empirical knowledge (i.e. I messed around and got 
it to work).  The T1 framing and encoding (esf,b8zs for example), will get 
you green lights on the T1 side of the channel bank, but if you hook up a 
phone to the correct pairs matching a channel output on the channel bank, 
you won't get dial tone.  You'll need to match the signalling (E&M, E&M 
Wink, fxs_ls, fxs_ks, etc.) between the channel bank and the asterisk t1 
configuration.  This will then allow you top provide correct signalling to 
the analog handsets giving you dial tone as well as off-hook, on-hook 
detection, etc.

So to answer your question, E&M is not a framing protocol, it rides on top 
of the framing and encoding protocols and is the protocol that interacts 
with the channel bank to provide correct signalling between the analog 
handset and the T1 interface of Asterisk.

-Chris
On 10:59 PM 9/6/2004, Ilia Mirkin wrote:
>I must not be explaining my question clearly. Is E&M a framing protocol
>that the signal that travels "through" the T1's channels conforms to, or
>is it a wire protocol that will drive a pair on the centronics
>connector. If it is the former case, then what drives the pair on the
>centronics connector, and will i be able to plug a standard analog phone
>into it (and expect it to work)? If it is the latter, then I assume that
>there is no easy way to connect an analog phone.
>
>Thanks to everyone for responding to my questions.
>
>---
>Ilia Mirkin
>[EMAIL PROTECTED]
>
>On Mon, 2004-09-06 at 11:33, Chris A. Icide wrote:
>> Ilia,
>>
>> Think of a channel bank as a concentrator.  In a single T1 channel bank,
>> you concentrate 24 analog two wire phone connections into a single 4 wire
>> digital interface.
>>
>> So in your case, the RJ45 connector is for the T1 interface to Asterisk,
>> the local CLEC, or whatever you intend to connect it to.  On the T1 side,
>> there has to be several layers of signalling and encoding.  Alot of this
>> information is superfluous, but may help you when it comes to 
understanding
>> your configs.
>>
>> Before we even talk about E&M signalling, you have the T1 framing and
>> encoding.  This is used to allow both ends of the T1 circuit to understand
>> how the 24 channels are being configured on the 4 wire circuit.  It's
>> generally going to be either sf, d4 or esf, b8zs.  This is known as the
>> framing and encoding.
>>
>> Once those are agreed upon, then we need to set up the way the T1 is going
>> to signal across the channels.  Normal phone lines (analog) use voltages,
>> resistances, and dtmf to signal what it is doing.  Since a T1 is a digital
>> circuit we can't do that, so we need to set up another way to signal, so
>> that the channel bank knows what to do when we send some kind of digital
>> signal.  In this case, this is the E&M signalling you asked about.
>>
>> Finally, you probably are looking for some way to plug your phone's RJ11
>> connecter into the channel bank.  Unfortunately it's not that easy.  That
>> big Centronics style connecter is where you actually have to plug up the
>> phone.  there are 24 pairs of contacts in that connecter that are
>> associated with each channel on the T1 circuit.  Historically, you would
>> connect up a cable of 50 conductors connected to the centronics connector
>> on one end, and then to one side a punch down block on the other (just a
>> quick-connect style access device for copper wire).  On the other side of
>> the punch down block, you would connect the wires that would then run to
>> the remote wall jacks, etc. where your phones plug in.
>>
>> The problem you have is wither by google mastery or just plain brute force
>> testing, you need to figure out the pinout of that centronics port before
>> you can connect up any phones successfully.
>>
>> -Chris
>>
>> On 10:56 PM 9/5/2004, Ilia Mirkin wrote:
>>  >While I understand everything that you have said, I'm still a little
>>  >confused. Yes - I have what looks like a centronics connector on the
>>  >back. So, I can do "t100p with e&m signalling" <-> "act-1241 e&m card"
>>  ><-> what? Namely, if the E&M card deals with the T1 end of the channel,
>>  >how do I get that to a real phone? Will it "just work" if I plug an
>>  >analog phone onto the correct pair coming out of the connector in the
>>  >back? If not, what is the output of the E&M card? (and, more
>>  >importantly, what would I need to do to hook it up to an analog phone?)
>>  >
>>  >Thanks for clearing things up.
>>  >
>>  >---
>>  >Ilia Mirkin
>>  >[EMAIL PROTECTED]
>>  >
>>  >On Sun, 2004-09-05 at 04:31, Steven Critchfield wrote:
>>  >> On Sun, 2004-09-05 at 03:10, Ilia Mirkin wrote:
>>  >> > hi,
>>  >> >
>>  >> > i have some newbie questions about channel banks. i have an adtran
>>  >> > act-1241 sitting around. it accepts D4 modules, and it contains a
>number
>>  >> > of e&m cards.
>>  >> >
>>  >> > first of all, how does th

Re: [Asterisk-Users] incoming number truncated

On Mon, 6 Sep 2004, Eric Wieling wrote:

> On Mon, 2004-09-06 at 16:34, Oliver Breidenbach wrote:
> > We are calling from a number in the same local area code and there 
> > seems to be only the 6 most significant numbers of the target adress 
> > arrive in Asterisk.
> 
> Try pridialplan=unknown

I thought the pridialplan only affected the outgoing messages, not the 
incoming?

Peter


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Re: [Asterisk-Users] offtopic - channel banks

2004-09-06 Thread Ilia Mirkin

I must not be explaining my question clearly. Is E&M a framing protocol
that the signal that travels "through" the T1's channels conforms to, or
is it a wire protocol that will drive a pair on the centronics
connector. If it is the former case, then what drives the pair on the
centronics connector, and will i be able to plug a standard analog phone
into it (and expect it to work)? If it is the latter, then I assume that
there is no easy way to connect an analog phone.

Thanks to everyone for responding to my questions.

---
Ilia Mirkin
[EMAIL PROTECTED]

On Mon, 2004-09-06 at 11:33, Chris A. Icide wrote:
> Ilia,
> 
> Think of a channel bank as a concentrator.  In a single T1 channel bank, 
> you concentrate 24 analog two wire phone connections into a single 4 wire 
> digital interface.
> 
> So in your case, the RJ45 connector is for the T1 interface to Asterisk, 
> the local CLEC, or whatever you intend to connect it to.  On the T1 side, 
> there has to be several layers of signalling and encoding.  Alot of this 
> information is superfluous, but may help you when it comes to understanding 
> your configs.
> 
> Before we even talk about E&M signalling, you have the T1 framing and 
> encoding.  This is used to allow both ends of the T1 circuit to understand 
> how the 24 channels are being configured on the 4 wire circuit.  It's 
> generally going to be either sf, d4 or esf, b8zs.  This is known as the 
> framing and encoding.
> 
> Once those are agreed upon, then we need to set up the way the T1 is going 
> to signal across the channels.  Normal phone lines (analog) use voltages, 
> resistances, and dtmf to signal what it is doing.  Since a T1 is a digital 
> circuit we can't do that, so we need to set up another way to signal, so 
> that the channel bank knows what to do when we send some kind of digital 
> signal.  In this case, this is the E&M signalling you asked about.
> 
> Finally, you probably are looking for some way to plug your phone's RJ11 
> connecter into the channel bank.  Unfortunately it's not that easy.  That 
> big Centronics style connecter is where you actually have to plug up the 
> phone.  there are 24 pairs of contacts in that connecter that are 
> associated with each channel on the T1 circuit.  Historically, you would 
> connect up a cable of 50 conductors connected to the centronics connector 
> on one end, and then to one side a punch down block on the other (just a 
> quick-connect style access device for copper wire).  On the other side of 
> the punch down block, you would connect the wires that would then run to 
> the remote wall jacks, etc. where your phones plug in.
> 
> The problem you have is wither by google mastery or just plain brute force 
> testing, you need to figure out the pinout of that centronics port before 
> you can connect up any phones successfully.
> 
> -Chris
> 
> On 10:56 PM 9/5/2004, Ilia Mirkin wrote:
>  >While I understand everything that you have said, I'm still a little
>  >confused. Yes - I have what looks like a centronics connector on the
>  >back. So, I can do "t100p with e&m signalling" <-> "act-1241 e&m card"
>  ><-> what? Namely, if the E&M card deals with the T1 end of the channel,
>  >how do I get that to a real phone? Will it "just work" if I plug an
>  >analog phone onto the correct pair coming out of the connector in the
>  >back? If not, what is the output of the E&M card? (and, more
>  >importantly, what would I need to do to hook it up to an analog phone?)
>  >
>  >Thanks for clearing things up.
>  >
>  >---
>  >Ilia Mirkin
>  >[EMAIL PROTECTED]
>  >
>  >On Sun, 2004-09-05 at 04:31, Steven Critchfield wrote:
>  >> On Sun, 2004-09-05 at 03:10, Ilia Mirkin wrote:
>  >> > hi,
>  >> >
>  >> > i have some newbie questions about channel banks. i have an adtran
>  >> > act-1241 sitting around. it accepts D4 modules, and it contains a number
>  >> > of e&m cards.
>  >> >
>  >> > first of all, how does this thing work? a t1 contains 24 channels, and i
>  >> > noticed that the channel bank has space for 24 cards. what do these
>  >> > cards do? what are their outputs? the ones that are in there have some
>  >> > outputs on the front marked "test", but nothing else. there are a number
>  >> > of wires coming out the back (48, if i had to guess), and it has a few
>  >> > ports on the front which seem to be able to take in a T1. am i correct
>  >> > in understanding that it is the card in the bank that determines the
>  >> > signalling style, and not the t1? as such, is there no way that i could
>  >> > use it in its current configuration to have it talk with analog phones
>  >> > (i.e. something like t100p -> act-1241 with e&m cards -> phone)? i'm a
>  >> > bit unclear on the different signalling types, and their
>  >> > intercompatibilities.
>  >> >
>  >> > if anyone could shed any light into this, i would very much appreciate
>  >> > it.
>  >>
>  >> Think of the T1 as 24 digital digital pathways. The coding of each
>  >> pathway must be compat

Re: [Asterisk-Users] [patch] allow the transfer keys from app_dial's't' and 'T' and hangup key 'H' to be configured via features.conf

2004-09-06 Thread Josh Roberson

Yeah, sorry i kinda jumped the gun on ya there..I plan to update it 
again pretty soon, but just for future reference, they normal method to 
apply a patch would be to use the patch command, as follows:

patch -p0 < patch.txt
usually this is done from the top-level source directory for the package 
you are trying to patch, replacing patch.txt with the name of the diff 
(patch).   Sometimes patch makers don't use paths in their patch, so you 
kinda have to know where the files are that you're trying to patch, but 
i try to make mine all work against the top-level source directory.

twisted
box100 wrote:
Wow, you guys are fast. My apologies, twisted. I realize there must have been a reason why it 
wasn't marked "resolved" and included in the CVS HEAD, but I was under the impression 
that those who wanted to and have the knowhow could download and apply the patch. Didn't mean to 
imply you or anyone else *stated* that it was finished, it just seemed from the dialog in the 
bug report that work on it had been completed and just not marked as such. My mistake.
It will be great when it is done whenever that happens since the added functionality 
really will increase the usefulness of Asterisk as a PBX.
Thanks for the speedy reply.
Roger

From: [EMAIL PROTECTED] on behalf of Josh Roberson
Sent: Tue 9/7/2004 00:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [patch] allow the transfer keys from app_dial's't' and 
'T' and hangup key 'H' to be configured via features.conf

Roger,
   at no point did I say I was finished with this patch.   I did get a
little frustrated early on in the development.   Currently this patch is
broken due to recent changes in cvs, and I'm about to tag it with a
post-1.0 tag in the bugtracker since there seems to be lots of interest
in it's existance, but it needs a little work.   Please do not make
assumptions as you have below,   because I am the author of this patch,
and I do *NOT* feel as though i'm finished, nor did i say so anywhere in
the bugnotes.
Thanks.
twisted
 

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[Asterisk-Users] carrier connection options

2004-09-06 Thread Peter Boot

Can anyone tell me if there is an alternative option to connecting Asterisk
as a SIP server to a carrier using T1/E1 lines or a carrier that will
terminate a routed SIP call ? 

I am looking for a more cost effective solution that will avoid the setup
and incremental cost of VOIP gateways and T1/E1 lines as the need for more
and more concurrent calls escalates.
Thanks in advance.

---
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[Asterisk-Users] iaxy vs sipura

2004-09-06 Thread Florin Andrei

I need a cheap simple adaptor for analog phones to use with Asterisk. It
should be some kind of "configure and forget" type of device, to use at
the office, or just throw it in a road warrior's bag and use it while
travelling, to call back to the "mothership".
I can't decide between iaxy and sipura. Can you guys help? Which one
would you use? (and why?)
I feel that iaxy might have an advantage while piercing through NAT
firewalls (at hotels and such), because of IAX, but i could be wrong.

Or can you recommend something else?

-- 
Florin Andrei

http://florin.myip.org/

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[Asterisk-Users] Re: Zaptel 'Under the Hood' Project

2004-09-06 Thread Vikram Rangnekar

+++ Victor Rini [06/09/04 18:12 -0700]:
> Hello,
> 
> After poking and prodding at Asterisk and Zaptel for over a couple years 
> now, I've dedicated some time to actually reading the code and trying to 
> figure it out.
> 
> It's been fascinating. With the driver source on one part of the screen 
> and a pdf of "Linux Device Drivers" on another part I've aquainted 
> myself with device driver programming and the interesting hardware on 
> the wildcards. I've always thought Asterisk and Zaptel were two of the 
> coolest FOSS projects around and now that I've
> spelunked through the code a little bit I'm curious:
> 
> Has anyone ever wrote a zaptel "under the hood" type of document, 
> discussing how the pseudo tdm bus works, the zaptel hardware, etc? If 
> so, please point me there.
> 
> If not, I'd like to take a stab at compiling a paper or article about 
> zaptel for a general audience, technically inclined but not hard core 
> technical, i.e. people like me who
> have used asterisk but always wondered how it worked down to the 
> hardware, spans, channels, chunks, samples level. Some help from the 
> community of course would
> be great, perhaps through using a blog or wiki.
> 
> Once the zaptel "dragon" is dispatched, I'd then focus on Asterisk.
> 
> What do you all think?
> 
> Regards,
> Victor
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Sounds extemly intresting maybe you should go right ahead and write a page on
that. 
-- 
regards
Vikram (http://www.vicramresearch.com)
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[Asterisk-Users] Zaptel errors with E100P + TDM40B

2004-09-06 Thread Webn1

Hello !

I try to use on the same server E100P + TDM40B without success.
When i add one one of them, no error on module load.

But when both are active, zaptel see strange configuration.

Here is my configuration :

zaptel.conf :
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

fxols=32-35

Error display :
# modprobe wct1xxp
ZT_CHANCONFIG failed on channel 32: No such device or address (6)
FATAL: Error running install command for wct1xxp

/proc/interrupts :
 0: 677723  XT-PIC  timer
  1:  8IO-APIC-edge  i8042
  8:  4IO-APIC-edge  rtc
  9:  0   IO-APIC-level  acpi
 14:   4409IO-APIC-edge  ide0
 15: 13IO-APIC-edge  ide1
177: 390032   IO-APIC-level  t1xxp
185: 193773   IO-APIC-level  wctdm
193:  22903   IO-APIC-level  eth0

I have also disable all motherboard extension as firewire, second ide
controler etc...

Any idea?

Thanks !



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RE: [Asterisk-Users] [patch] allow the transfer keys from app_dial's't' and 'T' and hangup key 'H' to be configured via features.conf

Wow, you guys are fast. My apologies, twisted. I realize there must have been a reason 
why it wasn't marked "resolved" and included in the CVS HEAD, but I was under the 
impression that those who wanted to and have the knowhow could download and apply the 
patch. Didn't mean to imply you or anyone else *stated* that it was finished, it just 
seemed from the dialog in the bug report that work on it had been completed and just 
not marked as such. My mistake.

It will be great when it is done whenever that happens since the added functionality 
really will increase the usefulness of Asterisk as a PBX.

Thanks for the speedy reply.

Roger

From: [EMAIL PROTECTED] on behalf of Josh Roberson
Sent: Tue 9/7/2004 00:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [patch] allow the transfer keys from app_dial's't' and 
'T' and hangup key 'H' to be configured via features.conf

Roger,

at no point did I say I was finished with this patch.   I did get a
little frustrated early on in the development.   Currently this patch is
broken due to recent changes in cvs, and I'm about to tag it with a
post-1.0 tag in the bugtracker since there seems to be lots of interest
in it's existance, but it needs a little work.   Please do not make
assumptions as you have below,   because I am the author of this patch,
and I do *NOT* feel as though i'm finished, nor did i say so anywhere in
the bugnotes.

Thanks.
twisted

box100 wrote:

> Can anyone tell me how I can implement the features added in the
> following link for call transfer? The authors seem to feel they are
> finished but it doesn't appear to have been integrated into what
> everyone can download. It is referred to as a patch but I don't
> understand how it could be applied. Here is the link:
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0002010
> 
> I guess I just don't understand how to apply patches
> 
> Thanks in advance,
> Roger Easlick
>
>
>
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RE: [Asterisk-Users] VM access

You could add an extension to your default context that takes you to VM:
 
exten => 500,1,VoiceMailMain
exten => 500,2,Hangup

Simply include the default context in your incoming context
 
include => default
 
Roger



From: [EMAIL PROTECTED] on behalf of Larry Shields
Sent: Mon 9/6/2004 12:48
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] VM access


Can someone tell me how to get to a mailbox login prompt when accessing the Asterisk 
VM remotely via a PSTN line?  I am running version CSV 8/25/04.
 
Thanks,
Larry
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Re: [Asterisk-Users] [patch] allow the transfer keys from app_dial's 't' and 'T' and hangup key 'H' to be configured via features.conf

2004-09-06 Thread Josh Roberson

Roger,
   at no point did I say I was finished with this patch.   I did get a 
little frustrated early on in the development.   Currently this patch is 
broken due to recent changes in cvs, and I'm about to tag it with a 
post-1.0 tag in the bugtracker since there seems to be lots of interest 
in it's existance, but it needs a little work.   Please do not make 
assumptions as you have below,   because I am the author of this patch, 
and I do *NOT* feel as though i'm finished, nor did i say so anywhere in 
the bugnotes.

Thanks.
twisted
box100 wrote:
Can anyone tell me how I can implement the features added in the 
following link for call transfer? The authors seem to feel they are 
finished but it doesn't appear to have been integrated into what 
everyone can download. It is referred to as a patch but I don't 
understand how it could be applied. Here is the link:
 
http://bugs.digium.com/bug_view_page.php?bug_id=0002010
 
I guess I just don't understand how to apply patches
 
Thanks in advance,
Roger Easlick


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[Asterisk-Users] [patch] allow the transfer keys from app_dial's 't' and 'T' and hangup key 'H' to be configured via features.conf

Can anyone tell me how I can implement the features added in 
the following link for call transfer? The authors seem to feel they are finished 
but it doesn't appear to have been integrated into what everyone can download. 
It is referred to as a patch but I don't understand how it could be applied. 
Here is the link:
 
http://bugs.digium.com/bug_view_page.php?bug_id=0002010
 
I guess I just don't understand how to apply patches
 
Thanks in advance,
Roger Easlick___
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RE: [Asterisk-Users] Re: Asterisk Conferencing using g729

Thanks, Tony, you answered a question about g729 licencing and * conferences that I 
wanted ask. Very enlightening. I was wondering about that because it seemed to be 
using a license for each connections, despite the Sipura natively supporting g729, but 
I wasn't sure that that is the way it has to be.

Another related question: Is there a way to just use g729 for the conference and for 
nothing else. The problem I have is that I have Broadvoice ( BV rocks, by the way) 
which requires ULAW and sends DTMF inband. If I allow g729 in the sip.conf, Asterisk 
complains that inband dtmf is only supported under ULAW and incoming dtmf does not 
work through Asterisk, something I must have.

Well I have partially solved the problem in the paragraph above. It appears that if I 
leave g729 out of the general section of sip.conf, but add allow=g729 to my SPA-2000 
device section in the sip.conf file I still get BV incoming dtmf to work and I get the 
SPA-2000 to use g729. If, however. I add allow=ulaw to the SPA-2000 section, it uses 
ulaw even though I am using setvar=g729 right before the redirect to the conference 
room as below:

exten => 3001,1,setvar(SIP_CODEC=g729)
exten => 3001,2,Meetme,1000,Maps

Asterisk does say that the codec is being changed to g729 but the SIP  SHOW CHANNELS 
command tells me the channel is using only ULAW. The real interest in g729 is saving 
OUTBOUND bandwidth and thus forcing g729 to be used with any device outside the 
firewall/router. At the same time I need ulaw. I would think that is what the setvar 
command as used above is for but it doesn't seem to have any effect. How do I force 
anyone from the outside to use only g729 to connect to my conferences but allow them 
to access the internal extensions using ulaw if they have it available?

Here and excerpt from my log:
Sep  7 00:03:13 NOTICE[-174232656]: chan_sip.c:1834 sip_answer: Changing codec to 
'g729' for this call because of ${SIP_CODEC) variable
sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)   Format
192.168.1.5  220191e0706e-51  00101/00102   ULAW

Thanks,
Roger

From: [EMAIL PROTECTED] on behalf of Tony Mountifield
Sent: Mon 9/6/2004 16:15
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Asterisk Conferencing using g729

In article <[EMAIL PROTECTED]>,
William Suffill <[EMAIL PROTECTED]> wrote:
> Good call Daniel I didn't even notice that.
>
> As far as number of license it really depends on how many concurrent
> calls you will be doing and if asterisk needs to transcode at all. If
> you call from g729 device to g729 you are fine but g729 to vm would be
> 1 license etc.

And if you are conferencing, you need one G.729 licence for each
conference participant, because Asterisk can't mix G.729 natively,
so it transcodes each channel from G.729 to Signed Linear.

Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual

2004-09-06 Thread Jamie Carl

Bob Knight wrote:
I have MIBs for whatever version I am running that I am more than
happy to share.  Anyone know where I can place these for public access.
Sort of like the freedomphones site for Polycom.  We could then
put pointers on the wiki.
Thanks for the info tho.  If mbrowse is console based it will be very 
useful. :)

It has gui (X, gtk I think) if that is what you mean by console based.
I can ssh into a remote * server and do get walks on my 1204's.

Bob,
I've managed to source the MIBs from another extremely helpful list 
member so hopefully I'm all sorted.  :)

As for posting them, as I'm sure there are others out there that are 
interested, there is a website called www.mibdepot.com which is trying 
to collect as many MIBs as possible and currently has a request for the 
APA III-4FXO MIB.  If you email it to the webmaster of that site he'll 
post it as part of his collection.  I found this site while I was 
looking for it myself so hopefully others will look there too as they 
already have quite a few MIBs available.

Jamie
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[Asterisk-Users] RE: multiline IP hardphone w/ FDX speakerphone?

2004-09-06 Thread Rana Dutt

I use Polycom IP 500's with Asterisk. These phones have 3 line appearances
and excellent full duplex speakerphones. They work very well with Asterisk,
and I was able to use the Web interface to set them up quite easily. The
default Web password given at voip-info.org is wrong, I added a comment on
the Polycom Phones page giving the correct one ("456"). With Asterisk, you
can do both consultative and blind transfers with this phone, and the
Conference button works as you would expect. It's a very high quality
product, and it has more line appearances than the cisco 7940G. I don't know
whether you can get them for less than $200, I paid about $240 each.
Rana Dutt
--

Message: 8
Date: Mon, 6 Sep 2004 19:24:41 +0200
From: "Stewart Nelson" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] multiline IP hardphone w/ FDX speakerphone?
To: <[EMAIL PROTECTED]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original

Could someone please recommend a reasonably priced IP phone
that works well with *, has a decent (full duplex, echo canceling)
speakerphone, has at least two line appearances, and can
transfer / conference reliably?

The Wiki lists 35 brands of hardphone, but:
1. Most seem to be toys.
2. For many, there is no info on e.g. speakerphone characteristics.
3. When one seems technically promising, e.g. Polycom IP500, there
   are *lots* of negative postings about support, integration, etc.

Is there anything decent out there for < $200?

Thanks,

Stewart

**

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[Asterisk-Users] IAX2/GSM VOIP troubleshooting

2004-09-06 Thread Michael George

Last week I was able to do some debugging of the problem I'm having with
IAX2/GSM, residential-grade broadband, and VOIP.

To summarize, I am having a great learning experience with * and Zap cards,
SIP and IAX2.  I hit a wall though, when I registered with iaxtel and tried
doing VOIP.

I spend the better part of a workday with the jitterbuffer and all sorts of
settings and finally started to conclude that my systems might just be
inadequate for what I was asking them to do.

However, on Friday I went to our other office to try a DP system out.  I got *
up and running on it and IAX2/GSM was doing VOIP just fine with Digium's
iaxtel number, even with two channels.  I figured that was the problem, not
enough interrupt handling and transcoding capacity in my systems.

But I brought that system back to this office and tried it here and I have the
same choppiness that I have with my other (single processor) systems here.
So, the problem has to be in my net connection.

My ISP is willing to work with me on debugging the issue, and that's good.
But I'm not sure what types of things to look for that we might be able to
remedy.

At the office where it works fine, we have a Comcast cable modem providing
broadband.  It will do up to 5Mb/s down and 256Kb up.  I'm not sure if it's
full or half duplex.  But it sounds gread.

At this office where I normally work, I have a wireless broadband connection
to a POP in a town 5mi from here which connects via wireless to a town 10mi
from there and then it's either phoneline or wireless to the ISP's ISP.  It
seems like it might be a dicey connection, but I've had tremendous luck with
it for over 3 years now.  As a matter of fact, I regularly use a MultiTech
MultiVOIP with a 9.6kb/s voice coder on it and hardly anyone notices that it's
a VOIP connection.  (It *is* a half-duplex connection, if that matters.)
However, with the 13kb/s voice coder of GSM it's so choppy it unusable.

I don't think the problem is in IAX2/GSM.  I don't think the problem is in
bandwidth (I regularly pull 40KB/s down on this connection and I don't think
there's any limiting going up).  But there must be *something* about my
broadband connection here that is causing problems with IAX2/GSM.

I am hoping some of you networking/telephony/VOIP gurus might have some
insight into what I might look into to get this resolved.  Sometimes when I call
Digium's iaxtel number I get the "Received mini frame before first full frame"
error, but often I don't.  I've checked the archives and the solutions there
for that were not helpful.  Is there perhaps fragmenting of the packet?  Maybe
IAX2 need a minimum transmission unit that isn't getting through?  Or maybe
it's sensitive to buffering that a router in transit migth be doing for more
efficient use of the bandwidth?

Next week I might be able to go to my ISP's location with a SIP phone and my
little * testing box to see if eliminating the two radio hops from their
location to me makes a difference.  If not, we have big trouble.  If that does
help, then we need to figure out what in the ISP's equipment is causing the
problems.  

Any helpful hints as to what I could investigate to get this narrowed down?

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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[Asterisk-Users] Zaptel 'Under the Hood' Project

2004-09-06 Thread Victor Rini

Hello,
After poking and prodding at Asterisk and Zaptel for over a couple years 
now, I've dedicated some time to actually reading the code and trying to 
figure it out.

It's been fascinating. With the driver source on one part of the screen 
and a pdf of "Linux Device Drivers" on another part I've aquainted 
myself with device driver programming and the interesting hardware on 
the wildcards. I've always thought Asterisk and Zaptel were two of the 
coolest FOSS projects around and now that I've
spelunked through the code a little bit I'm curious:

Has anyone ever wrote a zaptel "under the hood" type of document, 
discussing how the pseudo tdm bus works, the zaptel hardware, etc? If 
so, please point me there.

If not, I'd like to take a stab at compiling a paper or article about 
zaptel for a general audience, technically inclined but not hard core 
technical, i.e. people like me who
have used asterisk but always wondered how it worked down to the 
hardware, spans, channels, chunks, samples level. Some help from the 
community of course would
be great, perhaps through using a blog or wiki.

Once the zaptel "dragon" is dispatched, I'd then focus on Asterisk.
What do you all think?
Regards,
Victor
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Re: [Asterisk-Users] multiline IP hardphone w/ FDX speakerphone?

2004-09-06 Thread Kevin P. Fleming

SeshKanuri wrote:
Checkout the netweb-301 and 302 available on ebay now and also from Netweb
Group. This Phone is now available for $79.99 on ebay
Use the links below:
http://cgi6.ebay.com/ws/eBayISAPI.dll?ViewSellersOtherItems&userid=netwebgroup&include=0&since=-1&sort=3&rows=50
http://ipphone.eezeephone.com/
First, you know darn well you are not supposed to post ads here, even in 
response to others questions.

Second, when the original poster specifically asked for two line 
appearances and a full duplex speakerphone, what makes you think that 
spamming the list with an ad for a unit that does neither of those 
things will help your reputation with the Asterisk community?
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Re: [Asterisk-Users] Wait for Dialtone syntax in Dial cmd?

On Mon, 2004-09-06 at 19:18, Arick Davis wrote:
> I’ve been searching the archives for the proper “Wait for Dial tone”
> command in the “Dial(Zap/g1/18005551212)”  dial sting. Does anyone
> have an example of it’s use?

There isn't one.  However there is a "wait .5 second".  However it only
works on ANALOG Zap ports.

Dial(Zap/g1/w18005551212)


-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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Re: [Asterisk-Users] incoming number truncated

On Mon, 2004-09-06 at 16:34, Oliver Breidenbach wrote:
> We are calling from a number in the same local area code and there 
> seems to be only the 6 most significant numbers of the target adress 
> arrive in Asterisk.

Try pridialplan=unknown

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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[Asterisk-Users] Wait for Dialtone syntax in Dial cmd?

2004-09-06 Thread Arick Davis









I’ve been searching the archives for the proper “Wait
for Dial tone” command in the “Dial(Zap/g1/18005551212)”
 dial sting. Does anyone have an
example of it’s use?

 

Arick

 






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Re: [Asterisk-Users] DTMF information?

2004-09-06 Thread Steve Underwood

Chris Lee wrote:
I am looking at building an IVR product with a few interesting 
features and need some more information about how asterisk and VoIP 
work and what I can get from them.

As far as I can tell when I use ISDN/GSM telephone networks the DTMF 
information travels as data representing 'start tone' and 'stop tone' 
for each button pressed, it is then generated at the other end if an 
audio representation is required.
I am interested to know if I can get access to these events 'start 
tone' and 'stop tone' through the dialplan or an AGI or by acting as a 
VoIP device. Or of course if I am completely off track and should give 
up now.

I am looking to get the length of time a button was held down rather 
than that it was pressed.

Thanks for any help
ISDN never does this. GSM only does this between the handset and the 
base-station. You only see DTMF tones from outside the GSM network 
itself. For fancy IVRs, beware that the timing of DTMF from a GSM 
handset has nothing to do with the timing of the user's keypresses. 
Because the base-station generates the tones, it controls their timing, 
and always generates rather long slow pulses of DTMF tones.

Regards,
Steve
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Re: [Asterisk-Users] multiline IP hardphone w/ FDX speakerphone?

2004-09-06 Thread SeshKanuri

Stewart,

Checkout the netweb-301 and 302 available on ebay now and also from Netweb
Group. This Phone is now available for $79.99 on ebay

Use the links below:
http://cgi6.ebay.com/ws/eBayISAPI.dll?ViewSellersOtherItems&userid=netwebgroup&include=0&since=-1&sort=3&rows=50

http://ipphone.eezeephone.com/



- Original Message - 
From: "Stewart Nelson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, September 06, 2004 10:24 AM
Subject: [Asterisk-Users] multiline IP hardphone w/ FDX speakerphone?


> Could someone please recommend a reasonably priced IP phone
> that works well with *, has a decent (full duplex, echo canceling)
> speakerphone, has at least two line appearances, and can
> transfer / conference reliably?
>
> The Wiki lists 35 brands of hardphone, but:
> 1. Most seem to be toys.
> 2. For many, there is no info on e.g. speakerphone characteristics.
> 3. When one seems technically promising, e.g. Polycom IP500, there
>are *lots* of negative postings about support, integration, etc.
>
> Is there anything decent out there for < $200?
>
> Thanks,
>
> Stewart
>
> ___
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Re: [Asterisk-Users] Codecs for fax traffic

2004-09-06 Thread Steve Underwood

Eric Jacksch wrote:
Are there any codecs that are particularly good for fax traffic?  Any to avoid?
---
Eric Jacksch
[EMAIL PROTECTED]
 

See http://www.opencall.org/faq
Regards,
Steve
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Re: [Asterisk-Users] incoming number truncated

Update:
editing "channels/chan_zap.c" and setting "#define DEFAULT_CIDRINGS 2" 
and recompile seems to have fixed the problem although it still shows 
only the higher 6 numbers in the CLI console...

Very, very, very esoteric.
Cheers,
Oliver.
On 07.09.2004, at 01:30, Oliver Breidenbach wrote:
Peter,
thanks for trying to help.
I've enabled overlap dialing with no effect.
"pri intense debug span 1" gives this output:
< Informational frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
< N(S): 102   0: 0
< N(R): 105   P: 0
< 41 bytes of data
-- ACKing all packets from 104 to (but not including) 105
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
< Protocol Discriminator: Q.931 (8)  len=41
< Call Ref: len= 2 (reference 102/0x66) (Originator)
< Message type: SETUP (5)
< [04 03 90 90 a3]
< Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: 3.1kHz audio (16)
<  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
 [ 02 01 01 ce ]
> Supervisory frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> N(R): 103 P/F: 0
> 0 bytes of data
-- Restarting T203 counter
-- Restarting T203 counter
> [ 00 01 d2 ce 08 02 80 66 02 18 03 a9 83 81 ]
> Informational frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000EA: 1
> N(S): 105   0: 0
> N(R): 103   P: 0
> 10 bytes of data
-- Restarting T203 counter
Stopping T_203 timer
Starting T_200 timer
> Protocol Discriminator: Q.931 (8)  len=10
> Call Ref: len= 2 (reference 32870/0x8066) (Terminator)
> Message type: CALL PROCEEDING (2)
> [18 03 a9 83 81]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0
>ChanSel: Reserved
>   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3
>   Ext: 1  Channel: 1 ]
-- Accepting call from '123456789' to '876543' on channel 0/1, 
span 1

Any ideas?
Cheers,
Oliver.
On 07.09.2004, at 00:51, Peter Svensson wrote:
On Mon, 6 Sep 2004, Oliver Breidenbach wrote:
We are calling from a number in the same local area code and there
seems to be only the 6 most significant numbers of the target adress
arrive in Asterisk.
For example, we are calling 9123 and the CLI shows only the 
91
and tries to match that with a extension rule.

(We've set up a rule for _9XXX with StripMSD(5) and a rule for
extension 123.)
The strange thing is that if I dial the number with the area code 
(i.e.
089/9123) or from a mobile fone it works and properly connects to
the correct extension.
Perhaps the rest of the number is sometimes sent as overlap digits? 
Try
doing "pri intense debug span " and see if there are additioal 
digits
coming in as overlap. Or try enabling overlap dialing for that span.

Peter
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RE: [Asterisk-Users] Asterisk Conferencing using g729

Yes, thanks for the comment, and I did configure the sip.conf for the Sipura -- sorry 
for the confusion. The reference to iax.conf is because I am running FWD through IAX 
so I would need to configure iax for those connecting through FWD, wouldn't I?
 
Roger



From: [EMAIL PROTECTED] on behalf of William Suffill
Sent: Mon 9/6/2004 15:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Conferencing using g729



Good call Daniel I didn't even notice that.

As far as number of license it really depends on how many concurrent
calls you will be doing and if asterisk needs to transcode at all. If
you call from g729 device to g729 you are fine but g729 to vm would be
1 license etc.


On Mon, 06 Sep 2004 04:51:26 -0500, Daniel Jimenez <[EMAIL PROTECTED]> wrote:
>
>
> box100 wrote:
>
> > My iax.conf file includes the following under the general section
>
> A SIPURA is a SIP device, configure the codecs under sip.conf not IAX.conf.
>
> > disallow=all
> > bandwidth=low
> > allow=g729
> > allow=ulaw
> >
> > Thanks,
> > Roger Easlick
> >
> >
> > 
> >
> > ___
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> --
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>
>
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Re: [Asterisk-Users] incoming number truncated

Peter,
thanks for trying to help.
I've enabled overlap dialing with no effect.
"pri intense debug span 1" gives this output:
< Informational frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
< N(S): 102   0: 0
< N(R): 105   P: 0
< 41 bytes of data
-- ACKing all packets from 104 to (but not including) 105
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
< Protocol Discriminator: Q.931 (8)  len=41
< Call Ref: len= 2 (reference 102/0x66) (Originator)
< Message type: SETUP (5)
< [04 03 90 90 a3]
< Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: 3.1kHz audio (16)
<  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
 [ 02 01 01 ce ]
> Supervisory frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> N(R): 103 P/F: 0
> 0 bytes of data
-- Restarting T203 counter
-- Restarting T203 counter
> [ 00 01 d2 ce 08 02 80 66 02 18 03 a9 83 81 ]
> Informational frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000EA: 1
> N(S): 105   0: 0
> N(R): 103   P: 0
> 10 bytes of data
-- Restarting T203 counter
Stopping T_203 timer
Starting T_200 timer
> Protocol Discriminator: Q.931 (8)  len=10
> Call Ref: len= 2 (reference 32870/0x8066) (Terminator)
> Message type: CALL PROCEEDING (2)
> [18 03 a9 83 81]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0
>ChanSel: Reserved
>   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3
>   Ext: 1  Channel: 1 ]
-- Accepting call from '123456789' to '876543' on channel 0/1, span 
1

Any ideas?
Cheers,
Oliver.
On 07.09.2004, at 00:51, Peter Svensson wrote:
On Mon, 6 Sep 2004, Oliver Breidenbach wrote:
We are calling from a number in the same local area code and there
seems to be only the 6 most significant numbers of the target adress
arrive in Asterisk.
For example, we are calling 9123 and the CLI shows only the 91
and tries to match that with a extension rule.
(We've set up a rule for _9XXX with StripMSD(5) and a rule for
extension 123.)
The strange thing is that if I dial the number with the area code 
(i.e.
089/9123) or from a mobile fone it works and properly connects to
the correct extension.
Perhaps the rest of the number is sometimes sent as overlap digits? Try
doing "pri intense debug span " and see if there are additioal 
digits
coming in as overlap. Or try enabling overlap dialing for that span.

Peter
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[Asterisk-Users] Horrible noise instead of indications

2004-09-06 Thread Roger Schreiter

Hi,
I just upgraded asterisk from 0.7 to latest CVS-head.
Now the indications (ringing, busy, ...) are a horrible
noise.
What went wrong?
Roger.
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Re: [Asterisk-Users] what does the prilocaldialplan do?

On Mon, 6 Sep 2004, Oliver Breidenbach wrote:

> there is an option in zaptel.conf where you can configure a 
> "prilocaldialplan" to national, local, international and more.
> 
> What does that do?

That is what the "a subscriber number" (more or less caller id) is 
marked as when it is sent in the setup message. The "pridialplan" is what 
the called number is marked as.

Peter

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Re: [Asterisk-Users] incoming number truncated

On Mon, 6 Sep 2004, Oliver Breidenbach wrote:

> We are calling from a number in the same local area code and there 
> seems to be only the 6 most significant numbers of the target adress 
> arrive in Asterisk.
> 
> For example, we are calling 9123 and the CLI shows only the 91 
> and tries to match that with a extension rule.
> 
> (We've set up a rule for _9XXX with StripMSD(5) and a rule for 
> extension 123.)
> 
> The strange thing is that if I dial the number with the area code (i.e. 
> 089/9123) or from a mobile fone it works and properly connects to 
> the correct extension.

Perhaps the rest of the number is sometimes sent as overlap digits? Try 
doing "pri intense debug span " and see if there are additioal digits 
coming in as overlap. Or try enabling overlap dialing for that span.

Peter

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[Asterisk-Users] signaling multiple callers in a Queue

Hi there,
the X-Pro SIP phone has up to six lines. Is there a way to have 4 of 
them ring at the same time if there are 4 people in a Queue?

The idea is that we don't want an auto-voice system so our agents do 
need to see if there are multiple people in the queue and be able to 
handle more than one caller at the same time.

Any ideas?
Regards,
Oliver.
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Re: [Asterisk-Users] Codecs for fax traffic

2004-09-06 Thread Steven Critchfield

On Mon, 2004-09-06 at 17:09, Eric Jacksch wrote:
> Are there any codecs that are particularly good for fax traffic?  Any to avoid?

Google, google, google google.
http://www.google.com/search?hl=en&ie=UTF-8&q=fax+codec+site%3Alists.digium.com

please exert effort before sending a question to the list.

-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] Problem Loading asterisk_oh323-0.6.3b eith last *cvs...

2004-09-06 Thread Rafael J. Risco G.V

Hello
I´ve just install last cvs version (Mon Sep  6) of Asterisk with asterisk-oh323-0.6.3b and pwlib-v1_6_6-src.tar.gz, openh323-v1_13_5-src.tar.gz and .
 
this is the error loading asterisk with chan_oh323 module::
 
[cdr_csv.so] => (Comma Separated Values CDR Backend) [cdr_manager.so] => (Asterisk Call Manager CDR Backend)  == Parsing '/etc/asterisk/cdr_manager.conf': Found [format_sln.so] => (Raw Signed Linear Audio support (SLN))  == Registered file format sln, extension(s) sln|raw [chan_oh323.so]Sep  6 16:58:13 WARNING[1076236928]: loader.c:248 ast_load_resource: /usr/local/lib/liboh323wrap.so: undefined symbol: _ZNK26H323CapabilityRegistration8GetClassEjSep  6 16:58:13 WARNING[1076236928]: loader.c:429 load_modules: Loading module chan_oh323.so failed!
 
any idea?
 
thank you 
Rafael
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[Asterisk-Users] Codecs for fax traffic

2004-09-06 Thread Eric Jacksch

Are there any codecs that are particularly good for fax traffic?  Any to avoid?
---
Eric Jacksch
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Voicetronix OpenSwitch12

2004-09-06 Thread Lex Lethol

hi Flynn,

I have an OpenLine4 on my setup.  Everything appears to work finw and
I am not having the hangup detect but I am having problems when
voicemail tries to record via vpb channel.  Did you ever have that on
your OpenLine4?

I have not tried out the OpenSwitch12 but I am a bit scared with
voicetronix due to the lack of support and friendliness when debugging
any problem that comes up.

Just my 2 cents.

Lethol

On Mon, 06 Sep 2004 16:01:05 +0800, el Flynn <[EMAIL PROTECTED]> wrote:
> Hi all,
> 
> I used to have an OpenLine4 card, but decided against using it due to
> some problems with hangup detect. Does anyone on the list actively use
> Voicetronix's OpenSwitch12? What are your opinions on the card?
> 
> Cheers,
> Flynn
> 
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[Asterisk-Users] Asterisk vs. other PBX config

Hi there,
I noticed, that in my other PBX, I can (and have to) set my local area 
code and the "master number" (in our case a local 5 digit number that 
prefixes the extensions) of our PRI EuroISDN line. I imagine that this 
has some sort of use to those softwares. Is there a place where I have 
to put this information in asterisk config files?

Regards,
Oliver.
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[Asterisk-Users] what does the prilocaldialplan do?

Hi there,
there is an option in zaptel.conf where you can configure a 
"prilocaldialplan" to national, local, international and more.

What does that do?
I've tried to make sense of the source code, but I can't figure it out.
Regards,
Oliver.
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[Asterisk-Users] incoming number truncated

Hi there,
we have a strange problem with a E100P Digium card at a Deutsche 
Telekom S2M port. (Asterisk 1.0rc2)

We are calling from a number in the same local area code and there 
seems to be only the 6 most significant numbers of the target adress 
arrive in Asterisk.

For example, we are calling 9123 and the CLI shows only the 91 
and tries to match that with a extension rule.

(We've set up a rule for _9XXX with StripMSD(5) and a rule for 
extension 123.)

The strange thing is that if I dial the number with the area code (i.e. 
089/9123) or from a mobile fone it works and properly connects to 
the correct extension.

We had other pbxs on the same S2M line that work correctly.
Anyone else noticed this problem?
Regards,
Oliver.
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Re: [Asterisk-Users] cvs server problem

2004-09-06 Thread William Suffill

On Mon, 06 Sep 2004 13:22:51 +0300, Vladyslav <[EMAIL PROTECTED]> wrote:
> Today morning cvs server checkout problem:
> 
> cvs server: Updating asterisk-addons/format_mp3
> cvs server: failed to create lock directory for
> `/usr/cvsroot/asterisk-addons/format_mp3'
> (/usr/cvsroot/asterisk-addons/format_mp3/#cvs.lock): Permission denied
> cvs server: failed to obtain dir lock in repository
> `/usr/cvsroot/asterisk-addons/format_mp3'
> cvs [server aborted]: read lock failed - giving up
> 
> --
> Best regards
> Vlad
> 
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try again they should be round robin still and probably be fixed by now
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Re: [Asterisk-Users] Re: Newby question. Basic structure

Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>, Eric Wieling <[EMAIL PROTECTED]> wrote:
Rodolfo Grave wrote:

Can you explain further what a FXS and FXO port represents in a call 
process in general?
FXO port - Expects to RECEIVE dialtone and ring voltage
FXS port - Expects to PROVIDE dialtone and ring voltage

Or in other words, an FXO port connects to an analogue PSTN line,
and an FXS port connects to an analogue telephone.
And FXS port can also connect to an analog CO port of your PBX.  An 
FXO port could connect to an analog extension on your PBX too. 
Thinking of PSTN lines and telephones can limit the thinking of a 
newbie to just PSTN lines and analog phones.  There are MANY ways you 
can use FXS and FXO ports to connect devices togather.
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RES: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual.

2004-09-06 Thread miguel

Gonzalo, 

I have an APA III-4FXO and I tried using your configurations, I received the
message below:  

-- Executing Dial("SIP/2010-edfc", "SIP/[EMAIL PROTECTED]") in new stack
Sep  6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x814bf0c (len 774) to 192.168.199.5 returned -1: Operation not permitted
-- Called [EMAIL PROTECTED]
Sep  6 16:54:54 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x814409c (len 360) to 192.168.199.5 returned -1: Operation not permitted
  == Spawn extension (from-sip, 92217008, 1) exited non-zero on
'SIP/2010-edfc'
Sep  6 16:54:56 WARNING[1125350192]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x814409c (len 360) to 192.168.199.5 returned -1: Operation not permitted
Sep  6 16:54:57 WARNING[1125350192]: chan_sip.c:673 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 102 (Critical Request)
Sep  6 16:54:59 WARNING[1125350192]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x814409c (len 360) to 192.168.199.5 returned -1: Operation not permitted
Sep  6 16:55:00 WARNING[1125350192]: chan_sip.c:673 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 102 (Non-critical Request)

My configurations are:

SIP.Mib

sipUAServerStaticRegistrarHost = 192.168.199.4
sipUAServerStaticRegistrarPort = 5060
sipUAServerStaticProxyHost = 192.168.199.4
sipUAServerStaticProxyPort = 5060
sipUAServerStaticOutboundProxyHost = 192.168.199.4
sipUAServerStaticOutboundProxyPort = 5060

sipUA1PrefixCCAndAC = 0
sipUA1MainAlias = 
sipUA1FriendlyName = 
sipUA1OtherAliases = 
sipUA1MustUseSessionTimers = 0
sipUA1MaximumSessionExpirationDelay = 60
sipUA1AuthUsrPwd = 
sipUA1AuthValid = 1

My sip.conf and extension.conf are the same that you send.

Any help will be appreciated.

Kind regards,

Miguel 

>Date: Sat, 4 Sep 2004 15:38:42 -0700 (PDT)
>From: Gonzalo Gasca Meza <[EMAIL PROTECTED]>
>Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help.>
>   Anyone with user manual
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>   <[EMAIL PROTECTED]>
>Message-ID: <[EMAIL PROTECTED]>
>Content-Type: text/plain; charset="us-ascii"
>
>
>Here is my configuration for MEdiatrix 1204, by default the 1204 strips one
>digit, so it is not necessary to use:
>
>To dial OUTSIDE
>
>EXTENSIONS.CONF
>
>[locales]
>;ignorepat => 9

exten => _9,1,Dial(SIP/[EMAIL PROTECTED])
exten => _9,2,Congestion
exten => _9,102,Congestion

To receive calls

[from-pstn]
;Incoming calls from Mediatrix 1204, the 1204, sends an invite to
[EMAIL PROTECTED]


exten => ,1,Dial(SIP/100,20)
exten => ,2,Voicemail(u100)
exten => ,102,Voicemail(b100)
exten => ,103,Hangup



***

SIP.CONF

;Mediatrix Telecomm 1204
[Mediatrix]
type=peer
host=110.10.200.10
mask=255.255.255.255
context=from-sip
qualify=yes
canreinvite=yes
disallow=g729
nat = yes

In MEdiatrix 1204 use a program called Unit Manager Network a Configure the
first port as extension  for port 1, in option SIP. as user agent. also
edit registar an dproxy SIP as the IP address of Asterisk.

Works VERY GOOD with one line, although i have seen some scenarios with more
than 1 line which experince problems.




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Re: [Asterisk-Users] Re: Newby question. Basic structure

Thanks.
Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>, Eric Wieling <[EMAIL PROTECTED]> wrote:
 

Rodolfo Grave wrote:
   

Can you explain further what a FXS and FXO port represents in a call 
process in general?
 

FXO port - Expects to RECEIVE dialtone and ring voltage
FXS port - Expects to PROVIDE dialtone and ring voltage
   

Or in other words, an FXO port connects to an analogue PSTN line,
and an FXS port connects to an analogue telephone.
Cheers
Tony
 


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[Asterisk-Users] Re: Newby question. Basic structure

2004-09-06 Thread Tony Mountifield

In article <[EMAIL PROTECTED]>, Eric Wieling <[EMAIL PROTECTED]> wrote:
> Rodolfo Grave wrote:
> 
> > Can you explain further what a FXS and FXO port represents in a call 
> > process in general?
> 
> FXO port - Expects to RECEIVE dialtone and ring voltage
> FXS port - Expects to PROVIDE dialtone and ring voltage

Or in other words, an FXO port connects to an analogue PSTN line,
and an FXS port connects to an analogue telephone.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Re: Asterisk Conferencing using g729

2004-09-06 Thread Tony Mountifield

In article <[EMAIL PROTECTED]>,
William Suffill <[EMAIL PROTECTED]> wrote:
> Good call Daniel I didn't even notice that.
> 
> As far as number of license it really depends on how many concurrent
> calls you will be doing and if asterisk needs to transcode at all. If
> you call from g729 device to g729 you are fine but g729 to vm would be
> 1 license etc.

And if you are conferencing, you need one G.729 licence for each
conference participant, because Asterisk can't mix G.729 natively,
so it transcodes each channel from G.729 to Signed Linear.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Newby question. Basic structure

2004-09-06 Thread mo moe

hi guys i,m  very new to asterisk and i need a little help with it i just 
installed and configered the asterisk and i'm trying to dial from outside 
and get to my voicemail but  this what i'm getting from my asterisk box{Sep  
6 13:08:58 NOTICE[-1093239888]: chan_sip.c:3922 sip_reg_timeout}so please 
can someone help me out
thank you all.

From: Eric Wieling <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
<[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Newby question. Basic structure
Date: Mon, 06 Sep 2004 15:05:52 -0500

Rodolfo Grave wrote:
Can you explain further what a FXS and FXO port represents in a call 
process in general?
FXO port - Expects to RECEIVE dialtone and ring voltage
FXS port - Expects to PROVIDE dialtone and ring voltage
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Re: [Asterisk-Users] Newby question. Basic structure

Rodolfo Grave wrote:
Can you explain further what a FXS and FXO port represents in a call 
process in general?
FXO port - Expects to RECEIVE dialtone and ring voltage
FXS port - Expects to PROVIDE dialtone and ring voltage
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Re: [Asterisk-Users] Newby question. Basic structure

I was checking on the harware page of Digium and I found that there are
many TDM cards.

TDM10B - 1-port FXS bundle

Order Online

TDM40B - 4-port FXS bundle

Order
Online

TDM01B - 1-port FXO bundle

Order
Online

TDM04B - 4-port FXO bundle

Order
Online

TDM11B - 1-port FXS & 1-port FXO bundle

Order
Online

TDM22B - 2-port FXS & 2-port FXO bundle

Order
Online

TDM31B - 3-port FXS & 1-port FXO bundle

Order
Online

Can you explain further what a FXS and FXO port represents in a call
process in general?

More especifically, if I want to connect 2 * boxes through internet,
and I want box 1 to be able to receive a call from PSTN and allow the
caller to make a call trhough VoIP by dialing in his phone the desired
number, and then the box 2 should be able to receive a call through IP
and send it out to PSTN... what's the right configuration? I just want
this for personal use so minimum options is best althought if you have
the time and can explain every posibility will be great.

Regards and thanks once more.

RODOLFO

Rich Adamson wrote:

Is there a possible configuration in case I dont have a broadband
connection in the called-end, for example, a modem connection?

No, there is no modem support built into asterisk. The problem is that
modems typcially do not support the bandwidth needed for a single
phone call. About the minimum bandwidth needed (including IP overhead)
is roughly 30,000 bits/second full duplex.

Could I cheat Asterisk by connecting the linux boxing it's running to a
LAN which is connected to the internet using a
Modem? In this way, Asterisk will see a broadband connection... this is
assuming I dont care about the problem with
the bandwith of course.

Sure you could. Asterisk doesn't care as long as the remote destination
is accessible via IP, and the path/tunnel (regardless of what its built
on or with) has sufficient bandwidth to handle the traffic.

>From the sound of the questions, NATing and plain old IP routing are
probably some topics to review in general. Asterisk can handle NATing,
but unless you read up on exactly how to configure it for your
specific case (if needed), it's likely to be less then clear initially.

Rich

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Re: [Asterisk-Users] Asterisk Conferencing using g729

2004-09-06 Thread William Suffill

Good call Daniel I didn't even notice that.

As far as number of license it really depends on how many concurrent
calls you will be doing and if asterisk needs to transcode at all. If
you call from g729 device to g729 you are fine but g729 to vm would be
1 license etc.


On Mon, 06 Sep 2004 04:51:26 -0500, Daniel Jimenez <[EMAIL PROTECTED]> wrote:
> 
> 
> box100 wrote:
> 
> > My iax.conf file includes the following under the general section
> 
> A SIPURA is a SIP device, configure the codecs under sip.conf not IAX.conf.
> 
> > disallow=all
> > bandwidth=low
> > allow=g729
> > allow=ulaw
> >
> > Thanks,
> > Roger Easlick
> >
> > 
> > 
> >
> > ___
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> --
> Daniel Jimenez 
> 
> 
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Re: [Asterisk-Users] SIP rtp port forcing

2004-09-06 Thread Karl Brose

You can only restrict the range of ports used, in rtp.conf.
I suppose restricting it to 2 ports starting on even number might do it,
but if you're not using SIP on one end, how are you going to start a call?
You need to have at least rudimentary call control for SIP invite and SDP
exchange, and given that you now have SDP exchange you should be able
to accept any port presented by asterisk.
[EMAIL PROTECTED] wrote:
When a SIP call starts (INVITE / 200 OK), asterisk seems to create a 
random port number for voice (rtp) packets. Is it possible to force 
this port value (without using reinvite since i am trying to use SIP 
against something else than sip)

thanks a lot in advance

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[Asterisk-Users] only hear a few ring tones

2004-09-06 Thread Matthew Boehm

Hey gang,
 Sometimes when I call my cell thru * I only hear the "ringing" tone once or
twice then its completly silent until I pick up my cell phone and answer my
call.  Any ideas on why I only hear a few ring tones?

Thanks,
Matthew

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RE: [Asterisk-Users] VM access

Brad,

I like the idea that no inside extension rings when you want to check VM
from an outside line.

Thanks,
Larry

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brad Ediger
Sent: Monday, September 06, 2004 1:51 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VM access

Larry,
I have my extensions.conf set up to wait for a * before ringing the inside
phone, like this:

[incoming]
exten => ,1,Answer
exten => ,2,Ringing
exten => ,3,ResponseTimeout(2)

; Cover * for voicemail access
exten => *,1,VoiceMailMain(1)

; Ring IAXy after ResponseTimeout = 2 sec.
exten => t,1,Dial(IAX2/[EMAIL PROTECTED],20)
exten => t,2,Voicemail(u1)

This way I can dial in and hit * at the first ring to access VM without
ringing the inside line. DISA and the like could be achieved similarly.

Brad



  - Original Message -
  From: Larry Shields
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
  Sent: Monday, September 06, 2004 11:48 AM
  Subject: [Asterisk-Users] VM access


  Can someone tell me how to get to a mailbox login prompt when accessing
the Asterisk VM remotely via a PSTN line?  I am running version CSV 8/25/04.

  Thanks,
  Larry

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Re: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual

2004-09-06 Thread Bob Knight

Jamie Carl wrote:
Bob Knight wrote:
There is a linux package called mbrowse that you can use with your 
mediatrix mibs.
I can get and walk everything in my 1204's.
For some reason I have not had any success with writes, but I have not 
spent
that much time on it.

I don't even have the MIBs which is half the problem.  I can do certain 
things using windoze SNMP software, but not exactly being a guru on SNMP 
i'm guessing that without the MIBs i'm pretty much stuffed.

Anyone with MIBs they can send me?  hehe  Please? :)
I have MIBs for whatever version I am running that I am more than
happy to share.  Anyone know where I can place these for public access.
Sort of like the freedomphones site for Polycom.  We could then
put pointers on the wiki.
Thanks for the info tho.  If mbrowse is console based it will be very 
useful. :)
It has gui (X, gtk I think) if that is what you mean by console based.
I can ssh into a remote * server and do get walks on my 1204's.
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
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Re: [Asterisk-Users] SIP rtp port forcing

2004-09-06 Thread Kannaiyan Natesan




check   rtp.conf
 
-Kannaiyan

  - Original Message - 
  From: 
  [EMAIL PROTECTED] 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Monday, September 06, 2004 6:15 
  PM
  Subject: [Asterisk-Users] SIP rtp port 
  forcing
  When a SIP call starts (INVITE 
  / 200 OK), asterisk seems to create a random port number for voice (rtp) 
  packets. Is it possible to force this port value (without using reinvite since 
  i am trying to use SIP against something else than sip) thanks a lot in advance 
  
  

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Re: [Asterisk-Users] VM access

2004-09-06 Thread Brad Ediger

Larry,
I have my extensions.conf set up to wait for a * before ringing the inside phone, like 
this:
[incoming]
exten => ,1,Answer
exten => ,2,Ringing
exten => ,3,ResponseTimeout(2)
; Cover * for voicemail access
exten => *,1,VoiceMailMain(1)
; Ring IAXy after ResponseTimeout = 2 sec.
exten => t,1,Dial(IAX2/[EMAIL PROTECTED],20)
exten => t,2,Voicemail(u1)
This way I can dial in and hit * at the first ring to access VM without ringing the 
inside line. DISA and the like could be achieved similarly.
Brad

 - Original Message - 
 From: Larry Shields 
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
 Sent: Monday, September 06, 2004 11:48 AM
 Subject: [Asterisk-Users] VM access

 Can someone tell me how to get to a mailbox login prompt when accessing the 
Asterisk VM remotely via a PSTN line?  I am running version CSV 8/25/04.
 Thanks,
 Larry
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Re: [Asterisk-Users] Asterisk & sudo from httpd

2004-09-06 Thread Matthew Boehm

thats about the most unsecure thing I've ever seen.  there is a reason you
don't run apache as root and therefore having a script that sudo's is just
as bad.

try using the manager interface for better security.  * shouldn't be running
as root either if we want to get nitty-gritty about security.

Matthew

- Original Message - 
From: "Roland Zagler" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, September 05, 2004 4:52 PM
Subject: [Asterisk-Users] Asterisk & sudo from httpd


Hello!

I want to use "asterisk -rx "show version"" from a php script called in
the browser using the local apache, which runs as user "apache".
Asterisk is running as root.

I added the following line to /etc/sudoers using visudo:

 apacheALL = NOPASSWD: /usr/sbin/asterisk

When i am on the command line of my linux box it looks like this:


# sudo /usr/sbin/asterisk -rx "show version"

Asterisk 1.0-RC2 built by [EMAIL PROTECTED] on a i686 running
Linux

# sudo -u apache /usr/sbin/asterisk -rx "show version"

Unable to connect to remote asterisk


"strace" showed me that there is an access problem with
"/var/run/asterisk.ctl":


munmap(0xbf334000, 4096)= 0
socket(PF_FILE, SOCK_STREAM, 0) = 3
connect(3, {sa_family=AF_FILE, path="/var/run/asterisk.ctl"}, 110) = -1
EACCES (Permission denied)
close(3)= 0
time([1094419366])  = 1094419366
fstat64(1, {st_mode=S_IFCHR|0620, st_rdev=makedev(136, 0), ...}) = 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1,
0) = 0xbf334000
write(1, "Unable to connect to remote aste"..., 37) = 37
munmap(0xbf334000, 4096)= 0
exit_group(1)   = ?


System description:
Fedora Core 1
Kernel 2.4.22
Sudo 1.6.7p5
Apache httpd 2.0.50
Asterisk 1.0-RC2

Can anyone please help?

Thank you in advance!


Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
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RE: [Asterisk-Users] VM access




Ok I think I found what I am looking for.  You 
need to added two items to your configs.
 
; Add the following context to your extensions.conf 
(2001 is the main mailbox that all my messages are left 
in)
 
[vmlogin]
 
exten => 
a,1,VoicemailMain(2001)exten => a,2,Hangup
 
exten => 
i,1,Hangupexten => t,1,Hangupexten => h,1,Hangup
; Add the following to the voicemail.conf to define 
what extensions context will handle the * key or 0 key while in the voicemail 
application.
 
; Exit to specified context after the 
user presses * or 0exitcontext=vmlogin
 
 
Now 
when listening to my outgoing greeting, I can press the * key and get my mailbox 
login prompt.
 
--Larry

 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Larry 
ShieldsSent: Monday, September 06, 2004 12:14 PMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: 
[Asterisk-Users] VM access

On most VM systems you can press the * key or # key to get 
a login prompt during your greeting.  Is that not possible with this 
system?
 
Thanks,
Larry


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle 
GieseSent: Monday, September 06, 2004 11:55 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] VM access

You could dedicate a PSTN line(& phone number) 
for that purpose.  You could put a menu system(auto-attendant style) and 
just dial 8500(demo is set for this exten to be the gateway to VM).  Or if 
your operator answers, have her transfer your call to 8500.
 
Lyle

  - Original Message - 
  From: 
  Larry 
  Shields 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Monday, September 06, 2004 11:48 
  AM
  Subject: [Asterisk-Users] VM access
  
  Can someone tell 
  me how to get to a mailbox login prompt when accessing the Asterisk VM 
  remotely via a PSTN line?  I am running version CSV 
  8/25/04.
   
  Thanks,
  Larry
  
  

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Re: [Asterisk-Users] Newby question. Basic structure

Well, I wont get tired of saying thank you for the quick answers.

Rich Adamson wrote:

Is there a possible configuration in case I dont have a broadband
connection in the called-end, for example, a modem connection?

That's great.

Rich

NAT is the most secure option can you give some hints/references
about configuring Asterisk for working behind a NAT router?

RODOLFO

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Re: [Asterisk-Users] VM access

2004-09-06 Thread James Cloos

> "Larry" == Larry Shields <[EMAIL PROTECTED]> writes:

Larry> On most VM systems you can press the * key or # key to get a
Larry> login prompt during your greeting.  Is that not possible with
Larry> this system?

If you hist * during the outgoing message you'll get sent to the a
extension, if that exists in the current context.

Eg:

OPERATOR => 2121
; dump them to vm
exten => s,1,VoiceMail2(2345)
; if they enter 0
exten => o,1,Dial(OPERATOR)
; if they enter *
exten => a,1,Goto(ivr|2345|1)

-JimC
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Re: [Asterisk-Users] VM access

2004-09-06 Thread William Suffill

It is but you need to modify your dial plan to make it work.

I do it like such
[inbound] ; context that takes inbound calls and matches em and routes according
exten => 91808,1,Macro(stdexten,101,SIP/101) ; fwd
exten => 55,1,Goto(all-exten,101,1) 
; fwd goes start to my stdexten to 101 which doesn't have the options
to press * at vm
; the did goes to all-exten which has some added dialplan features to
make * work for vm

[all-exten]
exten => 0,1,Macro(stdexten,0,SIP/0)
exten => _[1-6]XX,1,SetVar(VMBX=${EXTEN})
exten => _[1-6]XX,2,NoOp(${VMBX})
exten => _[1-6]XX,3,Macro(stdexten,${EXTEN},${EXTEN})
exten => a,1,VoicemailMain(${VMBX})
exten => a,2,Hangup

the a extension is called when * is pressed while in vm . The macro
returns to the context and goes to the a extension



- Original Message -
From: Larry Shields <[EMAIL PROTECTED]>
Date: Mon, 6 Sep 2004 12:14:26 -0500
Subject: RE: [Asterisk-Users] VM access
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[EMAIL PROTECTED]>

 
On most VM systems you can press the * key or # key to get a login
prompt during your greeting.  Is that not possible with this system?
  
Thanks, 
Larry
 
 
 From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle
Giese
Sent: Monday, September 06, 2004 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VM access



 
 
You could dedicate a PSTN line(& phone number) for that purpose.  You
could put a menu system(auto-attendant style) and just dial 8500(demo
is set for this exten to be the gateway to VM).  Or if your operator
answers, have her transfer your call to 8500.
  
Lyle 
 
- Original Message - 
From: Larry Shields 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Sent: Monday, September 06, 2004 11:48 AM 
Subject: [Asterisk-Users] VM access 

 
Can someone tell me how to get to a mailbox login prompt when
accessing the Asterisk VM remotely via a PSTN line?  I am running
version CSV 8/25/04.
  
Thanks, 
Larry 

 
 

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[Asterisk-Users] multiline IP hardphone w/ FDX speakerphone?

2004-09-06 Thread Stewart Nelson

Could someone please recommend a reasonably priced IP phone
that works well with *, has a decent (full duplex, echo canceling)
speakerphone, has at least two line appearances, and can
transfer / conference reliably?
The Wiki lists 35 brands of hardphone, but: 
1. Most seem to be toys.
2. For many, there is no info on e.g. speakerphone characteristics.
3. When one seems technically promising, e.g. Polycom IP500, there
  are *lots* of negative postings about support, integration, etc.

Is there anything decent out there for < $200?
Thanks,
Stewart
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Re: [Asterisk-Users] Newby question. Basic structure

> Is there a possible configuration in case I dont have a broadband 
> connection in the called-end, for example, a modem connection?
> 
> 
> No, there is no modem support built into asterisk. The problem is that
> modems typcially do not support the bandwidth needed for a single
> phone call. About the minimum bandwidth needed (including IP overhead)
> is roughly 30,000 bits/second full duplex.
>   
> 
> Could I cheat Asterisk by connecting the linux boxing it's running to a 
> LAN which is connected to the internet using a
> Modem? In this way, Asterisk will see a broadband connection... this is 
> assuming I dont care about the problem with
> the bandwith of course.

Sure you could. Asterisk doesn't care as long as the remote destination
is accessible via IP, and the path/tunnel (regardless of what its built 
on or with) has sufficient bandwidth to handle the traffic.

>From the sound of the questions, NATing and plain old IP routing are
probably some topics to review in general. Asterisk can handle NATing,
but unless you read up on exactly how to configure it for your 
specific case (if needed), it's likely to be less then clear initially.

Rich



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RE: [Asterisk-Users] UK Disconnect supervision with TDM400P

2004-09-06 Thread Dan Tucny

On Wed, 2004-09-01 at 22:02, Edward Eastman wrote:
> Hi, thanks for the reply, only just got round to having a look at it again
> (annoying how real life gets in the way of the important stuff ;) 
> 
> I've had a go at ramping up the tx/rx gain but it doesn't seem to make any
> difference.  FWIW it's the same with the module in normal fcc mode.
> 
> Does anyone know if bt do normally provide disconnect supervision or whether
> it has to be done with e.g. busydetect (and can either be detected by the
> tdm400p in uk mode)?
> 
> Thanks
> 
> Ed

> Edward Eastman wrote:
> 
> > 
> > I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN
> line,
> > loading wcfxs with OPERMODE=UK.  All's working well, except if I get an
> > incoming call through my bt line, and the remote party hangs up, I get
> > approx 20secs of the bt line hungup tone before asterisk hangs up, which
> > leads (if nothing else) to the well documented 20secs of beep on vm
> problem
> > :)
> > 
> > I have tried: busydetect=yes / busycount=7 / other busycounts /
> > callprogess=yes but none of these make any difference.  I have
> > loadzone/defaultzone=uk and country=uk in indications.conf and fxs_ks
> > signalling.
> > 
> 
> Try increasing your RX gain in 1db steps, until it reliably hangs up.
> 
> I had a box with X100Ps which busydetected perfectly with default gain 
> settings. When they were replaced with TDM FXOs, busydetect stopped 
> working and I needed 3db of RX gain added to get it working again.
> 
> Regards,
> 
> Richard

Ed,

When someone does hang up on you with your BT line, what do you hear?
Here I get a click/pop following by a 4 second unobtainable tone
followed by a click/pop... The clicks are BT's 'k-break's... It
obviously doesn't seem to be what * expects... Investigating this is
something I'm hoping to have a look at soon, but, if you have time
beforehand

BTW have you used the IRC channel?

Dan
(dant)

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[Asterisk-Users] SIP rtp port forcing

2004-09-06 Thread boris . vincent


When a SIP call starts (INVITE / 200 OK), asterisk seems to create a random port number for voice (rtp) packets. Is it possible to force this port value (without using reinvite since i am trying to use SIP against something else than sip)

thanks a lot in advance
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RE: [Asterisk-Users] VM access




On most VM systems you can press the * key or # key to get 
a login prompt during your greeting.  Is that not possible with this 
system?
 
Thanks,
Larry


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle 
GieseSent: Monday, September 06, 2004 11:55 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] VM access

You could dedicate a PSTN line(& phone number) 
for that purpose.  You could put a menu system(auto-attendant style) and 
just dial 8500(demo is set for this exten to be the gateway to VM).  Or if 
your operator answers, have her transfer your call to 8500.
 
Lyle

  - Original Message - 
  From: 
  Larry 
  Shields 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Monday, September 06, 2004 11:48 
  AM
  Subject: [Asterisk-Users] VM access
  
  Can someone tell 
  me how to get to a mailbox login prompt when accessing the Asterisk VM 
  remotely via a PSTN line?  I am running version CSV 
  8/25/04.
   
  Thanks,
  Larry
  
  

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RE: [Asterisk-Users] spouse-friendly spa-3000 pstn interface

Rich,

At first I thought it may be a setting in the SPA (and maybe it is I don't
know), but I also had this problem when using the SPA-2000 to a x100p
interface.  I don't think it is a problem with the SPA because when I am in
a conversation utilizing the VoIP trunk (VoicePulse), and a second call
comes in I get CW/CID and when I send the flash it correctly toggles between
both parties.  

It is only when using the PSTN connection via x100p or SPA-3000 that I get
CW/CID and when I send the flash it gives me secondary dial-tone.  It seems
like Asterisk, which sits in-between, is misinterpreting why I am sending
the flash and thinks I want to xfer the call.  Under normal circumstances I
would want to get second dial-tone to xfer the call (though not necessary
since I use the # key to initiate xfer), but not when the system sees that
there is a "call-waiting" condition for that extension. In the case of a
waiting call I would want it to send the flash to the PSTN trunk and connect
me to the party that is waiting. I have not seen too many people complaining
about this so maybe it's just something I've mis-configured.

--Larry

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Monday, September 06, 2004 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] spouse-friendly spa-3000 pstn interface

> I have an SPA-3000 with a similar setup to yours.  Unlike yours, my 
> cordless phone system is on a SPA-2000 that connect through the 
> SPA-3000 for a PSTN connection. For LD I use 91xxx for (MCI) or 71xxx 
> for (VoicePulse).  In addition to trunk access codes 7 & 9, I use FWD by
dialing 8xx.
> 
> I also recently replaced our 2.4GHz phones with 5.8GHz phones.  I 
> picked the Uniden TRU8866 5.8GHz cordless system (I highly recommend 
> it).  It supports up to 10 handsets on one base unit and has excellent 
> audio quality.  What really sold me on this cordless phone system, 
> besides the audio quality, was the fact that it supports two lines.  I 
> have connected both lines to the SPA-2000 which allows us to utilize 
> the local PSTN and VoIP trunk simultaneously.
> 
> I like your post because I have also gone to great lengths to 
> integrate the
> * system into our home without alienating my wife or visiting friends 
> & family.  Fortunately I have been able to talk my wife into using the 
> * voicemail functions.
> 
> At first my wife was not too keen on the VM system.  It was the Uniden 
> sets that sold her on it.  The Uniden system supports Message Waiting 
> Indicator
> (MWI) when messages are left on the * system.  So she can see from any 
> handset (red slow flashing LED) that a VM message is waiting.  The 
> phone base and handsets also have a feature button marked for checking 
> VM.  So I programmed the button on all the cordless handsets to dial 
> the VM extension 2500#, which then prompts for the mailbox password.  
> So far so good, no echo and the clarity is excellent whether talking to a
PSTN or VoIP call.

We're using the Panasonic 2-line multi-remote cordless 2gig phones.

> The only problem I have not been able to figure out is flashing to the 
> PSTN line for call-waiting... It does not work and I get secondary 
> dial tone instead of the waiting call.  Any Ideas?

I've not got that far to test any of the flash functions, but will have the
same issue here (eg, call-waiting). Best guess is that 'should' work as the
spa's do have parameters to detect it (on the fxs side), and do something
with it. Since the default path is via the pstn side, it would seem like
that flash would be replicated through to the pstn. But obviously that's a
guess.

Saying the same thing in reverse, the spa engineers have addressed most of
the US calling functions and they addressed fxs flash. Since the flash
function is used for so many pstn purposes, if you were an spa engineer,
what would you do with it? (I'd have to guess its not documented very
clearly anywhere, but they're doing something it.)

Rich


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Re: [Asterisk-Users] Newby question. Basic structure

Thanks again Rich. I hope you dont mind to answer a few more
questions

My response is inline...

Hi! and thanks a million for your answer. You hace cleared many of the
doubts I had, including the differences between the cards. At the same
time, new questions has arised:

Is there a possible configuration in case I dont have a broadband
connection in the called-end, for example, a modem connection?

Could I cheat Asterisk by connecting the linux boxing it's running to a
LAN which is connected to the internet using a Modem? In this way,
Asterisk will see a broadband connection... this is assuming I dont
care about the problem with the bandwith of course.

Is it possible to set a route for the IP packages? This is to optimize
the packets transmission over internet.

I'm not sure what you mean with the above question. The voip packets
will typically be standard old UDP packets, and those packets are routed
over the Internet like any other IP packet based on whatever the ISP's
have elected to do. You don't have any choice on how those are routed
in any case.

That's exactly what I asked

I'm outside US, so, why should I use tdm instead of x100p?

The chip set on the x100p was designed specifically to match the US
telco standards. The card will work in some countries, but other
countries you'll hear a lot of echo, callerid may not work, etc, etc.
The TDM card has a much newer chip set that was intended to be used
in many different countries around the world. Therefore, stay with
the TDM (as opposed to the x100p).

Rich

OK. And thanks again.

RODOLFO

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[Asterisk-Users] DTMF information?

2004-09-06 Thread Chris Lee

I am looking at building an IVR product with a few interesting features 
and need some more information about how asterisk and VoIP work and what 
I can get from them.

As far as I can tell when I use ISDN/GSM telephone networks the DTMF 
information travels as data representing 'start tone' and 'stop tone' 
for each button pressed, it is then generated at the other end if an 
audio representation is required.
I am interested to know if I can get access to these events 'start tone' 
and 'stop tone' through the dialplan or an AGI or by acting as a VoIP 
device. Or of course if I am completely off track and should give up now.

I am looking to get the length of time a button was held down rather 
than that it was pressed.

Thanks for any help
Chris.
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Re: [Asterisk-Users] VM access

2004-09-06 Thread Lyle Giese




You could dedicate a PSTN line(& phone number) 
for that purpose.  You could put a menu system(auto-attendant style) and 
just dial 8500(demo is set for this exten to be the gateway to VM).  Or if 
your operator answers, have her transfer your call to 8500.
 
Lyle

  - Original Message - 
  From: 
  Larry 
  Shields 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Monday, September 06, 2004 11:48 
  AM
  Subject: [Asterisk-Users] VM access
  
  Can someone tell 
  me how to get to a mailbox login prompt when accessing the Asterisk VM 
  remotely via a PSTN line?  I am running version CSV 
  8/25/04.
   
  Thanks,
  Larry
  
  

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[Asterisk-Users] VM access




Can someone tell me 
how to get to a mailbox login prompt when accessing the Asterisk VM remotely via 
a PSTN line?  I am running version CSV 8/25/04.
 
Thanks,
Larry
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RE: [Asterisk-Users] spouse-friendly spa-3000 pstn interface

> I have an SPA-3000 with a similar setup to yours.  Unlike yours, my cordless
> phone system is on a SPA-2000 that connect through the SPA-3000 for a PSTN
> connection. For LD I use 91xxx for (MCI) or 71xxx for (VoicePulse).  In
> addition to trunk access codes 7 & 9, I use FWD by dialing 8xx.
> 
> I also recently replaced our 2.4GHz phones with 5.8GHz phones.  I picked the
> Uniden TRU8866 5.8GHz cordless system (I highly recommend it).  It supports
> up to 10 handsets on one base unit and has excellent audio quality.  What
> really sold me on this cordless phone system, besides the audio quality, was
> the fact that it supports two lines.  I have connected both lines to the
> SPA-2000 which allows us to utilize the local PSTN and VoIP trunk
> simultaneously.
> 
> I like your post because I have also gone to great lengths to integrate the
> * system into our home without alienating my wife or visiting friends &
> family.  Fortunately I have been able to talk my wife into using the *
> voicemail functions.  
> 
> At first my wife was not too keen on the VM system.  It was the Uniden sets
> that sold her on it.  The Uniden system supports Message Waiting Indicator
> (MWI) when messages are left on the * system.  So she can see from any
> handset (red slow flashing LED) that a VM message is waiting.  The phone
> base and handsets also have a feature button marked for checking VM.  So I
> programmed the button on all the cordless handsets to dial the VM extension
> 2500#, which then prompts for the mailbox password.  So far so good, no echo
> and the clarity is excellent whether talking to a PSTN or VoIP call.

We're using the Panasonic 2-line multi-remote cordless 2gig phones.

> The only problem I have not been able to figure out is flashing to the PSTN
> line for call-waiting... It does not work and I get secondary dial tone
> instead of the waiting call.  Any Ideas?

I've not got that far to test any of the flash functions, but will have
the same issue here (eg, call-waiting). Best guess is that 'should' work
as the spa's do have parameters to detect it (on the fxs side), and do
something with it. Since the default path is via the pstn side, it would
seem like that flash would be replicated through to the pstn. But
obviously that's a guess.

Saying the same thing in reverse, the spa engineers have addressed most
of the US calling functions and they addressed fxs flash. Since the flash
function is used for so many pstn purposes, if you were an spa engineer,
what would you do with it? (I'd have to guess its not documented very
clearly anywhere, but they're doing something it.)

Rich


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[Asterisk-Users] MEdiatrix APA 111-4FXO/FXS manual

2004-09-06 Thread Peter Mwondi

Hello there,

I have just laid my hands on a pair of used Mediatrix APA III-4FXO/FXS VoIP
gateways. Anyone with the manuals so that I can configure them for use on my
Asterisk SIP server ?

Peter Mwondi


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Re: [Asterisk-Users] Newby question. Basic structure

My response is inline...

> Hi! and thanks a million for your answer. You hace cleared many of the 
> doubts I had, including the differences between the cards. At the same 
> time, new questions has arised:
> 
> Is there a possible configuration in case I dont have a broadband 
> connection in the called-end, for example, a modem connection?

No, there is no modem support built into asterisk. The problem is that
modems typcially do not support the bandwidth needed for a single
phone call. About the minimum bandwidth needed (including IP overhead)
is roughly 30,000 bits/second full duplex.
 
> Is it possible to set a route for the IP packages? This is to optimize 
> the packets transmission over internet.

I'm not sure what you mean with the above question. The voip packets
will typically be standard old UDP packets, and those packets are routed
over the Internet like any other IP packet based on whatever the ISP's
have elected to do. You don't have any choice on how those are routed 
in any case.
 
> I'm outside US, so, why should I use tdm instead of x100p?

The chip set on the x100p was designed specifically to match the US
telco standards. The card will work in some countries, but other
countries you'll hear a lot of echo, callerid may not work, etc, etc.
The TDM card has a much newer chip set that was intended to be used
in many different countries around the world. Therefore, stay with
the TDM (as opposed to the x100p).

Rich
 
> Thanks again,
> 
> RODOLFO
> 
> Rich Adamson wrote:
> 
> >>From: Rodolfo Grave <[EMAIL PROTECTED]>
> >>
> >>I've being reading posts from the list since yesterday and I feel this 
> >>question was answered a lot time ago, but the list archives are a mess 
> >>(yet). I hope some one is willing to help me out.
> >>
> >>I want to set up this:
> >>
> >>caller - PSTN  (SOMETHING1) -- VoIP - (SOMETHING2) 
> >> PSTN
> >>
> >>I think this must be a very basic architecture, but I'm not sure wat 
> >>SOMETHING1 and SOMETHING2 are. I've been on this for a while now (around 
> >>two months) and till yesterday I haven't find Asterisk.
> >>
> >>Can you help me? I need to know hardware and software needs for this. I 
> >>have read a few about voIP and have some programming and configuration 
> >>skills under Linux and Windows.
> >>
> >>
> >
> >In terms of using asterisk to implement your diagram, "something1" and
> >"something2" are basic linux boxes equipped with:
> > - pstn interface card (such as x100p, tdm, isdn, T1 card)
> > - ethernet interface (to connect to your broadband internet)
> > - asterisk software
> >
> >The type of pstn interface card to use will be dependent upon how
> >many "simultanous" phone conversations you'd like to support. The
> >x100p card is a single pstn line interface; the tdm card supports
> >one to four (tdm04b bundle) pstn lines (the tdm card needs to be
> >purchased with fxo interface modules); the isdn card supports from
> >one to 23 (?) pstn logical interfaces (depending upon the exact
> >card purchased); or the T1/E1 card supports from 1 to 24 (T1) or
> >1 to 32 (E1-?) conversations. You'll find most of those cards at
> >www.digium.com under Hardware Products.
> >
> >If you are outside the US, consider the TDM card as opposed to
> >the x100p card.
> >
> >The size of Linux system will also be dependent on how many 
> >simultanous calls you want to support, and exactly how you have
> >the system configured. For a single call, some folks have it
> >running on old 300 mhz (and slower) box, while high traffic
> >volumes will require a much faster system.
> >
> >The asterisk software can be found on the digium home page 
> >(lower-left menu option), which points to www.asterisk.org
> >
> >Also, lots of good reference material at www.voip-info.org
> >
> >After you've configured your system with your favorite Linux
> >distro, download and compile asterisk per the instructions found
> >on the www.asterisk.org site. Pay attention to the last steps
> >on your screen that copies configuration samples to /etc/asterisk.
> >Read through those configuration files, read the info at
> >www.voip-info.org, and you should be able to get a system
> >running.
> >
> >
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> 
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RE: [Asterisk-Users] spouse-friendly spa-3000 pstn interface

 
Rich,

I have an SPA-3000 with a similar setup to yours.  Unlike yours, my cordless
phone system is on a SPA-2000 that connect through the SPA-3000 for a PSTN
connection. For LD I use 91xxx for (MCI) or 71xxx for (VoicePulse).  In
addition to trunk access codes 7 & 9, I use FWD by dialing 8xx.

I also recently replaced our 2.4GHz phones with 5.8GHz phones.  I picked the
Uniden TRU8866 5.8GHz cordless system (I highly recommend it).  It supports
up to 10 handsets on one base unit and has excellent audio quality.  What
really sold me on this cordless phone system, besides the audio quality, was
the fact that it supports two lines.  I have connected both lines to the
SPA-2000 which allows us to utilize the local PSTN and VoIP trunk
simultaneously.

I like your post because I have also gone to great lengths to integrate the
* system into our home without alienating my wife or visiting friends &
family.  Fortunately I have been able to talk my wife into using the *
voicemail functions.  

At first my wife was not too keen on the VM system.  It was the Uniden sets
that sold her on it.  The Uniden system supports Message Waiting Indicator
(MWI) when messages are left on the * system.  So she can see from any
handset (red slow flashing LED) that a VM message is waiting.  The phone
base and handsets also have a feature button marked for checking VM.  So I
programmed the button on all the cordless handsets to dial the VM extension
2500#, which then prompts for the mailbox password.  So far so good, no echo
and the clarity is excellent whether talking to a PSTN or VoIP call.

The only problem I have not been able to figure out is flashing to the PSTN
line for call-waiting... It does not work and I get secondary dial tone
instead of the waiting call.  Any Ideas?

--Larry

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Monday, September 06, 2004 9:38 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] spouse-friendly spa-3000 pstn interface


This post is simply documenting a spouse-friendly way of using the spa-3000
as both a fxs and fxo port for basic soho environments in the US, allowing
asterisk to participate as needed/wanted.

All home phones are connected _only_ to the spa-3000 fxs port.

The incoming home pstn line is connected _only_ to the spa-3000 fxo port.

Defined Line 1 (fxs) to register with asterisk via sip (extn ), with
silence suppression disabled.

Defined the PSTN Line (fxo) to register with asterisk via sip using a second
sip.conf entry (extn ).

PSTN User, defined PSTN Ring Thru Line 1 Ring Settings as "1".

In Line 1, defined Gateway Account #1 to point to asterisk, and created a
dialplan entry like:
 (*xx|81xxx.<:@gw1>|3xxx<:@gw1>|0<:@gw0>|[2-9]11<:@gw0>|xx.<:@gw0>)
Note: gw0 defaults to the pstn line per the spa-3000 doc.

Result:
1. If asterisk is down for any reason, all incoming pstn home calls still
ring through to the analog house phones.
2. Incoming pstn calls ring through to the house phones without any asterisk
involvement, and callerid is properly passed to the house phones. (Since my
* also has an fxo port attached behind the spa-3000,
* still rings and I can answer the call via it.) 3. Loss of commercial AC
power causes the spa-3000 to physically cut through the pstn line to the
house phones, leaving all house phones operative. Same with loss of an
ethernet physical connection.
4. Calls originating from within asterisk to the house phones use:
 exten => ,1,SetVar(ALERT_INFO=bellcore-r3)
 exten => ,2,Dial(SIP/,25,r)
 exten => ,3,Hangup
and ring the house phone with distinctive rings. Spouce recognizes incoming
pstn calls as different from voip/asterisk calls. (We have about six analog
phones behind the spa-3000 that properly ring on all calls.) 5. Calls placed
from the house phones (via fxs Line 1) use the above dialplan.
 a. LD calls starting with 81xxx. are routed via * to an ITSP (gw1)  b.
normal US local calls routed via pstn line (gw0)  c. calls to 3xxx are
asterisk extensions and are therefor routed
to asterisk (gw1)
 d. 911 (as well as 211, 311, etc) are routed via the pstn line.
 e. 0 calls are routed via pstn line
If spouse is having a problem with dialing LD via 81xxx, then dialing the
same call without the 8 routes the LD call via pstn line.
6. Calls originating from within asterisk can be routed to this pstn line
via Dial(Sip//...) if needed.
7. The old answering machine (that my spouse knows how to use) still accepts
all unanswered calls.

I fully understand the above approach keeps asterisk out of the loop most of
the time, and that was an objective (from a spouse perspective for now). It
represents the least intrusive integration without remedial training, etc.
:)  No need to purchase six sip phones either.

There has been no echo or any other negative issues thus far; haven't tested
a lot of things as yet though. (Eg, what happens when the house phone is
talking v

Re: [Asterisk-Users] Newby question. Basic structure

Hi! and thanks a million for your answer. You hace cleared many of the 
doubts I had, including the differences between the cards. At the same 
time, new questions has arised:

Is there a possible configuration in case I dont have a broadband 
connection in the called-end, for example, a modem connection?

Is it possible to set a route for the IP packages? This is to optimize 
the packets transmission over internet.

I'm outside US, so, why should I use tdm instead of x100p?
Thanks again,
RODOLFO
Rich Adamson wrote:
From: Rodolfo Grave <[EMAIL PROTECTED]>
I've being reading posts from the list since yesterday and I feel this 
question was answered a lot time ago, but the list archives are a mess 
(yet). I hope some one is willing to help me out.

I want to set up this:
caller - PSTN  (SOMETHING1) -- VoIP - (SOMETHING2) 
 PSTN

I think this must be a very basic architecture, but I'm not sure wat 
SOMETHING1 and SOMETHING2 are. I've been on this for a while now (around 
two months) and till yesterday I haven't find Asterisk.

Can you help me? I need to know hardware and software needs for this. I 
have read a few about voIP and have some programming and configuration 
skills under Linux and Windows.
   

In terms of using asterisk to implement your diagram, "something1" and
"something2" are basic linux boxes equipped with:
- pstn interface card (such as x100p, tdm, isdn, T1 card)
- ethernet interface (to connect to your broadband internet)
- asterisk software
The type of pstn interface card to use will be dependent upon how
many "simultanous" phone conversations you'd like to support. The
x100p card is a single pstn line interface; the tdm card supports
one to four (tdm04b bundle) pstn lines (the tdm card needs to be
purchased with fxo interface modules); the isdn card supports from
one to 23 (?) pstn logical interfaces (depending upon the exact
card purchased); or the T1/E1 card supports from 1 to 24 (T1) or
1 to 32 (E1-?) conversations. You'll find most of those cards at
www.digium.com under Hardware Products.
If you are outside the US, consider the TDM card as opposed to
the x100p card.
The size of Linux system will also be dependent on how many 
simultanous calls you want to support, and exactly how you have
the system configured. For a single call, some folks have it
running on old 300 mhz (and slower) box, while high traffic
volumes will require a much faster system.

The asterisk software can be found on the digium home page 
(lower-left menu option), which points to www.asterisk.org

Also, lots of good reference material at www.voip-info.org
After you've configured your system with your favorite Linux
distro, download and compile asterisk per the instructions found
on the www.asterisk.org site. Pay attention to the last steps
on your screen that copies configuration samples to /etc/asterisk.
Read through those configuration files, read the info at
www.voip-info.org, and you should be able to get a system
running.
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Re: [Asterisk-Users] offtopic - channel banks

2004-09-06 Thread Michael Welter

If you are in the US, go to your Graybar store and ask for a short 
"Amp50" cable and a "harmonica".  The harmonica has an Amp50 connector 
on one side and RJ14 jacks on the other.  For the cable, they'll ask you 
the gender of the connectors at each end--you'll have to determine the 
gender of the Amp50 connector on your channel bank.

Chris A. Icide wrote:
Ilia,
Think of a channel bank as a concentrator.  In a single T1 channel bank, 
you concentrate 24 analog two wire phone connections into a single 4 
wire digital interface.

So in your case, the RJ45 connector is for the T1 interface to Asterisk, 
the local CLEC, or whatever you intend to connect it to.  On the T1 
side, there has to be several layers of signalling and encoding.  Alot 
of this information is superfluous, but may help you when it comes to 
understanding your configs.

Before we even talk about E&M signalling, you have the T1 framing and 
encoding.  This is used to allow both ends of the T1 circuit to 
understand how the 24 channels are being configured on the 4 wire 
circuit.  It's generally going to be either sf, d4 or esf, b8zs.  This 
is known as the framing and encoding.

Once those are agreed upon, then we need to set up the way the T1 is 
going to signal across the channels.  Normal phone lines (analog) use 
voltages, resistances, and dtmf to signal what it is doing.  Since a T1 
is a digital circuit we can't do that, so we need to set up another way 
to signal, so that the channel bank knows what to do when we send some 
kind of digital signal.  In this case, this is the E&M signalling you 
asked about.

Finally, you probably are looking for some way to plug your phone's RJ11 
connecter into the channel bank.  Unfortunately it's not that easy.  
That big Centronics style connecter is where you actually have to plug 
up the phone.  there are 24 pairs of contacts in that connecter that are 
associated with each channel on the T1 circuit.  Historically, you would 
connect up a cable of 50 conductors connected to the centronics 
connector on one end, and then to one side a punch down block on the 
other (just a quick-connect style access device for copper wire).  On 
the other side of the punch down block, you would connect the wires that 
would then run to the remote wall jacks, etc. where your phones plug in.

The problem you have is wither by google mastery or just plain brute 
force testing, you need to figure out the pinout of that centronics port 
before you can connect up any phones successfully.

-Chris
On 10:56 PM 9/5/2004, Ilia Mirkin wrote:
 >While I understand everything that you have said, I'm still a little
 >confused. Yes - I have what looks like a centronics connector on the
 >back. So, I can do "t100p with e&m signalling" <-> "act-1241 e&m card"
 ><-> what? Namely, if the E&M card deals with the T1 end of the channel,
 >how do I get that to a real phone? Will it "just work" if I plug an
 >analog phone onto the correct pair coming out of the connector in the
 >back? If not, what is the output of the E&M card? (and, more
 >importantly, what would I need to do to hook it up to an analog phone?)
 >
 >Thanks for clearing things up.
 >
 >---
 >Ilia Mirkin
 >[EMAIL PROTECTED]
 >
 >On Sun, 2004-09-05 at 04:31, Steven Critchfield wrote:
 >> On Sun, 2004-09-05 at 03:10, Ilia Mirkin wrote:
 >> > hi,
 >> >
 >> > i have some newbie questions about channel banks. i have an adtran
 >> > act-1241 sitting around. it accepts D4 modules, and it contains a 
number
 >> > of e&m cards.
 >> >
 >> > first of all, how does this thing work? a t1 contains 24 channels, 
and i
 >> > noticed that the channel bank has space for 24 cards. what do these
 >> > cards do? what are their outputs? the ones that are in there have 
some
 >> > outputs on the front marked "test", but nothing else. there are a 
number
 >> > of wires coming out the back (48, if i had to guess), and it has a 
few
 >> > ports on the front which seem to be able to take in a T1. am i 
correct
 >> > in understanding that it is the card in the bank that determines the
 >> > signalling style, and not the t1? as such, is there no way that i 
could
 >> > use it in its current configuration to have it talk with analog 
phones
 >> > (i.e. something like t100p -> act-1241 with e&m cards -> phone)? 
i'm a
 >> > bit unclear on the different signalling types, and their
 >> > intercompatibilities.
 >> >
 >> > if anyone could shed any light into this, i would very much 
appreciate
 >> > it.
 >>
 >> Think of the T1 as 24 digital digital pathways. The coding of each
 >> pathway must be compatible on each end. With E&M cards, you signal with
 >> E&M and the line will work. The cards plug into a backplane where the
 >> controller routes the digital signal to the card and then optionally
 >> hook up the output from the card to a connector that consolidates many
 >> lines. Look for something that looks like an older 50 pin scsi D
 >> connector.
 >>
 >> If there is 2 RJ45 jacks on the front, and 2 5

Re: [Asterisk-Users] offtopic - channel banks

2004-09-06 Thread Lyle Giese

An analog phone need an FXS channel unit to drive them.  An analog phone
line needs to connect to an FXO channel unit.

An analog phone goes off hook by putting a short across the tip & ring.  The
FXS or a pots line from the Telco puts battery on the Tip and
Ground(actually it's a signal return and not a hard ground) on the Ring.
The short is the signal that the phone wants to make a call or answer an
incoming call.  The telco or FXS channel unit puts out ringing current to
ring the bell on an analog phone to tell you that there is an incoming call.

FXO = Foreign eXchange Office (faces the dial tone portion of an analog
phone line)
FXS = Foreign eXchange Station (faces the phone or station of an analog
phone line)

E&M signalling using a different set of wires to transmitt the on & off
hooks from one end to the other and is not directly compatible with analog
phones. E&M signalling is used for trunking, which is one way phone calls
get between switches(which Asterisk really is and also the telco office).

Those centronic connectors on the channel bank are where the analog leads
from the channel units appear.

Lyle

- Original Message - 
From: "Ilia Mirkin" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Monday, September 06, 2004 12:56 AM
Subject: Re: [Asterisk-Users] offtopic - channel banks


> While I understand everything that you have said, I'm still a little
> confused. Yes - I have what looks like a centronics connector on the
> back. So, I can do "t100p with e&m signalling" <-> "act-1241 e&m card"
> <-> what? Namely, if the E&M card deals with the T1 end of the channel,
> how do I get that to a real phone? Will it "just work" if I plug an
> analog phone onto the correct pair coming out of the connector in the
> back? If not, what is the output of the E&M card? (and, more
> importantly, what would I need to do to hook it up to an analog phone?)
>
> Thanks for clearing things up.
>
> ---
> Ilia Mirkin
> [EMAIL PROTECTED]
>
> On Sun, 2004-09-05 at 04:31, Steven Critchfield wrote:
> > On Sun, 2004-09-05 at 03:10, Ilia Mirkin wrote:
> > > hi,
> > >
> > > i have some newbie questions about channel banks. i have an adtran
> > > act-1241 sitting around. it accepts D4 modules, and it contains a
number
> > > of e&m cards.
> > >
> > > first of all, how does this thing work? a t1 contains 24 channels, and
i
> > > noticed that the channel bank has space for 24 cards. what do these
> > > cards do? what are their outputs? the ones that are in there have some
> > > outputs on the front marked "test", but nothing else. there are a
number
> > > of wires coming out the back (48, if i had to guess), and it has a few
> > > ports on the front which seem to be able to take in a T1. am i correct
> > > in understanding that it is the card in the bank that determines the
> > > signalling style, and not the t1? as such, is there no way that i
could
> > > use it in its current configuration to have it talk with analog phones
> > > (i.e. something like t100p -> act-1241 with e&m cards -> phone)? i'm a
> > > bit unclear on the different signalling types, and their
> > > intercompatibilities.
> > >
> > > if anyone could shed any light into this, i would very much appreciate
> > > it.
> >
> > Think of the T1 as 24 digital digital pathways. The coding of each
> > pathway must be compatible on each end. With E&M cards, you signal with
> > E&M and the line will work. The cards plug into a backplane where the
> > controller routes the digital signal to the card and then optionally
> > hook up the output from the card to a connector that consolidates many
> > lines. Look for something that looks like an older 50 pin scsi D
> > connector.
> >
> > If there is 2 RJ45 jacks on the front, and 2 50 pin D connectors on the
> > back, then it is likely that each card controlls 2 lines each. If there
> > is only 1 50 pin connector, then there is only 24 channels.
> >
> > Hope that helps.
>
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Re: [Asterisk-Users] offtopic - channel banks

2004-09-06 Thread Chris A. Icide

Ilia,
Think of a channel bank as a concentrator.  In a single T1 channel bank, 
you concentrate 24 analog two wire phone connections into a single 4 wire 
digital interface.

So in your case, the RJ45 connector is for the T1 interface to Asterisk, 
the local CLEC, or whatever you intend to connect it to.  On the T1 side, 
there has to be several layers of signalling and encoding.  Alot of this 
information is superfluous, but may help you when it comes to understanding 
your configs.

Before we even talk about E&M signalling, you have the T1 framing and 
encoding.  This is used to allow both ends of the T1 circuit to understand 
how the 24 channels are being configured on the 4 wire circuit.  It's 
generally going to be either sf, d4 or esf, b8zs.  This is known as the 
framing and encoding.

Once those are agreed upon, then we need to set up the way the T1 is going 
to signal across the channels.  Normal phone lines (analog) use voltages, 
resistances, and dtmf to signal what it is doing.  Since a T1 is a digital 
circuit we can't do that, so we need to set up another way to signal, so 
that the channel bank knows what to do when we send some kind of digital 
signal.  In this case, this is the E&M signalling you asked about.

Finally, you probably are looking for some way to plug your phone's RJ11 
connecter into the channel bank.  Unfortunately it's not that easy.  That 
big Centronics style connecter is where you actually have to plug up the 
phone.  there are 24 pairs of contacts in that connecter that are 
associated with each channel on the T1 circuit.  Historically, you would 
connect up a cable of 50 conductors connected to the centronics connector 
on one end, and then to one side a punch down block on the other (just a 
quick-connect style access device for copper wire).  On the other side of 
the punch down block, you would connect the wires that would then run to 
the remote wall jacks, etc. where your phones plug in.

The problem you have is wither by google mastery or just plain brute force 
testing, you need to figure out the pinout of that centronics port before 
you can connect up any phones successfully.

-Chris
On 10:56 PM 9/5/2004, Ilia Mirkin wrote:
>While I understand everything that you have said, I'm still a little
>confused. Yes - I have what looks like a centronics connector on the
>back. So, I can do "t100p with e&m signalling" <-> "act-1241 e&m card"
><-> what? Namely, if the E&M card deals with the T1 end of the channel,
>how do I get that to a real phone? Will it "just work" if I plug an
>analog phone onto the correct pair coming out of the connector in the
>back? If not, what is the output of the E&M card? (and, more
>importantly, what would I need to do to hook it up to an analog phone?)
>
>Thanks for clearing things up.
>
>---
>Ilia Mirkin
>[EMAIL PROTECTED]
>
>On Sun, 2004-09-05 at 04:31, Steven Critchfield wrote:
>> On Sun, 2004-09-05 at 03:10, Ilia Mirkin wrote:
>> > hi,
>> >
>> > i have some newbie questions about channel banks. i have an adtran
>> > act-1241 sitting around. it accepts D4 modules, and it contains a number
>> > of e&m cards.
>> >
>> > first of all, how does this thing work? a t1 contains 24 channels, and i
>> > noticed that the channel bank has space for 24 cards. what do these
>> > cards do? what are their outputs? the ones that are in there have some
>> > outputs on the front marked "test", but nothing else. there are a number
>> > of wires coming out the back (48, if i had to guess), and it has a few
>> > ports on the front which seem to be able to take in a T1. am i correct
>> > in understanding that it is the card in the bank that determines the
>> > signalling style, and not the t1? as such, is there no way that i could
>> > use it in its current configuration to have it talk with analog phones
>> > (i.e. something like t100p -> act-1241 with e&m cards -> phone)? i'm a
>> > bit unclear on the different signalling types, and their
>> > intercompatibilities.
>> >
>> > if anyone could shed any light into this, i would very much appreciate
>> > it.
>>
>> Think of the T1 as 24 digital digital pathways. The coding of each
>> pathway must be compatible on each end. With E&M cards, you signal with
>> E&M and the line will work. The cards plug into a backplane where the
>> controller routes the digital signal to the card and then optionally
>> hook up the output from the card to a connector that consolidates many
>> lines. Look for something that looks like an older 50 pin scsi D
>> connector.
>>
>> If there is 2 RJ45 jacks on the front, and 2 50 pin D connectors on the
>> back, then it is likely that each card controlls 2 lines each. If there
>> is only 1 50 pin connector, then there is only 24 channels.
>>
>> Hope that helps.
>
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Re: [Asterisk-Users] Four single-port FXO Cards in one * box

2004-09-06 Thread Lyle Giese

1) they will work.  Can depend on the phone system in your area and how far
you are from the telco office(in cable feet).

2) 4 would be hard to make work.  Each card needs a different and unique IRQ
and hates to share IRQ's with anything.

Lyle

- Original Message - 
From: "Andrew Newton" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, September 06, 2004 3:44 AM
Subject: [Asterisk-Users] Four single-port FXO Cards in one * box


> Hi,
>
> I have found some cheap single port FXO Cards on ebay. They are
> apparently X100P compatible like the cards sold by Digium
>
> My question is 1) Are they any good 2) will 4 such cards work in a
> single * box without too much trouble?
>
> Thanks
> Andy
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[Asterisk-Users] spouse-friendly spa-3000 pstn interface


This post is simply documenting a spouse-friendly way of using the
spa-3000 as both a fxs and fxo port for basic soho environments in
the US, allowing asterisk to participate as needed/wanted.

All home phones are connected _only_ to the spa-3000 fxs port.

The incoming home pstn line is connected _only_ to the spa-3000 
fxo port.

Defined Line 1 (fxs) to register with asterisk via sip (extn ), 
with silence suppression disabled.

Defined the PSTN Line (fxo) to register with asterisk via sip using 
a second sip.conf entry (extn ).

PSTN User, defined PSTN Ring Thru Line 1 Ring Settings as "1".

In Line 1, defined Gateway Account #1 to point to asterisk, and
created a dialplan entry like:
 (*xx|81xxx.<:@gw1>|3xxx<:@gw1>|0<:@gw0>|[2-9]11<:@gw0>|xx.<:@gw0>)
Note: gw0 defaults to the pstn line per the spa-3000 doc.

Result:
1. If asterisk is down for any reason, all incoming pstn home calls 
still ring through to the analog house phones.
2. Incoming pstn calls ring through to the house phones without any
asterisk involvement, and callerid is properly passed to the house
phones. (Since my * also has an fxo port attached behind the spa-3000,
* still rings and I can answer the call via it.)
3. Loss of commercial AC power causes the spa-3000 to physically
cut through the pstn line to the house phones, leaving all house
phones operative. Same with loss of an ethernet physical connection.
4. Calls originating from within asterisk to the house phones use:
 exten => ,1,SetVar(ALERT_INFO=bellcore-r3)
 exten => ,2,Dial(SIP/,25,r)
 exten => ,3,Hangup
and ring the house phone with distinctive rings. Spouce recognizes
incoming pstn calls as different from voip/asterisk calls. (We have
about six analog phones behind the spa-3000 that properly ring on 
all calls.)
5. Calls placed from the house phones (via fxs Line 1) use the above
dialplan.
 a. LD calls starting with 81xxx. are routed via * to an ITSP (gw1)
 b. normal US local calls routed via pstn line (gw0)
 c. calls to 3xxx are asterisk extensions and are therefor routed
to asterisk (gw1)
 d. 911 (as well as 211, 311, etc) are routed via the pstn line.
 e. 0 calls are routed via pstn line
If spouse is having a problem with dialing LD via 81xxx, then dialing
the same call without the 8 routes the LD call via pstn line.
6. Calls originating from within asterisk can be routed to this pstn
line via Dial(Sip//...) if needed.
7. The old answering machine (that my spouse knows how to use) still
accepts all unanswered calls.

I fully understand the above approach keeps asterisk out of the loop
most of the time, and that was an objective (from a spouse perspective
for now). It represents the least intrusive integration without
remedial training, etc. :)  No need to purchase six sip phones either.

There has been no echo or any other negative issues thus far; haven't
tested a lot of things as yet though. (Eg, what happens when the house
phone is talking via voip conversation and an incoming pstn call
arrives? etc.) The spa-3000 is running v2.0.1(GWd).

Rich


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[Asterisk-Users] x-lite and pound key

2004-09-06 Thread Randy Bush

[ wiki on xten/x-lite gets you to a 5mb pdf which tells you how to
  do a windows install.  deep :-(  ]

anyone know how to make x-lite be # key transparent, i.e. send the
key when it is poked?

randy

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Re: [Asterisk-Users] Newby question. Basic structure

> From: Rodolfo Grave <[EMAIL PROTECTED]>
>
> I've being reading posts from the list since yesterday and I feel this 
> question was answered a lot time ago, but the list archives are a mess 
> (yet). I hope some one is willing to help me out.
> 
> I want to set up this:
> 
> caller - PSTN  (SOMETHING1) -- VoIP - (SOMETHING2) 
>  PSTN
> 
> I think this must be a very basic architecture, but I'm not sure wat 
> SOMETHING1 and SOMETHING2 are. I've been on this for a while now (around 
> two months) and till yesterday I haven't find Asterisk.
> 
> Can you help me? I need to know hardware and software needs for this. I 
> have read a few about voIP and have some programming and configuration 
> skills under Linux and Windows.

In terms of using asterisk to implement your diagram, "something1" and
"something2" are basic linux boxes equipped with:
 - pstn interface card (such as x100p, tdm, isdn, T1 card)
 - ethernet interface (to connect to your broadband internet)
 - asterisk software

The type of pstn interface card to use will be dependent upon how
many "simultanous" phone conversations you'd like to support. The
x100p card is a single pstn line interface; the tdm card supports
one to four (tdm04b bundle) pstn lines (the tdm card needs to be
purchased with fxo interface modules); the isdn card supports from
one to 23 (?) pstn logical interfaces (depending upon the exact
card purchased); or the T1/E1 card supports from 1 to 24 (T1) or
1 to 32 (E1-?) conversations. You'll find most of those cards at
www.digium.com under Hardware Products.

If you are outside the US, consider the TDM card as opposed to
the x100p card.

The size of Linux system will also be dependent on how many 
simultanous calls you want to support, and exactly how you have
the system configured. For a single call, some folks have it
running on old 300 mhz (and slower) box, while high traffic
volumes will require a much faster system.

The asterisk software can be found on the digium home page 
(lower-left menu option), which points to www.asterisk.org

Also, lots of good reference material at www.voip-info.org

After you've configured your system with your favorite Linux
distro, download and compile asterisk per the instructions found
on the www.asterisk.org site. Pay attention to the last steps
on your screen that copies configuration samples to /etc/asterisk.
Read through those configuration files, read the info at
www.voip-info.org, and you should be able to get a system
running.


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RE: [Asterisk-Users] UK Callerid bug #1719 & TDM400p

2004-09-06 Thread Edward Eastman

Brilliant - thanks, took me half an hour but it's working now.

Just for the record, settings as follows:

The patch on http://bugs.digium.com/bug_view_page.php?bug_id=009
(ast-UK-and-DTMF-pol-CID.diff) doesn't seem to work for current cvs, but I
backed up to cvs as of 31/08/04 and that worked fine.

Zapata.conf:

usecallerid=yes
cidsignalling=v23
cidstart=polarity

usecallerid=uk doesn't work, has this changed somewhere along the way, or is
this something else?

Caller ID detects fine, although I get this logged to asterisk console:

Sep  6 13:56:22 WARNING[393238]: chan_zap.c:3369 zt_handle_event: Didn't
finish Caller-ID spill.  Cancelling.

I'll try and add this to the wiki when I get time

Thanks

Ed

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje
Sent: 06 September 2004 13:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] UK Callerid bug #1719 & TDM400p

Edward Eastman wrote:
>> Hi
>>
>>
>>
>> Is this patch
>> (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the
>> best/only way to get callerid working in the UK with a tdm400p?  I
>> thought I'd seen a patch that'd gone into cvs, but maybe I was just
>> imagining things ;)
>>
>>
>>

Check the bug tracker for id=9, there has been some development here. UK BT
CLIP and DTMF CLIP for the TDM400 (will not work for the X100P) is now
merged into one patch.

/Soren

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Re: [Asterisk-Users] SIP authentication problem

2004-09-06 Thread Olle E. Johansson

Kurt Bauer wrote:
Hi,
I have the following setup:
   E100P
 SER <> * <-> PBX
This works just fine, except when there are users on both boxes (ie. SER 
and asterisk), whose usernames are the same, although the realm is 
different.
At this point, Asterisk doesn't care about the realms. It's on my
to-do-list. If you look at chan_sip2, it's been in the comments
on "to-do" for a long time.
An example:
user '[EMAIL PROTECTED]' wants to call some extension in the PBX, but 
as user '[EMAIL PROTECTED]' exists too, * tries to authenticate 
the user, which it shouldn't do, at least I guess so.

Shouldn't asterisk differentiate between the realms ie. [EMAIL PROTECTED] != 
[EMAIL PROTECTED] ?
Yes, it should. BTW this is not realms, realm is in the auth header. This
is the domain part - and yes, asterisk should be aware of which domains
it is responsible for.
It's a bug in the architecture. A first step was to make Asterisk realm-
aware, so not all asterisk's in the world had the same realm.
/Olle
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[Asterisk-Users] Placing Asterisk between existing PBX and PSTN

2004-09-06 Thread Begumisa Gerald M

Hi,

I've read through the Asterisk handbook and I'd just like clarification
from someone that's implemented the above before.  Lets imagine I want to
use the CallingCard application and don't want to tell a client to buy a
channelbank (the analog extensions are too many to connect to FXS cards),
I figure I could set them up as below:


Original Existing Setup
---

 PSTN  +---+
--||   ||--A1
--|| PBX   ||--A1
--||   ||--A1
--||   ||--A1
   +---+

A1,A2,A3,A4 are analog extensions


Setup With Asterisk
---

 PSTN   +--+  +---+
--|||  ||||   ||--A1
--|FXO Card|| Asterisk ||FXS Card||  PBX  ||--A2
--|||  ||||   ||--A3
--|||  ||||   ||--A4
+--+  +---+

So they only pay for the Asterisk box, the FXO Card (TDM04B) and FXS Card
(TDM40B).

I'd appreciate any "yes/no/been there" answers.  I just want to make sure
about this, in case there's anyone that's done this before.

Thanks in advance.


Gerald.
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Re: [Asterisk-Users] Four single-port FXO Cards in one * box

2004-09-06 Thread Marconi Rivello

Hi, I have one Ambient MD3200 based modem, and it works. There are
problems, but I saw reporting of such problems also with Digium's
official hardware. I know that everyone should buy official Digium
hardware, because of their marvelous job with Asterisk, but US$ 100
was just too expensive for me, as I have only personal use for it.

On Mon, 06 Sep 2004 09:44:13 +0100, Andrew Newton
<[EMAIL PROTECTED]> wrote:
> Hi,
> 
> I have found some cheap single port FXO Cards on ebay. They are
> apparently X100P compatible like the cards sold by Digium
> 
> My question is 1) Are they any good 2) will 4 such cards work in a
> single * box without too much trouble?
> 
> Thanks
> Andy
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Re: [Asterisk-Users] UK Callerid bug #1719 & TDM400p

2004-09-06 Thread Soren Rathje

Edward Eastman wrote:
>> Hi
>>
>>
>>
>> Is this patch
>> (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the
>> best/only way to get callerid working in the UK with a tdm400p?  I
>> thought I'd seen a patch that'd gone into cvs, but maybe I was just
>> imagining things ;)
>>
>>
>>

Check the bug tracker for id=9, there has been some development here. UK BT
CLIP and DTMF CLIP for the TDM400 (will not work for the X100P) is now
merged into one patch.

/Soren

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RE: [Asterisk-Users] SIP Swissvoice de-register

2004-09-06 Thread Sergio Serrano

SIP version

IP10 SP v0.0.1 (Build 5)


Regards,

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Florian
Overkamp
Enviado el: lunes, 06 de septiembre de 2004 13:42
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: RE: [Asterisk-Users] SIP Swissvoice de-register


Hi,

> -Original Message-
>   I'm trying to configure a swissvoice IP10S but after a
> minutes this phones appears like UKNOWN in sip show peers and 
> it is unaccesible.
> This phone can make call but it can't receive calls.

What firmware are you running with ? Bog-standard IP10's come with H323
or MGCP. SIP is still in early stages (but coming soon).

Florian

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[Asterisk-Users] UK Callerid bug #1719 & TDM400p

2004-09-06 Thread Edward Eastman









Hi

 

Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719)
the best/only way to get callerid working in the UK with a tdm400p?  I thought I’d
seen a patch that’d gone into cvs, but maybe I was just imagining things
;)

 

Should this patch work against current cvs?  Of the 3 files
2 are .patch and one is .diff – what’s the difference between them,
and how should I apply the diff (at the moment I’m doing “patch –p1
< patchname.patch” for the others which seems to work, but I’m
doing this slightly blind and I’m not quite sure if this is correct.

 

Thanks

 

Ed

 

 






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RE: [Asterisk-Users] SIP Swissvoice de-register

2004-09-06 Thread Florian Overkamp

Hi,

> -Original Message-
>   I'm trying to configure a swissvoice IP10S but after a 
> minutes this phones appears like UKNOWN in sip show peers and 
> it is unaccesible.
> This phone can make call but it can't receive calls.

What firmware are you running with ? Bog-standard IP10's come with H323 or
MGCP. SIP is still in early stages (but coming soon).

Florian

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[Asterisk-Users] SIP Swissvoice de-register

2004-09-06 Thread Sergio Serrano

Hi all,
I'm trying to configure a swissvoice IP10S but after a minutes
this phones appears like UKNOWN in sip show peers and it is unaccesible.
This phone can make call but it can't receive calls.

Any idea?

Regards,
srsergio

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[Asterisk-Users] Newby question. Basic structure