Re: [asterisk-users] Cisco 7960 asterisk 1.8.22 ringlist.dat error

2011-02-15 Thread Tom Rymes

On 02/14/2011 12:04 PM, James Miller wrote:

I did the command listed, and its actually requesting RINGLIST.DAT, so I
changed the filename to match its request but now its showing in the
ring type setting:

Chirp 1

Chirp 2

24 24-ring-tone-1.raw

Att1 ring_att1.pcm


snip

Do you actually have those files in your TFTP directory? You need both 
the RINGLIST.DAT file that specifies what files are available and what 
they are called, PLUS the actual ring files themselves. All of my Cisco 
ringer files are .pcm files, like ATT,pcm, ATT2.pcm, etc.


Tom

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Re: [asterisk-users] Callback through extensions.conf?

2011-02-07 Thread Tom Rymes

On 02/07/2011 11:46 AM, Gilles wrote:

snip


Asterisk runs as root, and owns this file as well.


Have you tried setting the permissions of this file to world readable, 
to ensure that any user can read it and eliminate potential permissions 
problems?


Worth a shot. While you're at it, output from the ps command that shows 
the output, command line, and header for asterisk will help, too.


Tom

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Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer

2011-02-03 Thread Tom Rymes

On Feb 3, 2011, at 11:12 AM, Kevin P. Fleming wrote:

 The Queue() application can automatically pause members who fail to answer; 
 this would be the solution to your problem. With that solution in place, 
 though, the agent will still need to be able to un-pause when they return to 
 their desk, and since that is the case, they really should be taught to go on 
 pause when they leave their desk as well :-)

Not to mention that your caller has to wait for however long your agent timeout 
is when this happens the first time, which is bad customer service. 

I am a little confused as to what the OP wants the system to do? Call the 
proper agent, but when they don't answer, on the next call, it shouldn't call 
the same agent? OK, but for how long? 5 minutes? Until they manually unpause 
(current option as described by Kevin), 30 minutes? Should it then up their 
penalty? For how long? 

Maybe some more specifics would help here.

Tom
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Re: [asterisk-users] save the calls with asterisk

2011-01-31 Thread Tom Rymes

On 01/31/2011 12:51 PM, salaheddine elharit wrote:


I have asterisk installed in our call center and i want to know how to
do in order to save all the calls (inbound and outbound) if there is any
tool


Yes, there is.

Tom

PS: Sorry, I couldn't resist!

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Re: [asterisk-users] save the calls with asterisk

2011-01-31 Thread Tom Rymes

On 01/31/2011 12:51 PM, salaheddine elharit wrote:


I have asterisk installed in our call center and i want to know how to
do in order to save all the calls (inbound and outbound) if there is any
tool


OK, now to be somewhat more helpful, this is a common scenario. You 
should search for information on Call Monitoring and recording. 
Specifically, call queues have monitoring options that will likely fit 
your needs.


Are you running a GUI like FreePBX?

Tom

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Re: [asterisk-users] faxter

2011-01-30 Thread Tom Rymes
On Jan 30, 2011, at 4:21 AM, Pezhman Lali wrote:

 Dear,
  Faxter is an opensource email to fax gateway, 
 please check it, let me know if any bug.
 
 best

I'll get right on that.

Tom

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Re: [asterisk-users] Help determining SpanDSP version

2011-01-26 Thread Tom Rymes

On 01/25/2011 3:38 PM, Danny Nicholas wrote:

[snip]


Is there a good way to determine what version of SpanDSP I have
installed and whether the app_fax.so module is the same version?


[snip]


Try these two commands:
- whereis spandsp.so
- find /|grep spandsp.so


Those commands do point towards related pieces, and I think that 
/usr/include/spandsp/version.h might hold some clues, it doesn't shed 
any light on the app_fax.so module.


Please pardon my ignorance in this area, I'm sure it's straightforward. 
As for compiling, I have started with a packaged version, and will move 
to rolling my own as things move along.


Many thanks,

Tom

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Re: [asterisk-users] Recommended Windows client to display CID?

2011-01-26 Thread Tom Rymes

On 01/26/2011 9:04 AM, jon pounder wrote:

On 01/26/2011 08:52 AM, Gilles wrote:

If you like open source what are you doing running windows ?

Getting anything to work properly there which does network
communications is a huge PITA since every user has their own firewall
and different settings etc etc etc.


Unless, of course, you properly implement Group Policies (which is 
Windows Server only, IIRC, but still...)


On 01/26/2011 9:14 AM, Joel Maslak wrote:
 I have asterisk call out to a shell script which sends a jabber
 message to the user (along with links to any open tickets in our
 ticketing system associated with that CID).  All free, but requires
 work to build.

Ooh. I like this. Can you post a sample, or maybe a synopsis of what 
pieces you are using to tie this all together?


To answer the OP's question about XMPP clients, Spark from Ignite 
Realtime and Pandion are both good in my experience.


Tom

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Re: [asterisk-users] res_fax

2011-01-26 Thread Tom Rymes

On 01/26/2011 1:49 PM, Kevin P. Fleming wrote:

snip


Steve did not write res_fax (which where SendFAX and ReceiveFAX come
from)


snip

I am personally a little confused here, because I have a ReceiveFAX 
application when I unload the res_fax module and res_fax_digium module 
and load the app_fax module. In other words, I think that multiple 
modules provide applications named ReceiveFax and SendFAX.


Am I correct to infer that using app_fax.so is no longer recommended and 
that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now 
the way to go?


Tom

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Re: [asterisk-users] res_fax

2011-01-26 Thread Tom Rymes

On 01/26/2011 2:16 PM, Kevin P. Fleming wrote:

On 01/26/2011 01:12 PM, Tom Rymes wrote:

On 01/26/2011 1:49 PM, Kevin P. Fleming wrote:


snip


Am I correct to infer that using app_fax.so is no longer recommended and
that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now
the way to go?


That is correct. app_fax is deprecated (and that is why it is marked as
don't build by default), and res_fax plus a technology module
(res_fax_spandsp or res_fax_digium) is the replacement for it. All of
the work that the Digium team has done improving T.38 negotiation and
interoperability has gone into res_fax, not app_fax. Users of Asterisk
1.8.x should only choose to build app_fax if they have a specific need
for it (and if that's the case we'd like to know what the need is so we
can ensure that res_fax can satisfy it). Users of older Asterisk
releases will have app_fax by default (since res_fax was not included in
those versions), but if they want to use Digium's res_fax_digium module
they can download it along with res_fax and use them instead.


Gotcha. So, 1.6 users who install FFA get res_fax and res_fax_digium. 
Presumably, 1.6 users could also combine res_fax and res_fax_spandsp?


Steve - Will compiling the latest version of SpanDSP on a 1.6 system 
result in a res_fax_spandsp.so module?


Tom


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[asterisk-users] Help determining SpanDSP version

2011-01-25 Thread Tom Rymes
OK, I generally use Hylafax+IAXModem for our faxing, but I have been 
fiddling with FFA and SpanDSP for a while.


Is there a good way to determine what version of SpanDSP I have 
installed and whether the app_fax.so module is the same version?


Many thanks,

Tom

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Re: [asterisk-users] Inbound routes

2011-01-21 Thread Tom Rymes

On 01/21/2011 8:49 AM, Vitor Carlos Flausino wrote:


The system has 1 DAHDi card with 2 analog FXO ports (to pstn)


[snip]


However, it seams that when the call is received, the trunk does

 not inform the DID

This is because FXO ports do not support DID. You need to route the call 
based on the port it came in on, not based on a DID.


Tom

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Re: [asterisk-users] res_fax

2011-01-21 Thread Tom Rymes

On 01/21/2011 8:59 AM, Steve Underwood wrote:

On 01/21/2011 08:37 PM, Tom Rymes wrote:

On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote:


[snip]


Its easy to set up some t38modem channels and some iaxmodem channels for
receiving FAXes. Transmit is more problematic. With this split config,
you need to know in advance whether the particular number is accessible
by T.38 or by audio. Most people won't.

Steve


Good point.

Perhaps you could route via chan_clairvoyant?

Tom

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Re: [asterisk-users] Hi, agent intro-speech for outside caller

2011-01-20 Thread Tom Rymes
On Jan 19, 2011, at 11:08 PM, DSR wrote:

 Is there anyway to play prerecorded agent intro-speech (like Hello, my name 
 is ) to outside caller when agent picks up?

I don't know of a way to do that, but I can say that, as a caller, it is highly 
annoying. Your agents ought to be able to do that themselves, no?
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Re: [asterisk-users] context problem

2011-01-20 Thread Tom Rymes

On 01/20/2011 10:58 AM, Jonas Kellens wrote:

[snip]


I have the following registrations :

register = 119909:pas...@sip.prov.org/52525252
mailto:119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
mailto:119909:pas...@sip.prov.org/59595959


[snip]


Problem :

the call always enters : exten = _52525252

and never : exten = _59595959

Why is that ??


I may be wrong here, but I think you can only register once. The last 
registration received will overwrite the first one. You will need to 
specify a second entry and register that one separately. This is the 
same reason you cannot register two devices to the same extension.


Have you checked the logs and verified that the SIP provider actually 
sends 59595959 when you dial that number? Or do you get sent 52525252 no 
matter what?


Someone please correct me if I am wrong here.

Tom

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Re: [asterisk-users] Internode weirdness

2011-01-20 Thread Tom Rymes

On 01/19/2011 10:34 PM, Da Rock wrote:


WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
non-existing call leg on other UA. SIP dialog
'481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up.


Have you tried disallowing re-invites?

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Re: [asterisk-users] Asterisk to asterisk t.38

2011-01-20 Thread Tom Rymes

On 01/20/2011 4:26 PM, Amit Nepal wrote:


I have an Audio code gateway between two asterisk servers. The audio
code has PRI connected for PSTN. I can send faxes and receive faxes in
ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and
receive faxes. The only problem I am having is sending/receiving between
ast 1.4 and ast 1.6.

ATA (T.38 capable)  AST 1.6 AUDIO CODEAST
1.4ATA (t.38 Capable)


It sounds like you are trying to send a fax directly from AST 1.6 to 
AST 1.4 via t.38 that never hits the PSTN. Have you tried sending a 
fax from AST 1.6 out via the PSTN, and then back in via the PSTN to 
AST 1.4?


Tom

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Re: [asterisk-users] Asterisk to asterisk t.38

2011-01-20 Thread Tom Rymes
On Jan 20, 2011, at 5:52 PM, Amit Nepal wrote:
 On 1/20/2011 3:07 PM, Tom Rymes wrote:
 On 01/20/2011 4:26 PM, Amit Nepal wrote:
 
 I have an Audio code gateway between two asterisk servers. The audio
 code has PRI connected for PSTN. I can send faxes and receive faxes in
 ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and
 receive faxes. The only problem I am having is sending/receiving between
 ast 1.4 and ast 1.6.
 
 ATA (T.38 capable)  AST 1.6 AUDIO CODEAST
 1.4ATA (t.38 Capable)
 
 It sounds like you are trying to send a fax directly from AST 1.6 to AST 
 1.4 via t.38 that never hits the PSTN. Have you tried sending a fax from 
 AST 1.6 out via the PSTN, and then back in via the PSTN to AST 1.4?

 

 Yes Tom, I am sending via the PSTN  gateway which is audio code in my case.

Ok, are both boxes connected to the same audiocodes gateway? Or are they in 
different locations?

Can you try sending via regular POTS lines?

Tom
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Re: [asterisk-users] res_fax

2011-01-19 Thread Tom Rymes
On Jan 19, 2011, at 3:18 PM, Jason Parker wrote:

 On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
 I am working on some fax tools for some of my users. I am reading the
 https://wiki.asterisk.org docs for faxing.
 Is see Application_SendFax and Application_SendeFax has one been 
 discondinued?
 Any feed back on using the res_fax module would be apperciated. Any examples 
 or
 other.
 
 There was a typo in the res_fax documentation.  Application_SendeFax should 
 be the correct documentation.  I don't know where Application_SendFax is 
 coming from - it's probably old.  When the next import happens, 
 Application_SendFax should be replaced by the correct version (then I'll try 
 to remember to remove the bogus SendeFax copy).

Am I the only one confused here? (probably) It seems like you imply that 
SendeFax (which looks like a typo to me) is correct in the second sentence, 
then reverse yourself in the last parenthetical statement.



Tom
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Re: [asterisk-users] Top Posting

2011-01-19 Thread Tom Rymes

On Jan 19, 2011, at 10:06 AM, C F wrote:

 On Sun, Jan 16, 2011 at 9:47 PM, James Miller paramedi...@gmail.com wrote:
 When you get over 500 emails a day on your blackberry you have make a 
 decision on what is or is not worth reading at that moment.
 
 Its not lazy at all its cutting through the fluff and finding the emails 
 worth while.  When inside outlook you don't have the hot key b to scroll to 
 the bottom so again, I'd have to scroll down. Add up the time it takes per 
 email x 500 emails, you loose considerable amount of productivity.
 
 Oh C'mon this is definitely lazy never heard of CTRL+END it works in Outlook

How amusing that you follow that statement by being too lazy to trim all of the 
irrelevant crud after your comment by pressing ctrl-shift-end followed by 
delete. It works in Outlook.

Tom
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Re: [asterisk-users] Calling rules

2011-01-19 Thread Tom Rymes
On Jan 19, 2011, at 5:06 AM, Vitor Carlos Flausino wrote:

 In other words, which of the following is your situation:
 
 1.) User dials 0X, asterisk sends 0X to the telco.
 2.) User dials 0X, asterisk parses 0, strips it, and sends X
 to the telco.
 
 That might narrow it down.
 
 Option 2. 0 is to get an external line and XXX is passed to telco.
 
 -vcf

It seems to me that you are passing the 0 to the telco when the user dials 
all digits at once. When they dial the 0 first, the call gets sent to one 
extension (probably extension 0 or _0) and just connects them to the 
outside line, sending nothing to the telco. When they dial 0X, asterisk 
matches another extension (probably _0. or another that begins with _0), 
one that connects them to the outside line and sends everything out to the 
telco, including the 0. 

Just a guess, but it sounds right to me. If so, you need to modify the dial 
command to strip the 0 before sending it.

Tom
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Re: [asterisk-users] Top Posting

2011-01-18 Thread Tom Rymes

On 01/18/2011 10:18 AM, Andrew Thomas wrote:

Why do I top post?  Simple.  I read every message in the thread - and if
there are 10 messages (for example) in that thread - then why should I
have to read them all over again on the last one?


OK, this is a stupid thread, nobody is going to be convinced by anything 
I say, and by replying it, I am just feeding the trolls and prolonging 
everyone's agony.


But I can't resist.  This is the fifth or sixth post that makes this 
improper assumption.


Nobody, I mean NOT A SINGLE PERSON, is advocating bottom posting without 
trimming. The argument is between:


1.) Top Posting - No Trimming
2.) Bottom or interleaved posting WITH TRIMMING.

In fact, I'd rather you top post and trim than bottom post and not. 
That's one thing we can all agree on.


Tom
[going back to biting my lip]

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Re: [asterisk-users] res_fax_digium.so crashing

2011-01-18 Thread Tom Rymes

On 01/18/2011 3:21 PM, Steve Totaro wrote:


If you are swapping out systems in really busy offices that rely on
faxing to keep the doors open, do a whole bunch of testing.


I have no experience with Digium's FFA, beyond installing it and 
receiving a fax or two. So I can't really agree or disagree with your 
assertions. However, I can say that iFax does sell a paid/supported 
version of Hylafax that includes t.38 modems.


I've never used their version, but I have been a HylaFAX user for years 
(both with real modems and POTS lines and with Asterisk PRI and 
IAXModem), and it is an excellent solution. I also have some experience 
with their HylaFAX client, which is included with the server, and I can 
say it is very well done.


Might be worth the cash for large fax users like you describe.

Tom

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Re: [asterisk-users] Calling rules

2011-01-18 Thread Tom Rymes

On 01/18/2011 3:20 PM, Vitor Carlos Flausino wrote:

   == Spawn extension (DLPN_DialPlan1, 0924343424, 1) exited non-zero on 
'SIP/6005-0002'


Vitor,

Can you please clarify whether the 0 should be received by Asterisk 
and processed internally, or whether it should be passed to the DAHDI 
channel by asterisk?


In other words, which of the following is your situation:

1.) User dials 0X, asterisk sends 0X to the telco.
2.) User dials 0X, asterisk parses 0, strips it, and sends X 
to the telco.


That might narrow it down.

Tom

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Re: [asterisk-users] Selecting the E1 cards for the call

2011-01-16 Thread Tom Rymes

On Jan 16, 2011, at 7:29 AM, bilal ghayyad wrote:

 Dears;
 
 I am looking for the card that does not need an electrical power, which one? 
 Is the PCI express doing this?
 
 Regards
 Bilal

Bilal,

The only telephony cards that require a power connector are those with FXS 
ports for plugging in analog telephones as extensions. Any of the E1 cards you 
are looking at will not require any additional power beyond what the 
motherboard provides to the slot.

Tom
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Re: [asterisk-users] Asterisk stops responding

2011-01-15 Thread Tom Rymes
On Jan 15, 2011, at 2:01 AM, Carlos Chavez wrote:

 The problem is that Asterisk simply stops responding.  No calls in or out
 and you cannot even get to the CLI.  The process seems to be running but there
 is simple no activity.  All I see in the log files is:
 
 [Jan 14 16:30:46] WARNING[20747] pbx.c: Failed to create new channel thread
 [Jan 14 16:30:46] WARNING[20747] chan_sip.c: Failed to start PBX :(
 [Jan 14 16:30:47] WARNING[20745] pbx.c: Failed to create new channel thread
 [Jan 14 16:30:47] WARNING[20745] chan_dahdi.c: Unable to start PBX on 
 DAHDI/29-1
 
 After restarting Asterisk everything is back to normal.  The time between
 the first failure and the second was almost a month, between the second and
 third a few days.  

Carlos,

What is in the logs immediately preceding the warning you have posted here? 
Scan up a number of lines (more if you have a very verbose installation, like 
FreePBX) and see if anything pops out at you. Basically, you want to figure out 
what was happening on the server at the time of the crash? Incoming fax? Hangup 
of a Dahdi channel? Incoming Dahdi call, etc.

That will likely point you in the right direction.

Tom
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Re: [asterisk-users] Top Posting

2011-01-15 Thread Tom Rymes
On Jan 15, 2011, at 9:29 AM, Don Kelly wrote:

 That said, of course I want to follow this list's etiquette. I've posted a
 couple times asking how I can interleave responses in Outlook or what other
 approach can I take to make it practical to stop top-posting. Any
 suggestions?

Don:

Outlook-QuoteFix: http://home.in.tum.de/~jain/software/outlook-quotefix/

I found that program last night after reading one of the pages linked in this 
thread. The program isn't supported on OL 2007 and newer, but there is a link 
on the page to a macro for newer versions. Wish I had known about it years ago!

Also, http://mailformat.dan.info/config/outlook.html shows the general steps 
needed to make Outlook approximate standards. 

HTH,

Tom
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Re: [asterisk-users] Selecing the E1 cards for the call center

2011-01-14 Thread Tom Rymes
While we're at it, can someone please tell me whether I should be using 
vi or emacs? ;-)


Many thanks,

Tom

PS: Bilal: You have asked a nearly unanswerable question. Some prefer 
one, some prefer the other. Both cards are quality items. I can say that 
I only have experience with Sangoma T1/E1 cards, but our Digium FXO/FXS 
card works well, too.


On 01/14/2011 12:42 PM, bilal ghayyad wrote:

Hi All;

We would like to build a call center having 2 E1, but we would like to know 
which card to select:

Sangoma or Digium?

And card type to be PCI express or PCI 5.0V or PCI 3.3V ?

Any advise or special recommendations for the call center?

Regards
Bilal


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Re: [asterisk-users] Why are 4 ports used for a single call?

2011-01-14 Thread Tom Rymes

On 01/14/2011 4:19 PM, Bruce B wrote:

Also, to get the SIP very well as well, SIP uses both TCP/UDP 5060
right? and why are there recommendations of opening 5000-5082 UDP for
SIP along with 5060 TCP? Are there any niceties to that as well? maybe
video transmission stuff?


More likely, it's because only one client behind NAT can use port 5060, 
so other clients need to use other ports. Could be another reason, though.


Tom

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Re: [asterisk-users] Why are 4 ports used for a single call?

2011-01-14 Thread Tom Rymes

On Jan 14, 2011, at 5:24 PM, Bruce B wrote:

 So, simply pressing Reply and typing in the first line (using gmail webmail 
 without any clients) is a sin here? How is that top posting??? probably your 
 clients reading that way?

It may be a sin here, but it is certainly impolite many places, and illogical 
everywhere. This is because we normally read top to bottom, but top-posting 
forces you to read bottom to top.

Tom
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Re: [asterisk-users] Why are 4 ports used for a single call?

2011-01-14 Thread Tom Rymes
On Jan 14, 2011, at 6:45 PM, Bruce B wrote:

 You really want to read the LONG LONG signature from some people before you 
 read the actual latest message? I don't know about thatI guess it's a 
 preference.

Suffice it to say, Bruce, this subject has been hashed over thousands, nay, 
hundreds of thousands of times, and I doubt anything new can be had from doing 
it again. 

FYI, It is also considered good etiquette to remove any non-relevant 
information from the quoted text to keep it short and easy to parse, especially 
removing the automatically generated footers from the list.

As for your question about ports (see, I can stay on topic occasionally!), 
someone already mentioned something about some equipment using 5004 for RTP, 
IIRC, and I mentioned the common use of 5061, 5062, 5063, etc for multiple SIP 
clients behind NAT. There may be other reasons, too.

Tom
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Re: [asterisk-users] Why are 4 ports used for a single call?

2011-01-14 Thread Tom Rymes
On Jan 14, 2011, at 7:12 PM, Bruce B wrote:

 Thanks. That is in both TCP and UDP for SIP right? or simply UDP would do it 
 as well? I am talking strictly in case of Asterisk.

Asterisk 1.6 and newer support SIP over TCP. Older versions were UDP only, IIRC.

Tom
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Re: [asterisk-users] Top Posting

2011-01-14 Thread Tom Rymes
On Jan 14, 2011, at 8:52 PM, Don Kelly wrote:

I have nothing to add to the nascent flame war that I thought we had so 
narrowly avoided when I sent my last message. However:

 What did you mean, Andrew, about Don's multiple
 signatures which I think he will review?
 
 --Don

[snip]

Andrew meant these multiple signatures, and implied that, once you looked at 
them, you would realize it's a little redundant and not relevant to list users:

 --Don
 
 Don Kelly
 
 PCF Corp
 People Come First
 651 842-1000
 888 Don Kell(y)
 651 842-1001 fax


It's a free country, but given that you prefer a top-posting style where you 
don't trim previous messages (not judging here, just saying), you might 
consider omitting your signature for list posts, as it adds an additional eight 
lines to each message you send, which can really add up. Will it end hunger or 
bring about world peace? No. Will it be that little bit easier on everyone's 
eyes? Yes.

I think the main lesson from Andrew's post is that top or bottom posting 
doesn't matter anywhere near as much as trimming posts, so that only that 
portion of a previous message that you need for context is included, making the 
entire message compact and nicely legible.

Tom
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Re: [asterisk-users] queue_log in MySQL database

2011-01-13 Thread Tom Rymes

On 01/13/2011 11:25 AM, Danny Nicholas wrote:



*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, January 13, 2011 10:19 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] queue_log in MySQL database

Hello,

can /var/log/messages/queue_log be saved in a MySQL database ??

So it would be easier to work with...

Kind regards,
Jonas.

I’d say that depends on your release. Check this link

http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL


Specifically, you're looking for the part I added that mentions the 
changes to how extconfig.conf entries are referenced. You need to use 
the context name, not the database name.


You'll also want to note the information about changes made to the data 
structures for Asterisk 1.8.


As far as your request about tracking the time a call is in the queue, 
that's information that is directly available in the queue_log. One 
important question that you haven't asked is How do I track how long 
each user was logged in to the queue, even if they received no calls?. 
That will require additions to your login/logout context that write 
entries to the log each and every time a user logs in/out. You can then 
report on that data.


You might want to reconsider reinventing the wheel on this one. Have you 
checked into Queuemetrics at http://www.queuemetrics.com ?


Tom

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Re: [asterisk-users] queue_log in MySQL database

2011-01-13 Thread Tom Rymes

On 01/13/2011 2:07 PM, Tom Rymes wrote:


That will require additions to your login/logout context that write
entries to the log each and every time a user logs in/out. You can then
report on that data.


While there's a thread going on about this topic, and while I've written 
the above comment, can anyone confirm that the QueueLog command will 
indeed write entries out to the realtime queue_log, not just the file 
based log?


Many thanks,

Tom

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Re: [asterisk-users] Mail list Woes?

2011-01-09 Thread Tom Rymes
On Jan 9, 2011, at 8:27 AM, William Stillwell wrote:

 Anybody notice log delays in this list, and very small amount of traffic?

I have noticed multiple hour delays between sending messages and seeing them 
back.

Tom

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Re: [asterisk-users] Benefit of PRI vs SIP trunk calls

2011-01-08 Thread Tom Rymes

On Jan 6, 2011, at 8:08 PM, Joel Maslak wrote:

 On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson dicken...@cfmc.com wrote:
 Are there reasons to prefer the use of PRI over SIP or SIP over PRI?

[snip]

 I run the PBX for my organization which has about 160 extensions.  I
 wouldn't even think of doing anything but PRI for the main lines
 because (A) for our size organization where we are located, we're
 talking a couple hundred dollars a month difference between PRI and
 SIP in cost so it's nearly break-even in cost which means cost
 difference isn't a huge motivator, (B) it supports FAX, modems, and
 TTYs - perfectly, (C) Quality is 100% consistent.  In addition, the
 reliability is good enough that I'm willing to use it for 911.

[snip]

I have to agree with most of what Joel said in his message. For me, the main 
problem with many sip implementations is that your phone service will be only 
as reliable as your internet service. If you have a dedicated internet line 
that is highly reliable, that's not a big deal, but DSL, Cable, and the like 
aren't reliable enough for our needs.

Having said that, one downside of a PRI is that you are paying for all of those 
channels, even when you aren't using them. Companies like Paetec and most other 
large telcos are offering SIP trunks over an MPLS circuit, running on a T1 
loop. This covers the reliability problem, as you are running over the same 
type of circuit as your PRI, and it allows you to take advantage of unused 
channels as data bandwidth. This is especially helpful for folks who have a 
data T1 and a PRI, as they can get higher bandwidth for data when there isn't 
much voice traffic. Because they use G.729, you can also fit more calls on the 
same circuit. That choice of codec eliminates the ability to send/receive 
faxes, though, and it's likely expensive when compared to other SIP solutions, 
but it does appear to be pretty slick. 

Another benefit of SIP is that it doesn't require a Digium, Sangoma, or similar 
interface card in the server, simplifying migrations and reducing cost in many 
scenarios. 

Tom
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Re: [asterisk-users] Too Few Fax Detections

2011-01-07 Thread Tom Rymes

On Jan 6, 2011, at 8:56 AM, Kevin P. Fleming wrote:

 On 01/05/2011 08:12 PM, Thomas Rymes wrote:
 OK, after my last message about fax detection, I feel a bit better informed 
 and able to press forward. I started looking into this because I was getting 
 lots of false positive fax detection errors in the logs with faxdetect=both 
 set in chan_dahdi.conf.
 
 Anyhow, I do not currently use fax detection, and we have a dedicated Fax 
 DID on our PRI, so setting faxdetect=no works fine. Having said that, I 
 would like to sort it out as I may want to use fax detection in the future. 
 Unfortunately, I seem to be having odd results. I set faxdetect=incoming 
 last night and restarted dahdi and asterisk. Since that time, we have 
 received 17 faxes, but I only have three fax detections in my asterisk log, 
 so far as I can tell:
 
 # grep -i fax /var/log/asterisk/full
 [Jan  5 05:53:39] NOTICE[6686] chan_dahdi.c: Fax detected, but no fax 
 extension
 [Jan  5 10:24:27] NOTICE[11834] chan_dahdi.c: Fax detected, but no fax 
 extension
 [Jan  5 11:48:52] NOTICE[13804] chan_dahdi.c: Fax detected, but no fax 
 extension
 
 All three calls listed are indeed fax calls, and since there is no fax 
 extension in that context, the call just proceeds along as if nothing 
 happened (which is appropriate).
 
 My question is this: If I have received 17 faxes since enabling fax 
 detection, shouldn't I see ~17 entries in the log?
 
 How are you delivering the inbound FAX calls to your FAX machine? If you are 
 sending them back out a DAHDI channel (to an FXS port on an analog card, for 
 example), then as soon as the two channels are bridged the audio never comes 
 up to Asterisk (under normal circumstances), it stays in DAHDI, so the 
 Asterisk DSP can't detect the CNG tone. If the FAX machine answers the 
 incoming call fairly quickly, there may not be any opportunity for the CNG to 
 be detected. In addition, you may not be even receiving any audio from the 
 calling FAX machine until you answer the incoming channel (depending on your 
 PRI provider).
 
 If you want to have the best chance to detect each incoming FAX using the 
 Asterisk DSP, you'll have to answer the incoming channel as soon as it hits 
 the dialplan, then wait 3 or 4 seconds, then send the call onwards to your 
 actual FAX machine. FAX detection is really expected to be used on calls that 
 would otherwise be answered by a non-FAX endpoint (IVR, voicemail, user with 
 a phone, etc.)

That does make sense: the incoming calls are directed to a FreePBX ring group 
consisting of three IAXModems handled by HylaFAX, and I am fairly certain that 
they answer the call nearly instantaneously. 

I do realize that the detection is really intended for non-dedicated lines; I'm 
just trying to ensure it works before I start using it. 

Thanks for the response.

Tom
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Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)

2011-01-07 Thread Tom Rymes
On Jan 6, 2011, at 10:05 AM, Andy Graybeal wrote:

 On 01/05/2011 01:51 PM, Tom Rymes wrote:
 On 01/05/2011 7:50 AM, Andy Graybeal wrote:
 
 We've got two noisy kitchens that need to talk back and forth.
 
 Andy,
 
 Why, exactly, are you trying to combine an inter-kitchen intercom and
 your phone system? Might it make more sense to have a non-phone-based
 intercom system, plus a phone for making phone calls?
 
 Tom
 
 Tom,
 Good question.  I'm not sure, but maybe I was hoping to kill two birds with 
 one stone.
 
 I will take your suggestion into account as I'm not sure what to do.
 
 Do you have any intercom system recommendations?  Would it be POE also, and 
 something I could manage with Asterisk?
 
 -Andy

Unfortunately, I have no recommendations, but I was just thinking of a simple, 
dumb intercom for between the kitchens, plus a phone for when you need to make 
a call. Any old phone will do. Perhaps something simple and durable like this:

http://cgi.ebay.com/2554-Single-Line-Wall-Phone-w-Amp-Handset-Cortelco-AT-T-/200552549353?pt=LH_DefaultDomain_0hash=item2eb1dd0be9#ht_3437wt_1141

Tom
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Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)

2011-01-05 Thread Tom Rymes

On 01/05/2011 7:50 AM, Andy Graybeal wrote:


We've got two noisy kitchens that need to talk back and forth.


Andy,

Why, exactly, are you trying to combine an inter-kitchen intercom and 
your phone system? Might it make more sense to have a non-phone-based 
intercom system, plus a phone for making phone calls?


Tom

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Re: [asterisk-users] Clarification on DAHDI Fax Detection

2011-01-04 Thread Tom Rymes

On 01/04/2011 8:55 AM, Kevin P. Fleming wrote:

On 01/03/2011 06:47 PM, Thomas Rymes wrote:

On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote:

On 01/03/2011 11:26 AM, Tom Rymes wrote:


[snip]


OK. Either way, though, the changes to echo cancellation are not
affected by the faxdetect setting, right?


That is correct; the faxdetect setting and the echo canceller behavior
are completely unrelated.


Excellent.

[snip]


Is there a time limit to when DAHDI listens for faxes (say the first
10 seconds of a call?), or might it detect one in the middle of a ten
minute call?


I haven't double-checked, but I believe the software DSP will be in
place on the call until it sees a CNG tone, regardless of when that
happens during the call.


Wouldn't it make sense to be able to specify a time period after which 
chan_dahdi disables fax detection? Only calls that begin with a voice 
call and end with a fax would benefit from detection after the initial 
~8 seconds of a call, unless I am overlooking something.


If the DSP keeps listening and detects a spurious fax tone (I know I 
have seen the human voice incorrectly identified as CNG), it will send 
the call off to the fax extension if one exists in the same context. In 
fact, we ran into some issues with exactly that happening.


[snip]


Thanks for the clarification, there's a lot of conflicting info out
there.


Feel free to comment on wiki.asterisk.org if any of the information
there led you astray; we'd like to get that to be the most accurate
place for people to find this sort of information.


I'll give it a look. I had not specifically looked at the asterisk wiki, 
but Google searches brought up lots of messages confusing the fax 
operation of the echo canceler with the faxdetect= setting for DAHDI/Zaptel.


Thanks again,

Tom

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Re: [asterisk-users] VoIP PoE phones for restaurant

2011-01-04 Thread Tom Rymes

On 01/04/2011 8:52 AM, Andy Graybeal wrote:


Is it possible that I can run one cable to the phone, then run a cable
from the phone to a computer or another device and have those the phone
and computer or other device be on separate networks?
I'm sorry if this sounds newbish; I'm still learning.


I'm no networking expert, but no one else has answered, so I'll give it 
a shot.


It is indeed possible (quite common, actually) to run the wiring as you 
describe. If you want to keep the data and voice traffic separate, you 
can use VLANs to do so. Your switches will need to support VLANS, and 
you will need to configure VLANs to separate the voice and data traffic.


As I understand it, though, you are still subject to the bandwidth 
limitations of the underlying network, so it's still possible that heavy 
traffic from the PC might affect the voice traffic. QOS or other methods 
might be used to help avoid this.


For this reason, I personally prefer to keep my voice and data LANs 
physically separated when possible. Obviously, cost and complexity do 
increase somewhat. It's probably not a good solution for everyone, but 
it sounds like you have a pretty small installation and you might decide 
that the additional cost is justified.


Tom

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Re: [asterisk-users] MOH problems (asterisk 1.4.38)

2011-01-04 Thread Tom Rymes

On 01/04/2011 12:31 PM, Earl Terwilliger wrote:

Hi list,

I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am
getting this error :

WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such
file or directory


[snip]

Have you installed mpg123 or some other program to handle the mp3 files? 
I am fairly certain that Asterisk cannot handle mp3 natively (most 
likely for licensing reasons).


Tom


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[asterisk-users] DAHDI and dialdebounce

2011-01-04 Thread Tom Rymes
According to https://issues.asterisk.org/view.php?id=16339 , the default 
value for the dialdebounce parameter of the wctdm module has been 
changed to 32 and is now user configurable.


I have two questions:

1.) Am I correct in presuming that, if the default of 32 does not work 
for me, I would specify this option in the file 
/etc/modprobe.d/dahdi.conf (at least for my distro, which is Elastix 
built on CentOS)?
2.) In the issue linked above, Tzafrir asked if this value should also 
be changed for the wctdm24xxp module, but there is no indication as to 
whether the change is needed for wctdm24xxp, or if it has already been 
made. Can anyone clarify?


Many thanks,

Tom

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Re: [asterisk-users] Replacing digital pri card

2011-01-04 Thread Tom Rymes

On 01/03/2011 9:46 PM, Matt Watson wrote:

I don't imagine this would be too complicated - don't have any
experience with AsteriskNOW - but on a 'vanilla' linux distro it would
just be a matter of making sure dahdi is loading the correct drivers and
doing a couple of minor config file updates.



On Tue, Dec 28, 2010 at 3:01 PM, Tyler Davis tda...@zulily.com
mailto:tda...@zulily.com wrote:

I need to replace our current 1 port pri card with a quad port card.
I'm currently using the newest AsteriskNOW distro. Are there any
issues I should expect to run into? I'm hoping the transition will
be smooth, however I havent had to do this in the past.


It should be reasonably easy, but you will need to update your DAHDI 
configs, including chan_dahdi.conf. I think that Digium developed and 
included a DAHDI configuration module for FreePBX that makes that more 
intuitive.


If you use a non-Digium card, you'll need to update those 
configurations, too.


Tom

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[asterisk-users] Clarification on DAHDI Fax Detection

2011-01-03 Thread Tom Rymes

Hi folks,

I was hoping that someone might be able to help clarify some confusion I 
have on DAHDI Fax detection after spending some time searching. My 
understanding is this:


1.) Echo cancellation is automatically disabled upon recognition of a 
CNG tone, regardless of the faxdetect setting. This can only be disabled 
at compile time.
2.) faxdetect=incoming will, upon detection of a CNG tone, send the call 
to the fax extension.

3.) faxdetect=outgoing will ??

Also, do Digium cards with HW Echo Cancellation detect the CNG tones in 
hardware? If so, how does the faxdetect setting in DAHDI affect that 
behavior?


Many thanks,

Tom

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Re: [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?

2011-01-03 Thread Tom Rymes

On 01/03/2011 11:20 AM, Bruce B wrote:
Thanks a lot for that Sebastian. I will report back my findings when I 
find the resolution on this.


I'm a bit late here, but I can say that Sangoma support has always been 
extremely helpful when I call them. If you haven't already, definitely 
give them a call.


Tom

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Re: [asterisk-users] Asterisk GUI

2007-06-21 Thread Tom Rymes
On Jun 20, 2007, at 5:04 PM, Troy Ayers wrote:

 I would have been convinced if you had not top-posted!  heh


 Rob Schall wrote:
 Tom,

 I disagree with your argument for a number of reasons. Each of these
 reasons should be more than enough to convince you I'm correct and  
 you
 should do it my way and only my way.

 And for the record, VI and CLI.

 Rob

OK, Now I'm confused... I was prepared to accept Rob's argument due  
its beautiful, flawless logic. But Troy has a valid point: Rob did  
top-post, invalidating his point. But so did Troy, invalidating his  
point, so now I'm stuck. Whatever shall I do?

I think I'll just stick with my own opinion, seeing as both Rob and  
Troy are obviously idiots. (duh!)

;-)

Tom

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Re: [asterisk-users] Asterisk GUI

2007-06-20 Thread Tom Rymes
On Jun 19, 2007, at 12:37 PM, Senad Jordanovic wrote:

 Tom Rymes wrote:

[snip]


 How many times does it have to be said? Don't feed the trolls!

 Tom


 Tom...Who in your opinion is a troll?


 Senad

Well, technically, I was calling the original post a troll, not the  
original poster. More specifically, the usage of troll I am referring  
to resembles the fishing technique more than the mythological  
creature. Basically, a troll in this context is a post that someone  
makes simply for the purpose of starting a heated discussion on a  
very touchy subject. In other words, the original poster is  
trolling for people who will get all bent out of shape about their  
post and fire back a heated response.

For example, a user could post a message to the list asking I'm new  
to Linux and Asterisk. Should edit my dialplan by hand, use FreePBX,  
or buy a commercial solution? Imagine the response as you tried to  
convince them to buy PBXWare, FreePBX users try to convince them that  
they should start out using FreePBX, and others go on about how hand  
coding a dialplan is the one-true-way® to learn Asterisk. Generally,  
the original poster is just looking to get everyone stirred up over  
nothing.

In other words, Paul's original post of GUI bad! CLI good! was just  
the sort of post that is going to get folks fired up re-re-restarting  
the age-old discussion of which is better: CLI or GUI. Basically, it  
could be like posting any of the following:

- Which is better: emacs or vi?
- Which linux distribution is the best?
- Which is better: Macs or Windows?

All of these questions share the following:

1.) They have no right answer (macs are better for some, Windows for  
others, and linux for others still, not to mention OS/2, BSD, etc)
2.) People on the various sides of the debate have extremely strong  
feelings on the matter
3.) Nobody is likely to be convinced that the other side is right and  
that they are wrong.
4.) They have all been discussed thousands of times before, and  
nothing new is likely to be said on the matter.
5.) The only purpose served by the discussion, due to the reasons  
above, is to clutter up the mailing list.
6.) Any discussion thread regarding these sorts of topics is best  
avoided.

For a more thorough description of an internet troll, see the  
following wikipedia article:http://en.wikipedia.org/wiki/Troll_% 
28internet%29

In other words, if you see a post that is just going to result in a  
re-rehashing of the last rehash of a specific subject, just hit the  
delete key instead of clogging up the mailing list with yet another  
thread on whether a GUI or a CLI is better. (for example).

In Paul's defense, it looked to me like his original post was simply  
a joke that was misunderstood. (I thought it was funny, anyway)

I suppose I should take my own advice on this one, but sometimes I  
guess we all just can't resist. grin

Tom
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Re: [asterisk-users] Asterisk GUI

2007-06-19 Thread Tom Rymes

On Jun 16, 2007, at 4:37 PM, Tzafrir Cohen wrote:

 On Sat, Jun 16, 2007 at 08:55:24PM +0100, Senad Jordanovic wrote:
 Brett Crapser wrote:
 On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote:
 Paul Hales wrote:
 GUI bad! CLI good!

 PaulH

 Really...?

 So explain why every major PBX manufacturer has GUI of some sort?
 Surely they would have had CLI only if GUI is bad!!!


 Senad

 Senad - it is really to cover the inability of 'average' people to
 understand CLI.

 CLI is useful for small/simple dial tone installations. Anything  
 above
 that even very competent administrator will make syntax/logical  
 errors.

 Hence automation is required. Automation does not imply GUI.
 Bad GUIs get in the way of automation.

How many times does it have to be said? Don't feed the trolls!

Tom

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Re: [asterisk-users] NAT

2007-06-05 Thread Tom Rymes

On Jun 5, 2007, at 9:46 AM, Cosmin Prund wrote:


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Henry Cobb
Sent: Tuesday, June 05, 2007 4:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] NAT

On 6/5/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote:

Hi All!!

I have my asterisk working in my house (working with mandriva 2007

and

asterisk 1.4 svn). I´ve looking on the net but i couldn´t find the

way of

making work my asterisk in a real enviroment. Seems that the problem

of NAT

is a big problem. How can I sort out this, a mean crossing the NAT

and

having asterisk connected?


If you want to receive calls and not just place them and you have a
broadband connection with a dynamic IP then your server must register
with the VoIP provider and I suggest using IAX with the proper UDP
port assigned to your Atrisk server.

-HJC



NAT is not that big of a problem, not anymore.
Do a NAT search on http://www.voip-info.org - it'll get you  
started (got me started at least)


--
Cosmin Prund


Specifically, you need to set the following in sip.conf (if applicable)

nat=
localnet=
externip=
externhost=

You also need to configure your router to forward port 5060 and ports  
1-2 to your asterisk server.


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Re: [asterisk-users] SIP NAT ...

2007-06-01 Thread Tom Rymes

On Jun 1, 2007, at 9:45 AM, Gordon Henderson wrote:

[snip]

Both these SIP - external PSTN provider connections register OK on  
the * box, and outgoing calls placed over either connection works  
perfectly. Outgoing callerId (set by the external provider) works  
as expected. ) I have dialling prefixes for each 'line', nothing  
special there, that side of it all works as expected.


The problem is that only the last one in the sip.conf file actually  
accepts incoming calls when dialled from the PSTN side. (They have  
different PSTN phone numbers) If I swap their entries over in the  
sip.conf file, then the other one takes the calls.


[snip]

I may be mistaken here, but don't you need to use different ports for  
each line? ie: Port 5060 for line 1 and 5061 for line 2?


Tom
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Re: [asterisk-users] vmoutcall

2007-05-28 Thread Tom Rymes

On May 24, 2007, at 3:28 PM, Doug Lytle wrote:


Paul Aviles wrote:

Hello guys,

I have been looking for a way to call a cell phone after someone  
has left a



This can easily be done with database lookups and .call files

to accomplishing this? Most analog pbx's have this feature and I  
am amazed

Asterisk does not natively.


It can be done natively; within the dial plan.

Doug


An aside, this can also be done using the externnotify option in the  
voicemail.conf file. That option allows you to specify an external  
script that will run when VoiceMailMain() exits. Watch out, because  
(as I just found out yesterday, the hard way), this script is run  
both when a voicemail message is left and when a user logs out of  
voicemail after checking their messages.


From there, create a script in your favorite language (a simple  
shell script ought to work for your purposes) and have it create a  
call file to call out to the user and drop them into a context that  
plays an announcement (You have new voicemail) and then asks them  
to confirm with a keypress (Dial 2 for voicemail). At that point,  
set the IVR up so dialing 2 drops them into Voicemail() and you are  
good to go.


Tom
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Re: [asterisk-users] Integrated T1

2007-05-24 Thread Tom Rymes

On Thu, 24 May 2007, Jeremy Mann wrote:

 Can an asterisk box equipped with a Digium T1 card handle  
Integrated T1
 circuits?  I have a T1 with 768k data and the remaining channels  
voice,

 can the asterisk box do the Data routing + Voice processing?


I'm not certain, but I believe the Sangoma WANrouter/WANPipe cards  
are capable of this. Call Sangoma and ask them if it is possible.


Tom

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Re: [asterisk-users] TDM400P usada?

2007-05-05 Thread Tom Rymes

On May 5, 2007, at 12:06 PM, Rodrigo Mercado wrote:

Alguien tiene una TDM400P con modulo FXS usada a la venta ??,  
obviamente a precio de tarjeta usada...



saludos,


Rodrigo Mercado S.


For anyone who is not a Spanish speaker, Rodrigo is looking for a  
used TDM400P card with FXS modules. He is expecting a price that  
would correspond with a used card. (In other words, cheap)


Rodrigo:

1.) ¿Donde estás? ¿Cómo podria alguien dar un precio sin saber donde  
tendria que mandarlo? ¿España? ¿Puerto Rico? ¿Argentina?
2.) Si no hablas Inglés, seria mejor buscar una lista de Asterisk en  
Español, porque la mayoria de las personas aqui no hablen Español.


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Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation

2007-05-04 Thread Tom Rymes


On May 4, 2007, at 10:08 AM, Stephen Bosch wrote:


Tom Rymes wrote:


On May 3, 2007, at 12:20 PM, Stephen Bosch wrote:


Mats Karlsson wrote:

Take a look here:
http://www.voip.com.sg/voip_products/ 
voip_ip_phone_provisioning_tool.html




Ugh. This is a Win32 app, isn't it?


Wow,

The guy makes a useful application and provides it to the  
community for
free and you have the cojones to bitch and moan b/c it's a windows  
app?

Talk about looking a gift horse in the mouth!


[snip]


B. It's not a gift horse for me, because it's totally use*less* to me.


[snip]

It *is* a gift horse; he gave it to you, after all. The expression  
To look a gift horse in the mouth implies that someone gave you a  
horse and you looked into its mouth to see how old it is and whether  
it is of value to *you*. A modern comparison might be pulling up  
froogle to see how much a gift cost as someone gives it to you.


In other words, taking a gift from someone and questioning its value  
to you in front of that person's face is rude, inconsiderate, and bad  
form. I would understand a post along the lines of: Wow, that looks  
like a great program from the description, but it's a windows only  
app, and I don't run Windows. Does anyone know of something similar  
for Linux?


I dunno, I guess I'm not your mother, but then again, it seemed  
pretty rude considering the guy offered the program for free and you  
were criticizing the fact that he didn't develop a free linux app for  
you, too.


Tom
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Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation

2007-05-03 Thread Tom Rymes


On May 3, 2007, at 12:20 PM, Stephen Bosch wrote:


Mats Karlsson wrote:

Take a look here:
http://www.voip.com.sg/voip_products/ 
voip_ip_phone_provisioning_tool.html


Ugh. This is a Win32 app, isn't it?


Wow,

The guy makes a useful application and provides it to the community  
for free and you have the cojones to bitch and moan b/c it's a  
windows app? Talk about looking a gift horse in the mouth!


Tom
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Re: [asterisk-users] Summary of Trixbox vs. custom install

2007-02-16 Thread Tom Rymes

On Feb 15, 2007, at 7:01 PM, Stefano Corsi wrote:

Hello everybody. First of all thanks to all the people giving their  
opinion on the subject I proposed: Trixbox vs. custom install.  
You've all been very helpful.


[snip]

I also include a consideration from mine: I would happily use  
Trixbox, because I did FreePBX setup once and it was a real pain,  
but I'm very frightened by a few issues:


1) Trixbox Macho installation that installs everything without  
asking. I, for example, would like to use software RAID (maybe it's  
wrong with Asterisk, but I want to do it!). I wouldn't like doing  
it manually after Trixbox installation. I would like to have an  
installer doing it for me. Centos (ex redhat) installer does it, so  
why Trixbox choose to install everything without prompting?


Stefano,

Great summary. As an aside here, it is possible to install Trixbox on  
top of an existing CentOS installation by using the tarball, not the  
ISO. This works very well, with one issue I ran into. A fresh install  
of CentOS updated via yum will not have the correct version of the  
kernel to match the zaptel-modules RPM shipped with Trixbox (because  
it is no longer in the repositories). You can fix this problem two ways:


1.) Manually install the kernel from the Trixbox CD, which will fix  
the problem, if you prefer to work just the way Trixbox normally  
does. You should configure yum to not upgrade the kernel in this  
case, because that would break zaptel.
2.) You can download and manually recompile zaptel on your own.  
Either you will have to recompile zaptel every time that the kernel  
is upgraded by yum, or you should configure yum to not upgrade the  
kernel. (This is true of any zaptel install, not just Trixbox.)


See the bug i posted: http://www.trixbox.org/modules/xproject/ 
index.php?op=viewTicketMainid=27


Another resolution would be to provide an SRPM for the zaptel-modules  
package, which you (or the tarball install script) could rpmbuild -- 
rebuild against your current kernel.


Either way, this isn't a big problem so long as you know it's there.  
Worst case scenario, you just download and compile zaptel, which you  
would have had to do anyway for a non-trixbox install.


Tom
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Re: [asterisk-users] Trixbox vs. Custom install

2007-02-13 Thread Tom Rymes

On Feb 12, 2007, at 7:13 PM, Stephen Bosch wrote:


Lee Jenkins wrote:

Stefano Corsi wrote:


[snip]

The nice things about GUI's in my opinion is that routine chores  
such as

setting up extensions, dialing extensions, hunt groups, etc. are less
likely to contain scripting bugs or typos.  The downside from what I
gather with many GUI's is that the friendly abstraction that  
insulates
you from the nuts and bolts of scripting and configuration also  
makes it

difficult to customize the dialplan in some cases.


It also makes troubleshooting problems a handful-and-a-half. And  
woe is

you if you need kernel customizations to make your hardware work.


Not to start a flame-war, but I completely disagree. Troubleshooting  
a GUI is much easier, given that you don't have to scout for typos,  
transposed numbers, etc throughout the dialplan. With the GUI, you  
have to double check the information that you input into the GUI, but  
that's it. As for hardware, it should be no more difficult to get  
Trixbox to play nicely with hardware than any other Asterisk install.  
You may have to patch and/or recompile zaptel, asterisk, etc, but  
that's no different than what you would have to do with a non-Trixbox  
install. (and you really shouldn't have to in almost all cases)



I would say this -- if all you're ever going to use is VOIP trunks, by
all means use Trixbox. It's great for that. But if you're using any  
kind
of PSTN hardware (TDM cards, Sangoma) just stick with straight  
Asterisk.


Are you kidding? Sangoma actually has a version of Trixbox on their  
site that comes bundled with their drivers already installed (see  
http://wiki.sangoma.com/Trixbox-1xx ). All you have to do is  
configure the card(s) in the same way as you would with any Asterisk  
install.



I've just had my second go at Trixbox (version 2.0 now) and after
wasting a bunch of time with hardware problems, I'm going to  
replace it

with a generic install.


I would suggest (hopefully politely) that you not blame your lack of  
experience and ability on Trixbox. If you can get the Sangoma  
wanrouter software downloaded and compiled, along with Zaptel,  
Asterisk, libpri, etc, then you can certainly do the same on Trixbox,  
because all you have to do is yum search wanpipe  and then yum  
install the modules and utils packages. Once installed, follow the  
instructions on Sangoma's website to configure the card. If all else  
fails, you can easily call for support from Sangoma. Even if you  
choose not to use yum, it's just as easy to get a Sangoma board  
working under Trixbox as it is for any other Asterisk install.



Here's another reason to seriously consider generic: the userbase is
larger, AND they're more likely to know what they're talking about  
when

a problem does arise. Trixbox attracts a lot of amateurs who are
themselves new to IP telephony; that's why they choose it.


Valid point, but FreePBX (the program Trixbox uses for GUI Asteirsk  
config) also has a large userbase, and a number of Trixbox problems  
are not Trixbox specific, and can be addressed by the Asterisk  
community as a whole.


Of course, you should take this with a grain of salt since I tried  
[EMAIL PROTECTED]

(now TrixBox) for a total of 2 weeks before gutting it.


There is a good reason people don't stick with it for long.


Many people do not stick with Trixbox for long, and many others do.  
The crux of the issue is this: FreePBX/Trixbox, and most other GUIs  
will make it easier to get your system up and running, and they make  
it easier to maintain it, make changes, etc. (I am defining easier  
as requiring less technical familiarity with the underpinnings of  
exactly what is going on as well as less intimidating and error  
prone since no manual editing of configuration files is required.)  
On the other hand, emacs/vi/pico/whatevereditoryouprefer and the text  
config files without a GUI are more difficult, but offer greater  
flexibility.


S it comes down to Which is more important to you? Ease of  
use for you and/or your clients (who may want to control adds/moves,  
etc.) or greater flexibility and control? Once you answer that  
question, you can answer the question Which is better for me? The  
correct answer to that question may very well be different for you  
than it is for me. (and it may be different for you six months from  
now than it is today.)


Tom

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Re: [asterisk-users] Trixbox vs. Custom install

2007-02-13 Thread Tom Rymes

On Feb 13, 2007, at 11:53 AM, Tzafrir Cohen wrote:


On Tue, Feb 13, 2007 at 10:23:17AM -0500, Tom Rymes wrote:


[snip]


Not to start a flame-war, but I completely disagree. Troubleshooting
a GUI is much easier, given that you don't have to scout for typos,
transposed numbers, etc throughout the dialplan. With the GUI, you
have to double check the information that you input into the GUI, but
that's it. As for hardware, it should be no more difficult to get
Trixbox to play nicely with hardware than any other Asterisk install.
You may have to patch and/or recompile zaptel, asterisk, etc, but
that's no different than what you would have to do with a non-Trixbox
install.


Hmmm... I installed a trixbox system. 'yum update' failed to work, due
to funny games with yum's configuration. A default centos server
installation did not have the same issue.

This is just one example.


I have never run into this problem before, and the only change that I  
know of was to exclude the kernel from updates (to avoid having to  
recompile zaptel) Of course, if you want to update the kernel, change  
the yum settings and download and recompile zaptel. YMMV, so if it  
doesn't work for you, then act accordingly, I suppose. As a  
counterpoint to your example, I have installed Trixbox easily and  
successfuly many times with Sangoma hardware.



(and you really shouldn't have to in almost all cases)


A GUI does its absraction. By that it hides some information that it
deems irrelevant. In many cases this information is relevant.


My point that you quoted originally referred to the fact that you  
shouldn't normally have to recompile Zaptel, Asterisk, or anything  
else to get hardware working with Trixbox.  As for your comment about  
the GUI, I agree. My earlier e-mail tried to state that neither the  
GUI or the non-GUI method of installing and configuring Asterisk is  
better. The GUI is better for some, whereas the non-GUI is better for  
others. If the limitations imposed by the GUI are too much for your  
application, then the GUI isn't for you. If the relative difficulty  
of administering an Asterisk server without a GUI is too much for  
your application, then use the GUI.


One example: just figuring out if FreePBX actually dial, or not at  
all,

requires either a sufficiently-trained asterisk guy to review the
log/cli just to understand why a call did not go through.


I fail to see how this is different from a non-FreePBX setup? Don't  
you still need a sufficiently-trained Asterisk Guy to view the logs  
and CLI to determine why your custom dialplan didn't dial? Not to  
mention to create that custom dialplan in the first place? How does  
troubleshooting a non-GUI asterisk install require less technical  
know-how than troubleshooting a Free-PBX system?


Anyhow, I reiterate that I don't think that either solution is better  
than the other. Determine your requirements, weigh the pros and cons  
of the various GUIs and of running without a GUI and see which is the  
best fit for your requirements. I only object to those who say that  
No one should use Trixbox/FreePBX, it's too restrictive or Running  
Asterisk with a GUI is always Better. Both statements are erroneous.


Tom
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Re: [asterisk-users] Delay in Call Distribution using the Queue Application

2007-01-19 Thread Tom Rymes

On Jan 15, 2007, at 2:22 PM, [EMAIL PROTECTED] wrote:


Hello all,

we're using asterisk 1.2.12.1 in an Inbound callcenter using the  
queue application. If there are many calls in the queue, it  
sometimes takes up to 30 Seconds before a call is distributed to an  
agent.


For example there are 10 callers in the queue, an Agent is  
finishing a call and it takes up to 30 seconds before his phone  
rings again. We're already set the wrapuptime parameter in  
queues.conf to 0, for my point of view an agent phone that  
becomes available again should ring immediately after hanging up a  
call.


Does anybody know if there are any known issues or restrictions in  
the queue application in version 1.2.12.1?


You may be running into the limitation in Asterisk 1.2 (It's fixed in  
1.4, I think double check that) in how the queues distribute  
calls. Basically, the queue can only distribute one call at a time,  
so if you have two agents, both available, and two calls in the  
queue, asterisk will send call #1 to agent #1 first. Once that call  
is connected, Asterisk will then send  call #2 to agent #2. In other  
words, until asterisk distributes the first call, it can't distribute  
any other calls waiting in line.


One nasty side effect of this is that an agent who fails to log out  
and leaves their desk will add 30 seconds or so (the amount of time  
their phone rings before the queue gives up and tries the next agent)  
wait time to all of the calls waiting in the queue.


Tom
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Re: [asterisk-users] Identifying Queue on Cisco 7960

2007-01-12 Thread Tom Rymes


On Jan 12, 2007, at 10:07 AM, Robert Norton - SophMedia LLC wrote:


Hey Guys,
I apologize for my ignorance on this one.

I've got several 7960s running on Asterisk1.4 with 15 or separate  
queues and am trying to figure out a way to identify to the 7960s,  
what queue the incoming call is on? Is this possible at all?


Set your dialplan to prepend the queue name to the callerID  
information. For example, change John Smith to QueueName - John  
Smith


Tom
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Re: [asterisk-users] MWI across multiple servers

2006-12-07 Thread Tom Rymes

On Dec 7, 2006, at 4:14 AM, Jon Farmer wrote:

I decided to write my own simple voicemail application via AGI and  
store all voicemails in MySQL. The nice thing was the user can  
retrieve via phone (local and remote), via email attachment and  
also via web download.


You can listen to old and new messages and change your outgoing  
message too.


Regards

Jon


Jon,

Maybe you could post this application and a how-to to the wiki?

Tom
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Re: [asterisk-users] How to park calls on a specific extension

2006-12-01 Thread Tom Rymes

On Nov 30, 2006, at 8:55 PM, Brad Templeton wrote:


On Thu, Nov 30, 2006 at 02:50:21PM -0500, Tom Rymes wrote:

for example: In your example above where they can't figure out how to
transfer, why don't you edit features.conf and define the transfer
key as # or something. Then, when they have a call for Bill across
they way, they can do this:


In this case don't they need to have a t in every Dial as well?
And then there's the other direction.  Sometimes (actually quite
often) I like to transfer a call that I dialed, which requires
the T but means you interfere with typing touch tones to IVRs
that you call.


I'd have to double-check on that, but I am fairly certain the you  
would indeed need to put the 'T' and 't' options in the dial string.  
I think you would want to arrange your dialplan to make sure that you  
don't inadvertently make it so that incoming callers can transfer  
their own calls coming in. In other words, Dial command for incoming  
calls would have 'T' and for outgoing calls would have 't'. Double  
check the docs, though, because I am pulling this from (spotty)  
memory. As for not interfering with IVRs, I would suggest a two  
character transfer, such as #8, which reads #T if you look at the  
keypad. Tell your users, Press #T to transfer a call. I also use #7  
for pickup, as it is #P. Same sort of idea as using *86 or *VM  
for voicemail.



No, almost all IP phones have a transfer button, the nice thing
would be if somehow the UI for that could have been standardized,
at least for the phones that don't have screens and soft buttons
(which can extend the interface because they can show it to you).


Better configurability of the SIP phones would be nice. We use Cisco  
7940 phones, and changing the soft buttons on the bottom of the  
screen to hold commonly used functions would be nice, but AFAIK, that  
cannot be done in the SIP firmware.


Tom
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Re: [asterisk-users] How to park calls on a specific extension

2006-11-30 Thread Tom Rymes

On Nov 29, 2006, at 11:40 PM, Lacy Moore - Aspendora wrote:

[snip]

I went from a Lucent Merlin Legend system to Asterisk.  For me,  
it's a tradeoff for features.  To my users, it was a step  
backward.  I also upgraded an office from a Partner system to  
Asterisk.  To the users, it is a huge step backward.  They have yet  
to figure out how to transfer a call.  On their old system, they  
put the call on hold and pressed the line button at another phone.   
Today, they hold the phone against their leg so the caller doesn't  
here them yell for the person to come to the phone, and then the  
person who the call is for comes to that phone and answers the  
call.  It will remain that way for them, because learning how to do  
it the right way takes more work than the person coming to the  
phone, or so they say.


Sounds like you have stumbled upon one of the truisms of replacing  
office phone systems: people hate it when you take away their key  
system lines. So you're right, it would be a good idea for Asterisk  
to implement that functionality, and they are working on it, IIRC. It  
also sounds like you need to talk to your customers more before you  
roll out a system, so they know ahead of time what the interface will  
be and what changes they should expect. If you let them know ahead of  
time, and get management on board, you should be OK.


I won't be doing another Asterisk install for a while.  Customer #2  
has made sure of that by telling everyone how their new phone  
system sucks.  Until I can find a suitable solution, I am dead in  
the water.  And yes, I am trying to learn C so that I can write it  
myself, or modify something else to make it work.


Given the flexibility inherent in Asterisk, you really shouldn't have  
to code your own. It's a great skill, but not necessary.


But seriously, the attitude of either write it yourself or deal  
with it won't cut it for business users.  If Asterisk is only for  
geeks, then fine, it will work perfectly.


Well, not to be rude, but if you plan to sell, install, and maintain  
Asterisk systems, you shouldn't be just a business user, you should  
be at least a little bit geeky. I would suggest that Asterisk works  
excellently for business users, but it requires a person who is a bit  
of a geek to set it up properly for those business users so they  
don't notice how geeky it really is.


 If all phones behaved the same, it would help.  Cisco, using SIP,  
has no park button.  Cisco, using chan_sccp, has a great parking  
concept.  Polycom has a park button that doesn't appear to work  
with Asterisk.  We use Cisco (SIP) and Polycom.  Aastra and SNOM  
seem to have an easier parking interface.  The chan_sccp  
implementation not only reads back the parking spot, but also  
displays it on the screen.


Why don't you take the specific phone interface out of it? Most of  
your (and your users') gripes seem to be things that could be  
resolved with a little research, planning, and a better grasp of  
Asterisk configuration.


for example: In your example above where they can't figure out how to  
transfer, why don't you edit features.conf and define the transfer  
key as # or something. Then, when they have a call for Bill across  
they way, they can do this:


1.) Answer call, determine call is for Bill.
2.) Press #. Asterisk reads back Transfer.
3.) Dial parking extension number (700, for example)
4.) System reads back parking space number (703, for example)
5.) Call or shout to Bill You have a call on 703

This is not really much harder or more complicated than what they are  
used to with their old key system:

1.) Same as above
2.) Press Hold Button
3.) Look at phone to see which line #
4.) Call or shout to Bill You have a call on Line X

This approach also cuts out the press More button, press Transfer  
Button issue you mention below.


Getting users to make that change shouldn't really be that difficult,  
especially if you let the customer know what to expect from the  
beginning. Focus on management and stress the advantages they receive  
as a result of Asterisk being a full-fledged PBX, not a key system.  
Then explain that minor changes in the user interface are the small  
price they must pay for those advantages.


 What I have tried to do is the following scenario.  Assign two  
line keys as Park 720 and Park 721, and using third party patches,  
been able to monitor those lines (which are actually parking spots)  
using hints.  Also, using third party patches, I can transfer to  
those lines (transfer directly to a parking spot), but again, that  
is a several step process (it requires a blind transfer which take  
pressing transfer, then blind on the Polycom, this method, due to  
no BLF does not work on the Ciscos) that just won't happen in small  
businesses.  It just takes too many button presses.  Plus, as I  
mentioned, this is third party patches that aren't in the Asterisk  
main branch, and makes upgrades near 

Re: [asterisk-users] reduce dialtone volume on zap channel.

2006-11-22 Thread Tom Rymes


On Nov 20, 2006, at 2:21 PM, Don Pobanz wrote:


Eric ManxPower Wieling

No, you cannot change the volume of ONLY the dialtone
on a Zap interface.


I was afraid of that.


The most common problem with the first digit being
missed by the telco is that Asterisk is trying to
dial too soon after it goes off hook.  If
it is an analog port then prefixing your number
with w or ww will help.



My issue is prior to ever sending the digits somewhere else such as  
the pstn. It is just having asterisk recognize the dtmf when I  
press a button on the phone.


Don Pobanz


Dan,

If I have followed this thread correctly, your problem is that, when  
you pick up a local analog phone connected to asterisk through a zap  
channel, asterisk generates a dialtone, and everything works fine,  
except that the echo is intolerable. Then, you install an echo  
canceller, and then asterisk cannot reliably register your DTMF  
digits when you pick up the phone and dial. In other words, your  
problem shows up when you install the echo cancellers.


Do you have the echo cans installed between your local extension and  
the zap channel? I assume so, because otherwise they should have no  
effect on your DTMF. Maybe I'm missing something, but I was always of  
the impression that echo cancellers are installed between asterisk  
and the PSTN, not between the local handset and asterisk. That way,  
the echo canceller is only in the media stream when you place a call  
out to the PSTN. I assume that you don't have echo problems calling  
from one local analog extension to another. If you do, however, I  
would suggest that maybe your problem is bigger than just a DTMF issue.


Just a thought,

Tom
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Re: [asterisk-users] names of SIP aware firewalls

2006-11-07 Thread Tom Rymes
I have always been happy with Snapgear units. Most also have hardware based Encryption acceleration for IPSec and PPTP VPNs. As for setting up with SIP, use a VPN, or call their tech support line and they'll help you. Each unit comes with support for setup built-in last I checked.www.snapgear.comTom On Nov 6, 2006, at 6:24 PM, joe a. wrote:An open source firewall:www.ipcop.org lots of add ons, including a fairly new sip proxy:http://mh-lantech.css-hamburg.de/ipcop/e107_plugins/forum/forum_viewtopic.php?1847.5I think well of IPcop and the gent who did the add on, but have no experience with that particular one.joe a.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users  Tom RymesCascade Link Systemswww.cascadelinksystems.com(603) 375-1414"Intelligent technology solutions for small businesses." ___
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Re: [Asterisk-Users] [OT] Centrex Question

2006-04-07 Thread Tom Rymes
I would suggest that you let your fingers do the walking and call  
your local phone company's business sales office and talk to them  
about what your needs are and how centrex will fit. I have found that  
non-technical people (ie: your client) will occasionally mix up very  
important details in situations like this and you might find that  
they were actually talking about five centrex lines, not two. Talk to  
the phone company and let them explain it to you. They should know  
better than anyone


Tom

On Apr 8, 2006, at 12:05 AM, Brian Capouch wrote:


Alexander Lopez wrote:

With strange promos and tariffs, it is possible that Centrex 'lines'
offer a larger Caller area and may in fact be cheaper than  
standard POTS when other

services are added.
For example I need a bunch of POTS lines for our ISP a few, more than
10!, years ago.


What I'm trying to understand is whether their two proposed  
Centrex lines will allow for the five separate extensions that  
they have now with their PBX.


In other words, is there any way a 2x5 PBX (with the customer  
paying for 2x POTS lines) could be swapped out with two Centrex lines?


Thanks, all.  I'm learning a lot here.

B.
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Re: [Asterisk-Users] inbound routing with amp and TDM400

2005-12-21 Thread Tom Rymes

On Dec 20, 2005, at 7:38 AM, Rich Adamson wrote:

Is there a way (via AMP ) to route a call based on  the DID , or  
better

based on the inbound channel number

I had non problems doing that with AMP and with digium PRI Cards,  
but now

for the first time I am trying to use a TDM400 card

It seems that DID is non populated.


There is no such thing as DID in the analog TDM environment.


I also tryed this:


[snip]

Maybe this post I made a while back will be helpful:

http://lists.digium.com/pipermail/asterisk-users/2005-July/114811.html

As Rich said, there is no such thing as DID or DNIS on Analog/TDM lines.

Tom


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Re: [Asterisk-Users] Sip behind the NAT

2005-12-13 Thread Tom Rymes

On Dec 13, 2005, at 8:25 AM, Michael George wrote:


On Fri, Dec 09, 2005 at 03:23:31PM -0500, Tom Rymes wrote:

On 12/8/05, chawki hammoud [EMAIL PROTECTED]  wrote:


Hi:

i added these two lines to my general context ,but
nothing happened the same result the sound came in one
way for 3 seconds and stopped but it didnt hangup.

--- Jeffery Chen [EMAIL PROTECTED] wrote:


If your Astersik server behind NAT too, your need
modify SIP.conf like
this

externalIP= x.x.x.x
localnet= x.x.x.

hope this can help you


Make sure that you have ports 5060 and ports 1-2 UDP
forwarded to your Asterisk server. (Asterisk uses UDP for SIP, not
TCP!!!)

Also, in addition to the externip and localnet entries in sip.conf,
You need to add a nat=yes entry


I have a similar problem with a client's system.  They have * 1.0.x
behind a NAT with all the SIP phones on the local network.  Their VoIP
provider is outside the NAT (a Metaswitch at their ISP, connected  
to the

phone lines from there).

Their network guy has the firewall passing traffic on ports 5060 and
1-2 to the * system.

I have externalIP and localnet set, but nat=no (default) is the case
for this one.

Occasionally they will place outgoing calls and the other party  
does not

hear anything.  Usually another attempt at the call will pass audio
normally.

One person who makes about 100 calls a day remembers having this  
happen

on about 7 calls one day.

No one recalls this ever happening on incoming calls (though this  
client

primarily makes outgoing calls, I believe).

Apparently this has been happening for a while and they just now
mentioned it to me.

Would nat=yes in the general section of sip.conf make a  
difference in

this case?

Is there anything else I could look at that might alleviate this
problem?


Without being a smartass, the only way to find out is to see if it  
works. More obviously, if the Asterisk server has a NAT between it  
and the ITSP, then use nat=yes, if it doesn't, then use nat=no. Of  
course, if you set nat=no, then don't bother setting localnet or  
externip, either.


Also keep in mind that some routers' DMZ settings still leave your  
box behind NAT. They just forward all of the ports to the specified  
address. (Linksys routers do this.)


Tom


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Re: [Asterisk-Users] Sip behind the NAT

2005-12-09 Thread Tom Rymes

On 12/8/05, chawki hammoud [EMAIL PROTECTED]  wrote:


Hi:

i added these two lines to my general context ,but
nothing happened the same result the sound came in one
way for 3 seconds and stopped but it didnt hangup.

--- Jeffery Chen [EMAIL PROTECTED] wrote:

 If your Astersik server behind NAT too, your need
 modify SIP.conf like
 this

 externalIP= x.x.x.x
 localnet= x.x.x.

 hope this can help you


Make sure that you have ports 5060 and ports 1-2 UDP  
forwarded to your Asterisk server. (Asterisk uses UDP for SIP, not  
TCP!!!)


Also, in addition to the externip and localnet entries in sip.conf,  
You need to add a nat=yes entry


Tom


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Re: [Asterisk-Users] Preventing incoming calls from ringing SIP lines

2005-12-06 Thread Tom Rymes

On Dec 5, 2005, at 9:02 PM, James B. MacLean wrote:


Paul Redstone wrote:


Hi

We're using three line SIP phones (X-lite), very nice, with  
Asterisk 1.2


But we want to prevent either direct incoming calls or calls from  
other extensions from ringing if the user is
in another incoming call (i.e incoming into the extension), making  
an outgoing call or even checking their voicemail.



Just a newbie response, but what about the incominglimit= option in  
your /etc/asterisk/sip.conf?


I think incominglimit is deprecated. Doesn't x-Lite have a setting  
that lets you disable call waiting? That's how we handle this problem  
with Cisco 79XX.


Tom


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Re: [Asterisk-Users] Packeteer ? Edgemark ? How to not re-cable ?

2005-12-06 Thread Tom Rymes

On Dec 6, 2005, at 3:11 PM, A_ Navone wrote:


I have customer wtih 30 stations in cubicles but they only
have 1 rj45 per cubicle and that is for lan and internet.
I would prefer the voip to be on separate net connection for  
quality purposes
but customer does not want to recable.  How to avoid voice quality  
problems ?

I have read about devices like Edgemark or Packeteer that
can prioritize voip udp.  Is that true ?  Do they work ?
Thx in Advance


Well, you have a number of choices:

1.) Buy a phone that has a built-in switch (Cisco, Polycom come to  
mind). Unless your LAN is really, really heavily loaded, you  
shouldn't run into any problems whatsoever.


2.) I assume that your customer currently has one RJ-11 phone jack in  
each cube, in addition to the RJ-45 data jack. In that case, don't  
use VOIP phones or ATAs. Use analog phones or even ADSI analog phones  
connected to a T1 channel bank, connected in turn to a T1 card in the  
asterisk box. No QOS problems, no rewiring.


3.) If you don't want to buy phones that don't have a switch built- 
in, buy small netgear 5-port switches and use them instead at each  
workstation. One downside of this approach, however, is that the  
switch is just one more item for which you have to provide a backup  
power source if you want your phones to work when the lights go out.


Those would be my recommendations, in order of preference.

Tom


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Re: [Asterisk-Users] Sangoma Asterisk at home

2005-12-03 Thread Tom Rymes

On Dec 2, 2005, at 12:03 PM, Jess Coburn wrote:


Guys,

I'm curious if it's possible to asterisk at home and the sangoma T1  
cards together. I realize asteriskathome is traditionally used for  
at home, but I'd like to use it in a small office with a T1 and our  
hardware is a Sangoma card. I know all I need to do to get the  
sangoma working is recompile the zaptel but I can't seem to find  
the source, etc on the server after asteriskathome installs.


Jess


Jess,

Log on to [EMAIL PROTECTED] and type help-aah. It will return a list of commands  
you can use to configure your [EMAIL PROTECTED] system. (You did already do this  
and change all of the passwords, right?) One of those commands is  
rebuild_zaptel which will rebuild the zaptel drivers for you.


You will also need to do this every time a new kernel is installed  
when you update the server using yum.


You might find it helpful to go to asteriskathome.sourceforge.net and  
spend some time perusing the handbook.


Tom

PS: If you haven't changed the passwords on your system, do it now!!

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[Asterisk-Users] Sipura SPA-3000 SPA-2002 - Unable to dial *99

2005-11-30 Thread Tom Rymes

Hi folks,

I am having a small problem with a few Sipura units. The settings are  
pretty much factory stock: the unit is set up to not register and the  
IP address for the unit is static and defined in the SIP setup for  
that unit. All other calls are sent and received properly, this is  
the only problem I have. When I dial *99 from the phone connected to  
line 1, I cannot complete a call. Instead of completing the call, I  
still have a dialtone. The only thing I can think of is that the  
units are somehow setup to ignore 9 and still play the dialtone, but  
I haven't seen anything in the interface to specify that.


I have tried specifying various dialplans to make sure that *xx is  
sent to the Asterisk server, but no joy. Another oddity is that  
dialing *98 works just fine, only *99 fails.


If anyone else has run into this and found a solution, I'd love a  
pointer in the right direction.


Tom


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Re: [Asterisk-Users] Sipura SPA-3000 SPA-2002 - Unable to dial *99

2005-11-30 Thread Tom Rymes


On Nov 30, 2005, at 12:39 PM, Luki wrote:


When I dial *99 from the phone connected to line 1,
I cannot complete a call.


Go to the Regional tab in the advanced admin menu, find the Vertical
Service Activation Codes section. Remove which ones you don't want the
Sipura to handle (i.e. *99).


DOH!

Don't know why I didn't try those, because I did see them there.  
What's weird is that if I remove the entry for *99, I can now dial  
*99 and it works. However, there is still a definition for *98 in  
there as well, and that has worked all along.


Weird.

Tom


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Re: [Asterisk-Users] Problem with pulses dialing on asterisk 1.2

2005-11-28 Thread Tom Rymes

On Nov 28, 2005, at 3:00 PM, John Novack wrote:


Cyrille DERORY wrote:

I'm using asteriskathome 2.0 beta 6 (asterisk 1.2) with SIP  
softphones, 7905G cisco SCCP and analog phone( DTMF dialing). All  
is working nice, however when I change DTMF for an analog pulse  
dialing,my analog phone is not working.


I've found the following :

http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
pulse=yes

http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse 
+dialing

pulsedial=yes


Pulsedial=yes works

You should be aware that inserting a w into the dialstring does  
NOT work with pulsedial, however, so if your PSTN connection is a  
little slow, you may get misdials


Remember you need to restart after making the change.
To be really safe, reboot.


I think this might be a problem peculiar to Asterisk @ home, since I  
cannot get my install to accept pulsedial=yes, either. Even though I  
have tried specifying pulsedial=yes in zapata.conf before the  
channel= line and also in zapata-auto.conf, as well as in  
zapata_additional.conf (along with the extension's config), none of  
the above work. (I figured I'd try everywhere, even if it didn't make  
any sense...)


Anyhow, if I connect to the Asterisk console and run zap show  
channel 1, it reports Pulse phone: no even though pulsedial=yes  
is specified. I don't know how [EMAIL PROTECTED]/AMP would have modified zaptel to  
break this, but it is indeed possible that this is an [EMAIL PROTECTED] specific  
problem.


Tom


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[Asterisk-Users] Script to update externip for [EMAIL PROTECTED]/AMP [was Re: SIP Extension behind NAT, Asterisk on NAT (DMZ)]

2005-11-27 Thread Tom Rymes

On Nov 26, 2005, at 3:06 PM, Manny A. Wise wrote:

[snip]


The problem is not updating the FQDN name in dyndns.org..that part is
working great...the problem now is..how to get the IP change into the
sip_nat.conf... but I am sure has to be a way... :)


How about this? You have to add back in the first shebang line that  
defines /bin/sh as the program to use to run the script.


BEGIN SCRIPT-
# Script to update Asterisk's externip= setting with the current
# IP address. Writes changes to sip_nat.conf for [EMAIL PROTECTED]/AMP
# This script is only useful if you have a dynamic IP Address and
# are using NAT.

# define where to write temporary files
Tmp=/tmp/externipupdate$$.txt

# define the hostname to lookup
host=myhost.mydomain.dom

# Use dig to get our current IP address (assuming that it has been
# properly updated via DynDNS client or otherwise. Set the variable
# ip_address to the value of the IP address.

ip_address=`dig $host +short`

# Write the new settings to a temporary file and then overwrite the
# exisiting /etc/asterisk/sip_nat.conf file with the temporary file
# using the mv command.

echo nat=yes  $Tmp
echo externip=$ip_address  $Tmp
# Change the following line to reflect your local network. Add multiple
# localnet= lines if you have more than one local network.
echo localnet=10.0.0.0/255.255.255.0  $Tmp
mv $Tmp /etc/asterisk/sip_nat.conf

# Tell Asterisk to reload SIP to make the changes take effect

/usr/sbin/asterisk -rx sip reload
--END SCRIPT

That ought to do the trick. I tested it with my [EMAIL PROTECTED] config. I will  
leave the task of finding a way to automatically run this program  
when needed as an exercise to the reader. (cron would work if it  
changes at regular time intervals, I suppose)


Keep in mind that there will be a delay between when the address  
changes to when it is updated in DynDNS and then another delay as the  
change propagates throughout the DNS system. Lastly, depending on how  
you call the script, there will be a delay between when it propagates  
through DNS and when this script is run.


Maybe there is a way to use the DynDNS client to get the new IP  
address and write it to sip_nat.conf at the same time it updates the  
DynDNS service? Are there DynDNS clients that allow you to run an  
external program every time the IP changes? Kind of like Comedian  
mail's externnotify parameter?


Tom


Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

Intelligent technology solutions for small businesses.
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Re: [Asterisk-Users] Small office with all employee's offsite

2005-11-27 Thread Tom Rymes

On Nov 26, 2005, at 4:01 PM, Jason Marshall wrote:


I want all calls to come into the Asterisk box in the main office.


This is relatively easy, but how you do it depends on where the  
analog POTS lines are terminated. At the central office or at the  
employees' remote location? (I assume that they terminate at the  
remote locations)


You're right, I should have been clearer.  The way things are now  
is probably suboptimal, but here it is anyway.


We have one phone number, the line for which is terminated in the  
main office, which is where I'd like the server to be.


The two employees, offsite, have seperate lines which terminate in  
either location.


OK, then this is easy. Instal Asterisk in the central location, along  
with a Sipura SPA-3000. Configure that unit to answer the incoming  
POTS line and act as a VOIP gateway for Asterisk. Then configure two  
additional SPA-3000 units, one at each employee's location. Then,  
configure Asterisk (I recommend [EMAIL PROTECTED] for your setup, BTW) to  
route the incoming call to the right extension based on time of day,  
auto-attendant, whatever. The SPA-3000 units at each remote site will  
also be able to accept the employee's incoming POTS line and pass  
that call through to the phone they normally use without resorting to  
sending it to the Asterisk server and back. (It's all in the SPA-3000  
setup.


What we do, depending on who is on at that time, is forward the  
main number (which is hooked up to an old portmaster 2 via a modem,  
so reachable remotely) to whoever should be getting the calls.   
This is suboptimal for at least two reasons that I can see:  1)  
We're paying for a phone line which is basically never used -- the  
call forwarding happens at the telco's switch in the CO, so nothing  
ever comes in over that line;


This will not change, you're still looking at three lines in the  
scenario I outlined above. (Unless you switch to incoming VOIP, but I  
do *NOT*  recommend that.)


2) There's no way to record the calls, or to have a consistent  
voicemail prompt, nor is there any way to present the caller with  
any options if, for instance, the person who has the phone  
forwarded to him is busy, or has gone missing for whatever reason...


Asterisk will indeed solve this problem.

[snip]

If I put one of these at each of the two remote sites, could I set  
them up so that the employees' phones would ring whether the call  
was routed to them via VOIP, OR if I call their current phone  
number?  So if the server dies, or the DSL to the employees'  
locations dies, we could revert back to the lame way we're handling  
call routing now -- by just forwarding the main incoming line to  
one employee's number?


Yes, on both counts.

The downside of using a SPA-3000 at the remote location to answer  
the phone, send the incoming call to the asterisk server, and then  
send it back to the extension at the remote site is that you will  
use double the bandwidth. using SIP reinvites might help with  
that, though.


If I understand you, this scenario would be to intercept calls to  
each employee's current telephone number, redirect the call via  
VOIP into the Asterisk server, and then direct another VOIP call  
back to the employee's handset.  If that's what you mean, that's  
not what I hope to accomplish. No one knows each employee's actual  
telephone number.  It's all hidden with the call-forwarding of the  
main number to each employee's number.


Given that you have the one incoming line at the central location,  
you are good to go. Don't worry about the above.


I should see if my local bookstore has a copy, to save on shipping  
(and delays at the border).  If no one has it, I may very well take  
you up on your offer.  Do you have a paypal seller's account?


Yes! Feel free to make donations as often as you feel necessary... ;-)

Tom


Tom Rymes
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Re: [Asterisk-Users] A rather big setup.

2005-11-27 Thread Tom Rymes

On Nov 27, 2005, at 10:22 PM, Vedran Dakic wrote:

What worries me is the fact that when you have 100-200 offices -  
they're
used to having 2-3 lines only for them - one for fax, two for  
voice, etc.
So, in a way, having in mind around 200-300 outbound calls at peak  
time is
pretty much normal. Also, when you think of the number of phones  
- it
would only be normal to assume for people to have up to 1000  
internal phone

conversations peak (the less transcoding - the better, of course).


If you're providing them with analog lines that they would plug  
faxes, phones, etc into, you should use T1/E1 cards and Analog  
Channel banks. This choice, of course, affects:


I have a freedom of making whatever I want, so I can have a  
separate LAN for

VoIP purposes only - a bunch of dedicated patch panels, VLANs on Cisco
switches, or whatever. I'm just considering this setup way before  
it has to
go online because of the price of traditional PBX for this kind of  
setup
which can only make you hurl. And you know how much potential  
upgrades cost

for a setup like this - a traditional PBX can be a nightmare :(


If you are using analog channel banks instead of ATAs or SIP  
hardphones, then VLANS, etc are not necessary. Of course, you will  
then need wiring for however many lines into each office, but then  
again, that most likely already exists. (thus saving on investment...)


Also, if these tenants are not related, then why not run more than  
one Asterisk server and avoid interconnecting them? Sure, you'll have  
multiple systems to maintain, but they will be smaller, less complex  
systems. Also, since each company is unrelated, there is little  
benefit to having them all on the same server (no need to dial  
between offices, etc)


Tom


Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

Intelligent technology solutions for small businesses.


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Re: [Asterisk-Users] New Asterisk user - Dumb Questions

2005-11-27 Thread Tom Rymes

On Nov 27, 2005, at 8:52 PM, John Novack wrote:


Mike McMullen wrote:


[snip]


I have installed [EMAIL PROTECTED] 2.0 which uses Asterisk 1.2.

Everything appears to be running great with two exceptions:

1) On outbound dialing it appears that about 50% of the time the
   leading dial digit is lost by the phone company. I have 8 analog
   lines connected into two digium TDM400P cards.
   I think Asterisk is dialing the number before the carrier (SBC)
   is ready. Is there a way to create a delay in the dialing to allow
   SBC to be ready to take the dialing? Kind of like the w command
   in modem dialing. If so, what do I need to put and in which  
configuration

   file?


You are absolutely correct.
Asterisk does NOT listen for dialtone, and no one seems able or  
cares to fix that problem


IF, and only IF, you are dialing with DTMF, you can insert a series  
of w into the dial string.
Search the list archives for exactly where, then you will have to  
struggle with [EMAIL PROTECTED] to make the change and keep it from being  
overwritten.
If, on the other hand, one uses pulse dial, then the w  in the  
dialstring will not work.


As someone else mentioned, open AMP, configure your ZAP Trunk and put  
the 'w' in there. Works like a charm.


2) Is there a way to increase the volume of received vmail in  
email? I've
   tried listening to them on 2 different PCs and a laptop. Even  
with volume

   turned up all the way, they are very faint.


Another chronic Asterisk problem.
I am not sure there has been a fix to this either. Seems to only be  
there with the TDM400, or perhaps the TDM400 and the X100 cards..


You might want to consider looking at your ZAP txgain and rxgain  
settings. Be careful, because messing with them and setting them  
improperly can result in echo problems. However, it is possible that  
the ZAP channel rxgain needs to be boosted. You don't notice on  
normal telephone calls because you have your phone's volume control  
turned up to compensate. The Voicemail app, however, isn't able to  
compensate in the same way.


Tom

Tom Rymes
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(603) 375-1414

Intelligent technology solutions for small businesses.


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Re: [Asterisk-Users] Asterisk fax

2005-11-26 Thread Tom Rymes

On Nov 26, 2005, at 8:15 AM, Steve Totaro wrote:


From: Vedran Dakic [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 26, 2005 8:07 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk fax

Is there a way to use a regular (analog) fax machine with Asterisk? I
suppose it coule be achieved by

using some ATA device, but is it possible without that?

Cheers,

Vedran.


No.


More specifically, you can make it work using an ATA or a TDM400P  
card with an fxs port, but it is not likely to be reliable. If you  
send a few faxes here and there, that shouldn't be a big deal. If you  
are talking about an office where lots of faxing is done, the lack of  
reliability will likely be noticeable.


Tom

Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

Intelligent technology solutions for small businesses.


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Re: [Asterisk-Users] NewBie to Ast Server, help need for the configuration

2005-11-26 Thread Tom Rymes
On Nov 26, 2005, at 4:35 AM, ram wrote:Hi   thanks,   but later i posted that, [EMAIL PROTECTED] need new server to start with  as i have shortage of servers to go with that kind of setup so i have downloaded the Ast Source not [EMAIL PROTECTED] and compiled existing server setup, rather a new server so iam asking what are config files need to change yes iam working with that book, and understand things iam asking there any on hand to startup and immediatly setup  so 5 users can start calling out side,  later i go with new server and start wit the fresh setup later   ramram,A few things:1.) People here will be happy to help you out if it is clear that you have already tried to help yourself by using Google, reading voip-info.org, and reading the documentation. However, it seems obvious that you haven't, so you are unlikely to find someone who is willing to do all of the work for you (we've all got our own jobs to do). So before you post here, Google it, google it some more, then go to the wiki. Search the archives of this list, you name it. All of the questions you have asked thus far could easily have been answered if you had simply looked for the answer yourself. If you have already done your homework and you still can't find an answer, *then* post to the list, but don't ask us to figure it out for you.Some google searching tips for Asterisk (type these along with your search terms):site:voip-info.org (This tells Google to search only the wiki.)site:lists.digium.com (This tells Google to search only the list archives)2.) If you want to install [EMAIL PROTECTED] on a server that already has linux installed (like your existing FC4 install), go to asteriskathome.sourceforge.net and read the handbook. There is a tar.gz file you can download that will install over an existing CentOS linux install. It might or might not work with FC4. If you have problems with that, ask a question on the [EMAIL PROTECTED] forum at sourceforge.3.) You don't need to use specialized server hardware to install [EMAIL PROTECTED] for 5 users. Just install [EMAIL PROTECTED] on any Pentium 3 or better desktop, laptop, whatever and use that for your demo. Heck, for 5 users, a Pentium 2 would probably work, considering this isn't a production machine. Assuming that goes well, then buy some more robust hardware for your production install.4.) If you don't have any machines that fit the description in 3.), then scare up an extra hard drive for your desktop machine that you use daily. Then, remove your existing HDD (This is important, b/c [EMAIL PROTECTED] will overwrite it if you leave it installed!). Install the new hard drive and install [EMAIL PROTECTED] on it. configure Asterisk and get it working and do your demo. If you need to fire up your normal desktop install, remove the new HDD and swap back your original HDD. (Not ideal, but it'll work and it will save you a ton of time compared to trying to install and configure Asterisk any other way. When you have a small or even tiny budget, some good old Yankee ingenuity comes in handy...even if you aren't a good old Yankee)Tom Tom RymesCascade Link Systemswww.cascadelinksystems.com(603) 375-1414"Intelligent technology solutions for small businesses." ___
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[Asterisk-Users] Re: SIP Extension behind NAT, Asterisk on NAT (DMZ)

2005-11-26 Thread Tom Rymes

On Nov 26, 2005, at 12:22 PM, Manny A. Wise wrote:


On Nov 25, 2005, at 7:00 PM, Manny A. Wise wrote:


Great!!, this did the trick, now we have audio...
We are using a Sipura 2000 for testing
The Sipura now can call out and have audio...the only problem left
is that
the sipura can't receive calls, when the extension is dialed, the
recording
says, the person is on the phone.any ideas???

I changed the externip=, localnet= and nat=yes in sip.com and in the
extension setup in amp nat=1.. missing anything

THANKS

Manny


-Original Message-
From: Tom Rymes [mailto:[EMAIL PROTECTED]
Sent: Friday, November 25, 2005 10:48 PM
To: Manny A.Wise
Subject: Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk  
on a public

domain

It sounds as if your extension isn't registered. Make sure that the
extension is configured as dynamic in sip.conf (or AMP) and as
nat=yes. Also, make sure that the Sipura is configured through its
web interface to register and it has the right user and password
entered. Once this is done, when you type 'sip show peers' from the
CLI your Sipura's extension should be listed, and show a 'D' and an
'N' for dynamic and nat.

Also, it sounds like you are using AMp and or [EMAIL PROTECTED], so make sure 
that
you put the nat, externip, and localnet parameters in the
sip_nat.conf file, *NOT* the sip.conf file, as that is likely to get
overwritten by AMP. From my installation (obviously, substitute your
external IP for the xxx.xxx.xxx.xxx below...):

[EMAIL PROTECTED] root]# cat /etc/asterisk/sip_nat.conf
nat=yes
externip=xxx.xxx.xxx.xxx
localnet=10.0.0.0/255.255.255.0

Other than that, I recommend further google and voip-info spelunking
expeditions to track down your problem.


Tom
We found the trouble...
What is happening is that my friend connection is changing IP  
address way to
often.he is on a DSL line in Peru...and is chaging when we less  
expect

it, now that we know, we are monitoring it
Anyway...the problem left is that if we use the IP, it work  
perfectly both

ways ...but if we use his name.dyndns.org name audio doesn't work...
I though that externip=201.240.220.123  or  externip=name.dyndns.org
Should do the same work.?
The dyndns.org support is build on his Ztxel modem, and so far I  
never had a

single problem accessing the site by webrowser...

Any thought??


Manny,

Two things:

1.) please send your replies to the list and not to me personally!  
That way the discussion ends up in the list archives and will be  
there to help the next person who has this problem.


2.) I am not certain if the externip= option can take a  FQDN as an  
argument. The page I looked at on the wiki (see below) was not clear  
on this. I suggest using trial and error to see if works and then  
update the wiki page once you know.


3.) If your external IP is constantly changing, I searched for  
site:voip-info.org externip on Google and found this page on the wiki:


http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+externip

which has a link to this message in the list archives:

http://lists.digium.com/pipermail/asterisk-users/2004-November/ 
071349.html


which discusses this problem. Basically, every time that your IP  
address changes you need to update the IP with dyndns, and you need  
to change the externip= option. Anyhow, if your address changes and  
you do not update the dyndns service, the SIP client will be looking  
for the wrong IP (the old IP.).


HTH,

Tom


Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

Intelligent technology solutions for small businesses.


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Re: [Asterisk-Users] Asterisk fax

2005-11-26 Thread Tom Rymes

On Nov 26, 2005, at 12:47 PM, Andrew Nowrot wrote:


More specifically, you can make it work using an ATA or a TDM400P
card with an fxs port, but it is not likely to be reliable. If you
send a few faxes here and there, that shouldn't be a big deal. If you
are talking about an office where lots of faxing is done, the lack of
reliability will likely be noticeable.


Hi

I don't see what seems to be a problem. I use the fax machine with
diguim 3 fxs 1fxo card and everything works fine. How this lack of
reliability can reveal itself?
I also can't say that I use my fax very often (maybe 20 times a day).

Cheers


Well, shocking as it might be, I could be wrong. I might be wrong  
about that, though... smirk


Seriously, I can't say for sure. In the past there have been some  
posts on the list suggesting that the TDM400P didn't play nicely with  
fax traffic. It could be that this problem has since been resolved,  
or maybe heavier fax traffic would reveal a problem, I dunno.


However, I do think it is fairly clear that using an ATA is a less  
than ideal solution for any serious faxing, since the fax protocol  
often doesn't play nicely with the tendency of VOIP to occasionally  
lose packets. YMMV, though, so try it out and see how it works for you.


Tom


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Re: [Asterisk-Users] Small office with all employee's offsite

2005-11-26 Thread Tom Rymes

On Nov 26, 2005, at 12:48 PM, Jason Marshall wrote:

I'm sure these questions have been answered at some point, but I'm  
too new to this stuff to know the right words to plug into the  
search function to find what I need.


We'll let it go just this once... ;-)

I have never touched Asterisk before, but have wanted to for some  
time. Now I finally think I'm going to bite the bullet, as I have a  
real-world application for it


Awesome! Welcome to the community.

My office consists of two employees, neither of whom work in the  
office physically.  Here is what I'd like to do.  Hopefully someone  
can tell me what I need to do/buy/configure/install to make it work...


OK, all of this is possible, and as someone else mentioned, the  
easiest, it just works way to accomplish this is through  
[EMAIL PROTECTED], which you can find at http:// 
asteriskathome.sourceforge.net. How you accomplish it will depend on  
a few variables, though.



I want all calls to come into the Asterisk box in the main office.


This is relatively easy, but how you do it depends on where the  
analog POTS lines are terminated. At the central office or at the  
employees' remote location? (I assume that they terminate at the  
remote locations)


I want all incoming calls to be recorded (not as concerned about  
outgoing calls).


[EMAIL PROTECTED] can handle this. It's in the extension setup.

Both employees have regular POTS telephone lines (one fellow has a  
land line and a cell, the other has just a land-line).


Again, it will be important to know where these lines terminate.

I'd like callers to be presented with a short menu of options, the  
behavior of which might change depending on the time of day (for  
instance, at night, I'd like both the sales and support calls  
to go to one employee, while during the day I'd like sales to go to  
one person, and support to go to another.  I'd also like to have an  
answering machine (built into Asterisk?) pick up calls that go  
unanswered.


IVR Auto-Attendants are built into [EMAIL PROTECTED]/AMP. They are called Digital  
receptionists, IIRC. Voicemail is also built-in.


I guess that's about it.  I looked at the Digium TDMxx cards, but  
don't really know what I need in the way of FXO's and FXS's to pull  
off what I want to do.


This is why it's important to know where the phone lines terminate.  
If they are in the office you can use a TDM400P with two FXO ports.  
You can also use an ATA such as the Sipura SPA-3000 that has an FXO  
port built-in. If the lines terminate at the remote locations, then  
the second option is your only one, unless you put a server in both  
locations. (which is a bit overkill...)


The downside of using a SPA-3000 at the remote location to answer the  
phone, send the incoming call to the asterisk server, and then send  
it back to the extension at the remote site is that you will use  
double the bandwidth. using SIP reinvites might help with that, though.


As an added bonus, if someone knows of a VOIP adapter that allows  
one to plug an analog phone into it AND accept both VOIP and normal  
phone calls to the same phone, that would be cool (and might make  
things easier to configure, without making each extension 100%  
dependent on VOIP).


The SPA-3000 is capable of doing this. configuring one the first time  
can be a bit of a bear, but Google is your friend...


Thanks in advance.  I'm really looking forward to finally doing  
something with Asterisk, one of the most exciting projects I've  
looked at for a while!!


Well, good luck and, incase you haven't gathered, google,  
lists.digium.com, asteriskdocs.org, and voip-info.org are your best  
online resources for help. Also, the new book Asterisk: The Future  
of Telephony is a great resource. It's available online as a  
download under the creative commons license, and it is also published  
by O'Reilly http://www.oreilly.com/catalog/asterisk/index.html


Come to think of it, I have an extra O'Reilly official version of the  
book that I will sell for $30 shipped. (Never used, I already have  
another copy...)


Tom


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www.cascadelinksystems.com
(603) 375-1414

Intelligent technology solutions for small businesses.


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Re: [Asterisk-Users] Command line

2005-11-25 Thread Tom Rymes

On Nov 25, 2005, at 7:33 AM, Tony Spencer wrote:

Hi

I’m pretty new to using Asterisk and have searched to find an  
answer to my question but have failed to.


I was wondering if you can use Asterisk from the command line to  
make it make an outgoing call and issue other commands whilst it’s  
in the call?


Sort of like when you use Minicom with a modem connected to a  
serial port and send it AT commands.


 Thanks

Tony


Tony,

If you have a sound card installed and properly configured in your  
Asterisk server, then you can plug in a microphone and headset and  
make calls from the CLI using the dial command.


If you want to automate having the system make phone calls, google  
and search voip-info.org for info on .call files. Basically, you  
create a file that specifies to asterisk where to call, using which  
channel, and what to do once the call is connnected. You then copy  
the file to /var/spool/asterisk/outgoing and the call is executed as  
defined.


Tom


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(603) 375-1414

Intelligent technology solutions for small businesses.


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Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

2005-11-25 Thread Tom Rymes

On Nov 25, 2005, at 7:00 PM, Manny A. Wise wrote:


-Original Message-
From: Tom Rymes [mailto:[EMAIL PROTECTED]

[snip]

On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote:
[snip]

Well, as the user stated on the original message, the asterisk
server is behind a NAT and the client is also behind a NAT..

if you make it work just by opening ports, let me know..I have
never been able to get it to work, that's why I don't use sip, just
plain iax2 for everything. J

Manny


Manny,

I have this working as I write this. (I just hung up the phone.) In
fact, I brought a Cisco 7940G to a completely unknown nat-ed network
the other day, plugged it in and started making calls right away.
Here's the setup I have for this specific configuration:

1.) Asterisk server behind NAT. (Setup as DMZ on Linksys WRT54G, but
it's still NAT. I just don't have to forward ports this way)
2.) externip, localnet, nat settings configured in the sip.conf file
(sip_nat.conf for [EMAIL PROTECTED])
3.) Cisco phone (or whatever SIP UA you choose) configured for NAT
(via the SIPMAC.cnf file for Cisco)
4.) Lather, rinse, repeat if necessary

Hopefully that will work for you. I'd rather use IAX and avoid these
problems altogether, but I have yet to find an IAX hardphone I am
willing to use. In fact, for softphone use, I do indeed use IAX via
LoudHush for the mac. (Great piece of software, BTW. No connection
here, just a happy user...)

Tom


Great!!, this did the trick, now we have audio...
We are using a Sipura 2000 for testing
The Sipura now can call out and have audio...the only problem left  
is that
the sipura can't receive calls, when the extension is dialed, the  
recording

says, the person is on the phone.any ideas???

I changed the externip=, localnet= and nat=yes in sip.com and in the
extension setup in amp nat=1.. missing anything

THANKS

Manny


It sounds as if your extension isn't registered. Make sure that the  
extension is configured as dynamic in sip.conf (or AMP) and as  
nat=yes. Also, make sure that the Sipura is configured through its  
web interface to register and it has the right user and password  
entered. Once this is done, when you type 'sip show peers' from the  
CLI your Sipura's extension should be listed, and show a 'D' and an  
'N' for dynamic and nat.


Also, it sounds like you are using AMP and or [EMAIL PROTECTED], so make sure that  
you put the nat, externip, and localnet parameters in the  
sip_nat.conf file, *NOT* the sip.conf file, as that is likely to get  
overwritten by AMP. From my installation (obviously, substitute your  
external IP for the xxx.xxx.xxx.xxx below...):


[EMAIL PROTECTED] root]# cat /etc/asterisk/sip_nat.conf
nat=yes
externip=xxx.xxx.xxx.xxx
localnet=10.0.0.0/255.255.255.0

Other than that, I recommend further google and voip-info spelunking  
expeditions to track down your problem. I think that voxilla.com also  
has good resources on the Sipuras


Tom


Tom Rymes
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(603) 375-1414

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Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

2005-11-24 Thread Tom Rymes

On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote:

[snip]
Well, as the user stated on the original message, the asterisk  
server is behind a NAT and the client is also behind a NAT….


if you make it work just by opening ports, let me know..I have  
never been able to get it to work, that’s why I don’t use sip, just  
plain iax2 for everything… J


Manny


Manny,

I have this working as I write this. (I just hung up the phone.) In  
fact, I brought a Cisco 7940G to a completely unknown nat-ed network  
the other day, plugged it in and started making calls right away.  
Here's the setup I have for this specific configuration:


1.) Asterisk server behind NAT. (Setup as DMZ on Linksys WRT54G, but  
it's still NAT. I just don't have to forward ports this way)
2.) externip, localnet, nat settings configured in the sip.conf file  
(sip_nat.conf for [EMAIL PROTECTED])
3.) Cisco phone (or whatever SIP UA you choose) configured for NAT  
(via the SIPMAC.cnf file for Cisco)

4.) Lather, rinse, repeat if necessary

Hopefully that will work for you. I'd rather use IAX and avoid these  
problems altogether, but I have yet to find an IAX hardphone I am  
willing to use. In fact, for softphone use, I do indeed use IAX via  
LoudHush for the mac. (Great piece of software, BTW. No connection  
here, just a happy user...)


Tom


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(603) 375-1414

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Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

2005-11-24 Thread Tom Rymes

On Nov 24, 2005, at 12:14 PM, Bharath wrote:

I found out that I have a faulty Belkin Router which was causing  
the problem. I tried forwarding ports as well as DMZ'd the Sip  
device but still could'nt not hear the voice. So i plugged the sip  
device directly to the cable modem  it worked fine. So my guess is  
that though I have set up the router to forwards port to the sip  
device there is something happening at the router that is blocking  
the RTP ports (1-2).

Thanks


Before you blame the router, make sure that you forwarded UDP ports  
5060 and 1-2, not TCP. (Though I guess the DMZ setup would  
have taken care of that...)


Tom


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Re: [Asterisk-Users] simple setup

2005-11-19 Thread Tom Rymes


On Nov 19, 2005, at 2:50 PM, Gregory Seidman wrote:


I'm trying to set up a very simple system. I have:

1) a LinkSys BEFSR41 NAT router

2) a machine (VIA EPIA running Debian GNU/Linux 3.1, a.k.a. Sarge)  
running
   Asterisk (1.0.7, because that's what's in Debian stable and I  
don't want

   to mess with anything beyond that until I get things working)

3) an account with BroadVoice

4) an IAXy ready to be attached to an analog phone and my LAN

I need to know:

1) what ports to forward on the router to the Asterisk machine


For SIP: 5060  1-2
For IAX: 4569
(This assumes that you haven't changed the defaults in sip.conf/ 
iax.conf)



2) how to configure the IAXy (it came with no documentation)


As with almost everything Asterisk, the documentation can be found at  
voip-info.org, asteriskdocs.org, google, etc. ;-)



3) and how to configure the IAXy extension for Asterisk


See above.

I have already configured Asterisk according to BroadVoice's  
instructions
for the most part, but I wasn't sure what to do about the extension  
so my

register line reads:

register = 10-digit phone number:10-character  
passcode@sip.broadvoice.com


I may be wrong here, but if you leave out the extension, the calls  
will be directed to the s extension in whichever context you specify  
for incoming calls. Can anyone verify that?


[snip]

P.S. Please don't tell me about how much better 1.2.x is than 1.0.x  
and how

 I should upgrade. I'll get there, but it is not my first step. My
 first step is is getting a working line to make my wife happy.


No worries, use whatever you like. However, if you are trying to get  
the easiest, quickest startup for Asterisk, consider [EMAIL PROTECTED]  
It's an .iso that, when you boot from the CD, wipes your HDD (SO BE  
CAREFUL) and installs linux, Asterisk, AMP, etc.


You won't get your hands as dirty, and maybe not learn as much right  
out of the chute, but it will be much closer to Just works than  
installing Asterisk alone. (And your wife is likely to appreciate  
this...)


Tom

Tom Rymes
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(603) 375-1414

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Re: [Asterisk-Users] IAXmodem

2005-11-18 Thread Tom Rymes
On Nov 18, 2005, at 11:08 PM, Anton Krall wrote: I know you can send faxes using a hylafax client for windows and sending thru hylafax and iaxmodem out from asterisk but I was wondering, how do you receive faxes? can you received them as tiff and then convert to pdf and send via email or something?Go to google.com and type in "HylaFAX". The first link displayed is the hylafax.org website. Go there and click on "How-To" in the sidebar. There you will find "4. Received Fax Delivery and Viewing".HTH,TomPS: Yes, it is extremely easy to distribute received faxes as PDF files via email by configuring the FaxDispatch file in HylaFAX. You can even distribute to different e-mail addresses based on CallerID, Modem, etc.Tom RymesCascade Link Systemswww.cascadelinksystems.com(603) 375-1414"Intelligent technology solutions for small businesses." ___
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[Asterisk-Users] Help with shell script for externnotify

2005-11-17 Thread Tom Rymes

Hi folks,

I am working on a shell script that I can use with the externnotify  
command in voicemail.conf. All is well and seems ready to rock,  
except I can't figure out how to tell the script what e-mail address  
to send the mail messages to. I warn you ahead of time that I am no  
scripting guru.


Basically, I have 14 after-hours mailboxes that all have different e- 
mail addresses. I want this script to parse the mailbox number from  
the original command ($2), and then somehow look that up mailbox's  
address and send an e-mail. It then checks every five minutes to see  
if the message has been retrieved, and escalates things as necessary.  
I don't mind the messy solution of defining all 14 addresses in the  
script itself, though it would be nice to look it up from  
voicemail.conf or something eventually.


I started out using /bin/sh for the scripting, but I assume that this  
is limiting me with what I can do with variables, etc. The only way I  
can think of off the top of my head is to use some sort of nested  
variable, like this subset of my script:


Tmpmail=/tmp/dispatch$$.mail
[EMAIL PROTECTED]

# Send an e-mail to the appropriate address for this mailbox:
echo 724-6066 $2  $Tmpmail
echo New Voicemail Message #$3 in mailbox $2 $Tmpmail
cat $Tmpmail | mail -s $mailsubject $ext$2

now, the $ext$2 is what I mean by nested variable. Basically, if I  
can find a way to make this evaluate as $ext108, and then as  
[EMAIL PROTECTED], I would be a happy camper. Is there any way to do this?


I know that this is technically not an Asterisk question, but then  
again, it is. If anyone has wrestled with this in their own  
externnotify scripts, please let me know


Tom

PS: I know that perl and other scripting languages would be better  
for this, but this seemed simpler to me at the time I started. If it  
can't be done using sh, then I'll start over in perl, but if there's  
a way to make this work, it's the only thing standing between me and  
a script that works exactly as I want.



Tom Rymes
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(603) 375-1414

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Re: [Asterisk-Users] Asterisk and Agents

2005-11-15 Thread Tom Rymes

On Nov 15, 2005, at 7:38 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:


Hallo all,

I've a question regarding the agent concept of asterisk. If I login  
an agent

(using AgentLogin), this agent is directly ready to receive calls.

From the most other ACD-systems I know that an agent first logs  
into the

system and then has to set himself ready to receive calls. So the most
common agent states are login, ready, not ready, wrapup and logoff.

How is ready/notready/wrapup implemented in asterisk?


Well, Asterisk 1.2 implements agent pause, so paused/unpaused would  
correspond to not ready/ready.


Wrapup is handled as a fixed time in the the queue setup for  
Asterisk. You define wrapuptime=xx as a number of seconds. Asterisk  
then waits that long before considering the agent to be available.


Tom


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Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for DigiumBoards

2005-11-15 Thread Tom Rymes

On Nov 15, 2005, at 10:02 AM, Kevin Hanson wrote:


Kevin Hanson wrote:

If what you say is true, then I'm hosed.  I've got six things  
sharing IRQ 255 according to lspci -vb:


Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI  
Controller #1
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI  
Controller #2
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI  
Controller #3

Intel Corporation 82801DB/DBM (ICH4/ICH4-M) USB2 EHCI Controller
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) SMBus  
Controller

Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface

Is this what is causing all my echo?

I'll try disabling USB in bios and see what happens.

Cheers,
Kevin


Well, my bios doesn't let me disable usb.  Drat.

Cheers,
Kevin


Kevin,

Have you tried swapping the PCI slot to see if that helps? Does your  
BIOS allow you to reassign IRQ numbers?


Tom


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Re: [Asterisk-Users] Asterisk hobby box

2005-11-15 Thread Tom Rymes


On Nov 15, 2005, at 2:30 AM, Dmitry Ivanov wrote:


On Tuesday 15 November 2005 06:26, [EMAIL PROTECTED] wrote:

Hi. I'm setting up an Asterisk hobby box for me to play around with.
Is it possible to use a regular 56k modem and a regular home phone
for it?


Yes, but forget G.711.

BTW, some SIP-phones have built-in modem :)


Unless I'm mistaken, this is not true. Some modems will work, but  
they are extremely rare. Your average USR/Rockwell/etc. modem will  
not work. Search on voip-info.org for X100P clone and read up on  
which will work. A regular phone line will work just fine, though,  
assuming that you get a way to interface it with your system:


Digium X100P
Digium TDM400P w/FXO Port
Digium TDM2400P w/FXO Port
ATA with FXO Port (Like Sipura SPA-3000)

Tom


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Re: [Asterisk-Users] Asterisk hobby box

2005-11-15 Thread Tom Rymes

On Nov 15, 2005, at 6:16 PM, Logan wrote:

[snip]

I am trying to set up a small hobby box on Debian Linux to play  
around with. This will in no way be in a production evironment or  
even a semi-production environment. Asterisk will be installed on  
my personal Linux box. I have a generic Soft56k modem (which I hope  
to soon replace) in there now. It doesn't work for PPP in Debian  
because it's a Winmodem. - That's irrelavent. I was wondering if  
it was feasable to istall Asterisk on this box and have that modem  
(or whatever modem) with a regular telephone wired to the Phone  
port. I'm hoping to spend very little (under $50) or none, if  
possible.


Logan,

You need to figure out what type of modem you have now. If, by  
chance, it happens to be a Digium X100P clone (very unlikely), then  
it will work. However, if it isn't an X100P clone, it will not work.  
Period. (Unless you code up a driver yourself).


That being said, and as I mentioned earlier, your cheapest choice is  
to go to eBay and search for X100P. However, IMNSHO, your best choice  
is to shell out somewhere around $100 for a Sipura SPA-3000. This  
will provide a way for you to connect your home phone line to  
asterisk, and a way for you to connect an analog phone to asterisk as  
an extension.


It would be a good idea for you to spend some time on google, voip- 
info.org, asterisk.org, asteriskdocs.org, etc. searching for  
information.


Tom


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Re: [Asterisk-Users] Asterisk hobby box

2005-11-15 Thread Tom Rymes

On Nov 15, 2005, at 9:29 PM, Logan wrote:


Philip Edelbrock wrote:


On Nov 15, 2005, at 5:40 PM, Logan wrote:

As stupid as this may seem ::cough::, how do you test to see if   
there is voltage on the phone port? Would you plug in a phone  
that  doesn't require a AC power and runs off the voltage from  
the phone  line?


Thanks though... ^_^
Logan.


I was thinking volt meter, but actually I like your idea of just   
jacking in a phone and seeing if push-button tones and stuff  
work,  with and without the line-side being plugged in.


Phil


It worked, thank goodness. But does that mean it'll work... I just  
looked on the Asterisk-Users message list and for some reason Tom  
Rymes messages.


In response to Tom: I'm sure it's not an Digium-anything. It's a  
cheapo Office Depot replacement for when my original was struck by  
lightning. As I had said, I'm really not wanting to spend any  
money. It's not important that I get Asterisk up and running. Allow  
me to repeat, it's only a hobby box. But you're welcome to give me  
one... ;)


Logan,

Two things:

1.) Check the make, model, chipset, etc of your modem and figure out  
if it is an X100P clone. If it is, you can use it to hook up your  
incoming POTS line to your asterisk server.
2.) If you want to use the phone port of your modem to connect an  
analog phone to Asterisk as an extension, then it will not work.


Tom


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Re: [Asterisk-Users] Anybody tried it from India ?.

2005-11-14 Thread Tom Rymes


On Nov 14, 2005, at 2:50 AM, Dinesh wrote:

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

Sent: Monday, November 14, 2005 1:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Anybody tried it from India ?.

On Nov 14, 2005, at 12:37 AM, ram wrote:


Hi

its not legal in india
connecting to PSTN to VOIP

ram


Asterisk doesn't necessarily mean VOIP. He could set it up using ZAP
channels only and not have any VOIP in use at all.

Tom


Its illegal to interconnect it to the local pstn (from abroad).

Dinesh.


I still don't see how this would stop him from using no VOIP  
protocols and plugging it in to the PSTN. Just use Asterisk as a PBX,  
no VOIP, no bypassing the Indian telephone monopoly (assuming that  
there is one).


Tom

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Re: [Asterisk-Users] newbie question regarding asterisk

2005-11-14 Thread Tom Rymes

On Nov 14, 2005, at 6:21 AM, Markos Paraskevopulos wrote:

Hello everyone,

I’m new to VoIP and despite a lot of reading, I’m kind of more  
confused than before.


I have following question – we currently have hardware Alcatel PBX  
and approx. 50 phones in the company. I was wondering if we would  
need to change the phone service provider, because they don’t  
provide VoIP services if we were about to switch to Asterisk  
instead of the Alcatel PBX?


Or can Asterisk maintain current functionality plus adding VoIP by  
simply switching the alcatel pbx for Asterisk server?


I hope I’m making at least a bit of sense.

 Thanks in advance for help

Confused

Markos

Markos,

The answer to your question is Maybe. It depends on how you connect  
your existing PBX to the PSTN, and it depends on what you want from  
your system.


Asterisk is completely capable of connecting to standard analog and  
digital (T1/E1/PRI) phone circuits. You do not need to use VOIP to  
connect Asterisk to the phone network. However, how you will go about  
doing this depends on your call volume and budget. How many incoming/ 
outgoing phone lines you have, how much long distance you dial, and  
local telco rates all play a part here.


The easiest way is to figure out how you connect the existing PBX,  
and then you can research to see if Asterisk will support that  
technology. (Chances are that it does). For example, if your Alacatel  
connects to the PSTN via a T1/E1 Circuit, then you could buy an T1/E1  
interface card from Digium or Sangoma and plug the T1/E1 right into  
your Asterisk server. If you have multiple analog POTS lines, then  
it's more complicated, but there are solutions for that, too (digium  
X100P, TDM400p, TDM2400p, various SIP gateways, multiple Sipura  
SPA-3000, etc...)


Then you might want to research your other options and make sure that  
you are using the most cost effective solution for your needs (This  
all depends on how you use the PSTN and what the local rates and  
availability are). The most basic knowledge you will need is the  
difference between a T1/E1 style connection and a regular analog POTS  
line. For example, if you have multiple analog lines, you might be  
able to save money by getting a full or fractional T1/E1.


If you're still completely confused and you don't have a lot of  
telecom knowledge, you might want to consider hiring a consultant to  
help you out.


Tom

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