Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Gonzalo Servat]

On Fri, Nov 21, 2008 at 2:49 PM, Noah Miller <[EMAIL PROTECTED]>wrote:

> > And FreeSWITCH can't handle that?
>
> Freeswitch can provide many PBX features with additional modules, but
> asterisk can provide more, and its implementations of such items are
> more time tested.  One of freeswitch's big strengths is its ability to
> handle many SIP registrations.  This is not asterisk's strength (at
> least not historically).  One of Asterisk's big strengths is its
> multitude of services and features.  This is not freeswitch's
> strength.  Combine freeswitch and asterisk to get the best of both
> worlds.
>

I was preparing a reply that would argue the need to have Asterisk involved
if FreeSWITCH is there, but given the name of the list and the potential to
piss people off, I'll leave it at that.

- Gonzalo
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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Gonzalo Servat]

On Fri, Nov 21, 2008 at 1:48 PM, Alex Balashov <[EMAIL PROTECTED]>wrote:

> Gonzalo Servat wrote:
> > On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller <[EMAIL PROTECTED]
> > <mailto:[EMAIL PROTECTED]>> wrote:
> >
> > [..snip..]
> >
> > With that many extensions, I'll second using a SIP registrar like
> > Freeswitch or OpenSer.  Just use asterisk to provide the services.
> >
> >
> > Is Asterisk even needed?
>
> Potentially, no.  But if you intend to provide subscriber/PBX features,
> it is needed as a UA feature box(s).
>

And FreeSWITCH can't handle that?

- Gonzalo
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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Gonzalo Servat]

On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller <[EMAIL PROTECTED]>wrote:

> [..snip..]

With that many extensions, I'll second using a SIP registrar like
> Freeswitch or OpenSer.  Just use asterisk to provide the services.
>

Is Asterisk even needed?

- Gonzalo
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Re: [asterisk-users] Spam from DIDForSale <[EMAIL PROTECTED]>

[Gonzalo Servat]

On Thu, Nov 6, 2008 at 2:11 PM, David Gibbons <[EMAIL PROTECTED]>wrote:

> I'm glad I'm not the only one who got that. I sent them a nasty response
> earlier this morning...
>

I got the same crap from them. I can't imagine anyone buying from a company
that spams subscribers of a mailing list to get business. But I guess spam
exists for a reason; there's always some moron out there that purchases
goods from these delinquents.

- Gonzalo
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Re: [asterisk-users] FWD $30 membership-fee

[Gonzalo Servat]

On Thu, Aug 7, 2008 at 2:04 PM, Joseph <[EMAIL PROTECTED]> wrote:

> I just received an email notice from FWD about $30 membership fee.
> My question: Is the email genuine? Did anybody else receive it?
>
> I'm just trying to be sure that it is real and not a scam.
> The (FWD) does not do anything to authenticate such emails (implementing
> GPG/PGP signature etc.)
>
> If the email is genuine, I hope they will improve their service; as of now
> their IAX server is not working.
>

I just went to www.freeworlddialup.com and the top banner says something
about Paid Membership which links to:

http://www.acteva.com/booking.cfm?bevaid=138192

I'm not sure whether they will discontinue offering the "free" accounts. I
sure hope not, they would have to consider a name change too. I don't really
use FWD but I'm sure a lot of users would be affected.

- Gonzalo
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[asterisk-users] md5secret for IAX?

[Gonzalo Servat]

Hi All,

In an effort to avoid storing cleartext passwords for users, I was looking
around to see if iax.conf had a similar setting to sip.conf's 'md5secret'.
It looks like it doesn't. I've set auth=md5 which, by the look of it, makes
Asterisk work out the md5 on the fly of the cleartext password and compares
it to whatever the client sends (in md5). Unfortunately this means storing
the password in cleartext.

Is it possible or am I forced to store the password in cleartext?

Thanks
- Gonzalo
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Re: [asterisk-users] zaptel alarm

[Gonzalo Servat]

On Wed, Apr 2, 2008 at 10:28 AM, Greg Woods <[EMAIL PROTECTED]> wrote:

> I've been a happy user of asterisk for over a year just for a small home
> setup (a Digium TDM400P with one POTS line and three internal extensions
> plus a couple of SIP phones). I recently moved from running Fedora Core
> 6 running * 1.4.1 compiled from source and zaptel 1.4.7 to Fedora 8,
> using the RPM package for * 1.4.18 and zaptel 1.4.9 . It worked fine for
> a week, but suddenly I can't dial out any more. Incoming calls work
> fine, and outgoing calls through my Teliax line (IAX2) work fine, but I
> get a fast busy if I try to dial out through the land line. All that
> appears in the messages log is:
>
> [Apr  2 07:13:48] WARNING[24249] chan_zap.c: Detected alarm on channel
> 4: No Alarm
> [Apr  2 07:13:48] NOTICE[24242] chan_zap.c: Alarm cleared on channel 4
>
> "core set debug 3" doesn't give any more detail.
>
> I have tried stopping asterisk, restarting zaptel (unloading and
> reloading), and starting asterisk, but the behavior persists.
>

Hi Greg,

This might be a long shot, but I had this exact same problem recently and it
turned out to be a problem with the actual line. Have you tried plugging in
a phone directly to the line? Check that you have a dial tone and that you
can receive and make calls.

- Gonzalo
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Re: [asterisk-users] Asterisk not picking up (some) calls due to zaptel detecting and clearing alarms

[Gonzalo Servat]

On Thu, Mar 27, 2008 at 1:56 PM, Tzafrir Cohen <[EMAIL PROTECTED]>
wrote:

> > Any suggestions??
> >
> > I'm using Asterisk 1.6.0-beta4 and Zaptel 1.4.9.2.
>
> A freshly-built Asterisk? Built vs. zaptel 1.4.9.2 ?
>

Yes, I built 1.6.0-beta4 just recently with zaptel 1.4.9.2. As per your
suggestion on IRC, I've checked out, compiled and installed Zaptel from SVN
(1.4 branch). I reloaded the zaptel modules but ... no go. Do I need to
recompile Asterisk too?

Shouldn't it have picked up the alarm as a red alarm on the channel?


I've no idea to be honest.


> (Besides the problem. Is 1.4 SVN recommended for that at the moment?)
>

Also no idea. I was told to use 1.4 if I'm using Asterisk 1.6 so I went with
that.

- Gonzalo
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[asterisk-users] Asterisk not picking up (some) calls due to zaptel detecting and clearing alarms

[Gonzalo Servat]

Hi All,

For the most part, the PBX works as it should. Occasionally people complain
that they call and the PBX doesn't pick up. Other times it looks like the
call is answered by Asterisk but I still hear ringing and I start listening
to the IVR menu a few seconds into it.

As for Asterisk not picking up, I see the following in the logs:

[Mar 27 13:32:29] NOTICE[13197]: chan_zap.c:7071 ss_thread: Got event 18
(Ring Begin)...
[Mar 27 13:32:30] NOTICE[13197]: chan_zap.c:7071 ss_thread: Got event 2
(Ring/Answered)...
-- Executing [EMAIL PROTECTED]:1] Answer("Zap/2-1", "") in new stack
-- Executing [EMAIL PROTECTED]:2] Wait("Zap/2-1", "1") in new stack
[Mar 27 13:32:30] WARNING[13197]: chan_zap.c:4203 get_alarms: Unable to
determine alarm on channel 2
[Mar 27 13:32:30] WARNING[13197]: chan_zap.c:4424 zt_handle_event: Detected
alarm on channel 2: No Alarm
  == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
[Mar 27 13:32:31] NOTICE[13197]: chan_zap.c:7514 handle_init_event: Alarm
cleared on channel 2

The above messages repeat themselves a number of times as the ringing
continues and causes Asterisk to try and pick up the call again (and fails
with the alarm thrown which then gets cleared).

Any suggestions??

I'm using Asterisk 1.6.0-beta4 and Zaptel 1.4.9.2.

Thanks in advance!
Gonzalo
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Re: [asterisk-users] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min

[Gonzalo Servat]

I think this type of abuse is well deserved due to the way he intended to
advertise his "business", so I'll add a bit of wood to the fire. How about
the sign-up form?? Some serious HTML design work going on there.

- Gonzalo

On Fri, Mar 21, 2008 at 1:15 PM, Tim Nelson <[EMAIL PROTECTED]> wrote:

> The template website, page titles, and Gmail contact address surely aren't
> very convincing. Another crappy VoIP reseller that will fail in a few months
> taking a handful of customers down... assuming they're legit to begin with.
>
> --Tim
>
>
> - Original Message -
> From: "Outback Dingo" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> Sent: Friday, March 21, 2008 11:06:31 AM (GMT-0600) America/Chicago
> Subject: Re: [asterisk-users] www.cdsportal.net wholesale voip
> provider --starting at 1.1 cent per min
>
> My first thought looking at the site was "SCAM"!!!  maybe my second
> thought would be "SCRAM" ... is this company even "legit"
>
> On Fri, Mar 21, 2008 at 10:35 PM, Tim Nelson <[EMAIL PROTECTED]>
> wrote:
>
> > Apparently the list description of "Non-commercial Discussion" isn't
> > clear enough. And now the obligatory beat down:
> >
> > "Instant Emergency Response" and "Delay Free Connection"... WOW! I don't
> > even have to call for support because when I have an emergency, response is
> > INSTANT. On top of that... they've also figured out how to eliminate
> > latency!!! Super duper!
> >
> > But wait, theres more!!! They are interconnected with major US carriers
> > like QUEST!!! Not to be confused with QWEST... the little telco company that
> > misspells it's name to differentiate itself from the ULTRA MEGA HUGE telco
> > QUEST.
> >
> > 
> >
> > Tim Nelson
> > Systems/Network Support
> > Rockbochs Inc.
> >
> >
> > - Original Message -
> > From: "Ignacio Ortega A." <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> > Asterisk-Users@lists.digium.com>
> > Sent: Friday, March 21, 2008 10:20:17 AM (GMT-0600) America/Chicago
> > Subject: [asterisk-users] www.cdsportal.net wholesale voip
> > provider --starting at 1.1 cent per min
> >
> > starting a 1.1 cent per min, rates may be better depending volume
> > technical support
> > we support all codecs using SIP / IAX2
> > predictive dialers, call centers and telemarketers are allowed
> > free test account.
> >
> > if you have any question just contact us
> > [EMAIL PROTECTED]
> >
> >
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>
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Re: [asterisk-users] Calling a Macro with arguments in AstApplication/AstApplicationData

[Gonzalo Servat]

On Sat, Mar 15, 2008 at 8:25 PM, Gonzalo Servat <[EMAIL PROTECTED]> wrote:

> [..snip..]
>
> AstExtension: 210
> AstApplication: Macro
> AstApplicationData: call-ext,SIP/testuser&IAX2/testuser,210
>
> When I dial this extension, I see the following in the log:
>
> -- Executing Macro("SIP/testuser-082b11f8",
> "call-ext,SIP/testuser&IAX2/testuser,210")
> [Mar 15 16:14:06] DEBUG[6060]: pbx.c:2679 pbx_extension_helper: Launching
> 'Set'
> -- Executing [EMAIL PROTECTED]:1] Set("SIP/testuser-remote-082b11f8",
> "LOCAL(arg1)=SIP/testuser&IAX2/testuser") in new stack
> [Mar 15 16:14:06] ERROR[6060]: app_stack.c:370 local_write: Tried to set
> LOCAL(arg1), but we aren't within a Gosub routine


Found a solution to this so I'm replying to myself just incase anyone else
runs into this problem. Probably not the best solution but it works. Since
I'm using AEL, pbx_ael creates the macro for me in the dialplan and it
automatically adds:

Set(LOCAL(argument)=value);

.. for every macro argument. Turns out I can just use ${ARG1} and ${ARG2}
(etc) in the macro itself and it works fine!

Best regards,
Gonzalo
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[asterisk-users] Calling a Macro with arguments in AstApplication/AstApplicationData

[Gonzalo Servat]

Hi All,

This question is probably more for the LDAP experienced users/developers as
I'm sure it would work fine if I weren't using LDAP (but I am, and I'm
almost there with the setup!!!).
I've setup an extension with the following:

AstExtension: 210
AstApplication: Macro
AstApplicationData: call-ext,SIP/testuser&IAX2/testuser,210

When I dial this extension, I see the following in the log:

-- Executing Macro("SIP/testuser-082b11f8",
"call-ext,SIP/testuser&IAX2/testuser,210")
[Mar 15 16:14:06] DEBUG[6060]: pbx.c:2679 pbx_extension_helper: Launching
'Set'
-- Executing [EMAIL PROTECTED]:1] Set("SIP/testuser-remote-082b11f8",
"LOCAL(arg1)=SIP/testuser&IAX2/testuser") in new stack
[Mar 15 16:14:06] ERROR[6060]: app_stack.c:370 local_write: Tried to set
LOCAL(arg1), but we aren't within a Gosub
routine


.. which means I'm not able to access these arguments inside the Macro :(
FWIW, I've defined this Macro in extensions.ael (not .conf).

Any ideas??

Thanks in advance!
- Gonzalo
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Re: [asterisk-users] LDAP

[Gonzalo Servat]

On Fri, Mar 7, 2008 at 9:52 AM, Faraz Khan <[EMAIL PROTECTED]> wrote:

> It does work. Did you do the switch statement in extensions.conf?
>
> If not check voip-info for "Asterisk Realtime Extensions"
>

Hi Faraz,

I just realised I never replied to this message. Yes, you were right. I
simply had to add "switch" to the right context and it worked smoothly.

I've actually managed to get it setup the way I want it (I'm going to write
a HOWTO when I get a few minutes on how I did it, for the next person). I
just managed to get VoiceMailMain() and Voicemail() to work straight from
LDAP which is way-cool. I was wondering if you know (or if it's even
possible) to set the different voicemail settings that one can normally set
in voicemail.conf into LDAP (I'm talking about things like the user's
voicemail password, email address for sending voicemails and the last column
that specifies the different voicemail switches).

Thanks very much again for your help!!

Best regards
Gonzalo
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Re: [asterisk-users] How to find out the IP of the calling party?

[Gonzalo Servat]

On Fri, Mar 14, 2008 at 9:01 AM, Rizwan Hisham <[EMAIL PROTECTED]>
wrote:

> I dont know about IAX, but for SIP users you can use the function
> SIP_HEADER(headername) to get the information u need from the sip packets.
> for example you can use SIP_HEADER(From) which will give you the From header
> containing the IP address of the caller. You will have to apply regex on it
> to extract the ip.
>

Hi,

Thanks to everyone for replying. I did look at SIP_HEADER and it could work,
however, I was hoping for something I could use for both SIP and IAX. I
think I might have to go with the solution I didn't really like (use 2
accounts for each user with different contexts and permit/deny lines). I
thought there must be a way to do it using AGI but it doesn't look like it.

Regards,
Gonzalo
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[asterisk-users] How to find out the IP of the calling party?

[Gonzalo Servat]

Hi All,

I'm trying to achieve the following:

- If  logs in from home, they can dial internal extensions
only (this is to avoid employees going wild on local/mobile calls from home)
- If  logs in from the office, they can call anyone they want.

Since I have my users defined in an LDAP tree, I'd like to stick to
one-account-per-user (each account is setup for both - IAX and SIP logins -
to allow the employee to use IAX from home and SIP at work, or whatever
combination they prefer).

So, I thought I would simply look at the IP address of the originating call.
If the SIP/IAX user has an IP address outside the local subnet -> allow
calls to extensions only. Else -> allow all. I thought the best way of doing
this would be using AGI with a Perl script. The only problem I'm having is
determining the IP address of the originating call. I can't find any channel
variable that gives me this info.

The reason why I mentioned that I'd like to stick to one-account-per-user is
that I know I could fix this simply by having 2 accounts per user (one that
allows connections from the local subnet, and the other to login from
outside and use different contexts for each), but it'd be much nicer to
avoid having 2 accounts per user.

If anyone has any suggestions on how to achieve the above, I'd love to read
them!

Thanks in advance.
Gonzalo
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Re: [asterisk-users] Asterisk not transcoding between installed codecs

[Gonzalo Servat]

On Wed, Mar 12, 2008 at 12:58 PM, Brent Davidson <
[EMAIL PROTECTED]> wrote:

>  Do you have canreinvite=no in the sip client configuration?  If not then
> the two sip phones are probably issuing a reinvite command and taking
> asterisk out of the call path.  If that happens and the phones can't reach
> consensus on a codec then you run into audio problems.  If you're not a
> provider and just using asterisk as a PBX then it's probably better to set
> the phones up with a matching codec set and allow them to establish a direct
> connection between each other to keep load off the Asterisk server.
> Otherwise set canreinvite=no and Asterisk should transcode correctly.
>

Brent,

Thank you vry much for replying. I thought the message went unseen but
found your reply when I went to look at the thread :)

You're absolutely right. Looks like the SIP client was messing up (or
something) when different codecs were used. I tried canreinvite=no and it
worked perfectly, but as you said, it's best to bypass Asterisk when talking
between clients on the same network. I tried a different IAX client and it
had no problems using different codecs (with canreinvite=yes) so all is
good.

Thanks again!
Gonzalo
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[asterisk-users] Asterisk not transcoding between installed codecs

[Gonzalo Servat]

Hi All,

I have 2 SIP clients configured and connected to Asterisk. When I place a
call from SIP1 to SIP2, if both codecs are the same then everything works as
expected. I then allowed one of the clients to use alaw instead of ulaw and
there were audio problems (couldn't hear the other end, etc). Same thing
happened when I tried to use gsm<->alaw/ulaw.

Any ideas? I'm using 1.6.0-beta4.

Thanks!
Gonzalo
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Re: [asterisk-users] LDAP

[Gonzalo Servat]

On Fri, Mar 7, 2008 at 8:46 AM, Gonzalo Servat <[EMAIL PROTECTED]> wrote:

> On Fri, Mar 7, 2008 at 5:50 AM, Faraz Khan <[EMAIL PROTECTED]> wrote:
>

> Also please keep this list in your replies. I have no problems
> > answering personal emails but both of us might get more feedback if we
> > post our progress on the list! :)
> >
>
Faraz,

I just noticed on the bounty you posted up that you mentioned that simple
extensions work fine. I can't even get this to work! I've added:

extensions => ldap,"dc=argentina,dc=hrsmart,dc=com",extensions

.. to extconfig.conf and:

; Extensions Table
[extensions]
context = AstContext
exten = AstExtension
priority = AstPriority
app = AstApplication
appdata = AstApplicationData
additionalFilter=(objectClass=AsteriskExtension)

... to res_ldap.conf and created a user of type AsteriskExtension, set the
context, etc. When I dial the extension defined in AstExtension, it doesn't
attempt a lookup on LDAP at all. Any ideas?

Best regards,
Gonzalo
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Re: [asterisk-users] LDAP

[Gonzalo Servat]

On Fri, Mar 7, 2008 at 5:50 AM, Faraz Khan <[EMAIL PROTECTED]> wrote:

> Gonzalo,
> Please let us know what you mean by 'stops working' - it should spit
> out errors or wrong queries to ldap.
>

Basically what I mean by that is that in the slapd debug, no activity was
going on when I tried to authenticate from a SIP client. After restaring
Asterisk a few times, it magically started working again.

It's not really concerning me as it hasn't done it since. I also filed a bug
in regards to the userPassword issue:

http://bugs.digium.com/view.php?id=12163


> Also please keep this list in your replies. I have no problems
> answering personal emails but both of us might get more feedback if we
> post our progress on the list! :)
>

Absolutely. Replying to you off-list was unintentional.

Best regards,
Gonzalo
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Re: [asterisk-users] LDAP

[Gonzalo Servat]

Hi again :)

I've downloaded, compiled & installed 1.6.0-beta4 --with-ldap. After a few
hours of messing with it, I've managed to get it to say that it has
connected successfully to the LDAP backend (by looking at the output of
"realtime ldap status").

I've modified extconfig.conf to what it should be (after reading many
different configs on the subject). The trouble I'm having now is actually
authenticating with a SIP user. I am running slapd in debug mode (slapd -d
4095) and I would have expected to see lots of activity on the console when
I attempt to authenticate as a SIP user, but I see none at all. Is this
normal?

Thanks!

Regards,
Gonzalo

On Thu, Mar 6, 2008 at 12:37 AM, Gonzalo Servat <[EMAIL PROTECTED]> wrote:

> Hi All,
>
> I've just compiled Asterisk 1.4.18 and I'm planning on using an LDAP tree
> where the users will each have their account, SIP username/password,
> extension number, context, etc. My first question is: can this be done with
> 1.4.x? If so, where can I get the res_config_ldap from??
>
> I googled quite a bit and found a res_config_ldap that looks to be coded
> for 1.2. Is anyone running Asterisk with LDAP? Is it stable?
>
> Thanks in advance.
>
> Regards,
> Gonzalo
>
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[asterisk-users] LDAP

[Gonzalo Servat]

Hi All,

I've just compiled Asterisk 1.4.18 and I'm planning on using an LDAP tree
where the users will each have their account, SIP username/password,
extension number, context, etc. My first question is: can this be done with
1.4.x? If so, where can I get the res_config_ldap from??

I googled quite a bit and found a res_config_ldap that looks to be coded for
1.2. Is anyone running Asterisk with LDAP? Is it stable?

Thanks in advance.

Regards,
Gonzalo
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Re: [asterisk-users] Is GoVarion a fraud ???

[Gonzalo Servat]

On 10/23/07, Alan Lord <[EMAIL PROTECTED]> wrote:
>
> Luis Antonio Prata Barbosa wrote:
> > Hi,
> >
> > Some days ago I spent about US$700,00 in a Tormenta III board in
> > www.govarion.com . I used credit card.
> > I didn't receive any answer for my emails and there is no telephone
> > number to contact them..
> >
> > Now, I'd like to cancel this order, because I couldn´t wait so long, and
> > my credit card was billed.
> >
> > Is www.govarion.com  a fraud   Does anybody
> > know something about them ??
>

Not sure about fraud, but I did find this:

http://threebit.net/mail-archive/asterisk-users/msg37367.html

Somebody complaining about them in 2006. It looks like they exist, they're
just very very slow.

Good luck!
- Gonzalo
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Re: [asterisk-users] asterisk audits

[Gonzalo Servat]

On 9/27/07, Tilghman Lesher <[EMAIL PROTECTED]> wrote:
>
> On Wednesday 26 September 2007 18:39:31 Mark Quitoriano wrote:
> > Some company asked me to do audits with there asterisk boxes. Is there a
> > standard that i should be following in auditing? anyway can give me a
> start
> > what to do with asterisk audits?
>
> Have you considered the ethics of getting yourself hired to do something
> you
> don't know how to do?  Worse, have you considered the ramifications of
> posting
> to a publically archived list that you got yourself hired to do a job
> you're
> unqualified for?


I agree with what you're saying (personally I wouldn't accept the job),
however I think that it's his business whether he accepts it or not. The one
who will face the client will be him, not you, so the replies should
probably stick to the technical aspects and not ethical matters.

- Gonzalo
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Re: [asterisk-users] Community PBX?

[Gonzalo Servat]

On 6/15/07, Kyle Sexton <[EMAIL PROTECTED]> wrote:


I'm wondering if anyone out there is running a community PBX for their
local Asterisk User Groups or area Linux groups.  I've been thinking of
setting one up but am stuck as to what services to provide that people would
actually find useful.  I know that I could setup simple SIP->SIP to allow
everyone to call each other, but that's not generally too fun.



.. and about 500 million other places already offer this :)

One suggestion would be to setup something like a "virtual office" where you
could allow users to point their DIDs to their SIP/IAX account on your
server and receive calls, maybe allow free calls to freecall destinations,
voicemail, stuff like that.

As an example of a community project, there's www.voipuser.org which, I
think, had the right idea. They turn their income from inbound calls into
outbound minutes for the community to use. Pretty smart!

- Gonzalo
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Re: [asterisk-users] Macro inside macro

[Gonzalo Servat]

On 8/13/06, Attilla De Groot <[EMAIL PROTECTED]> wrote:
[..snip..]

Sorry, didn't thought it was relevant, since the entire macro gets
executed, but here it is.

;recording
exten => _*22*XXX,1,Macro(record,conference,${EXTEN:4))


I think what you probably want is:

exten => _*22*X.,1,Macro(record,conference,${EXTEN:4})


exten => _*23*.,1,Macro(record|dialout|31455200025|SIP/${EXTEN:4}
@voipbuster)


If you have _*23*., it means it will match *23 as well as
*23*, but not *23*123456 which is probably what you
want. Try:

exten => _*23*X.,1,Macro(record|dialout|31455200025|SIP/${EXTEN:[EMAIL 
PROTECTED])

Also, from memory, the "h" extension gets executed from the main
context. After making the above changes, try adding this:

exten => _*23*X.,h,System(/etc/asterisk/mail.sh [EMAIL PROTECTED]
${CALLFILENAME} &)

.. and remove the "h" extension from macro-record.

Let me know if the above helps.

Regards,
Gonzalo.
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[asterisk-users] Re: Ever donate Software to Digium? If you did your afool.

[Gonzalo Servat]

On 8/9/06, Yaakov Menken <[EMAIL PROTECTED]> wrote:

I really don't understand the complaint. Fonality gets a $5 mil.
investment for building its own system on top of Asterisk -- no
complaint. But Mark & Co. can't get VC for their own business /
enterprise / support architecture?

Everything that "we-all" is contributing is part of the open source
Asterisk system, which Mark dreamed up and which continues to be an
unbelievably exciting piece of software.

One day soon I hope programmers who work for me will contribute to the
code base. The fact that Digium just got $13.7 mil. to keep doing what
they're doing merely ensures that we who use the open-source base will
all be benefiting from a stable and growing team of professionals
pushing the project forward.

Congrats, Digium. I for one am delighted, and look forward to joining
the "fools."


The original post must be the biggest troll in history.

I think all of us are thankful for having Asterisk. Infact, I'm glad
that Digium is getting good $$ out of it (apart from all the hardware
sales) since many of us can't afford to pay Mark the sort of money he
deserves for his time spent writing Asterisk. I know Asterisk has
earned me a few bucks from the various installations I've done, so I'm
happy to hear this bit of news.

Of course, Asterisk is a community-supported project so my thanks also
goes to all the people that have made it what it is today. Maybe one
day soon I will be able to contribute to it (apart from a few Wiki
additions I've done).

Cheers,
Gonzalo
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Re: [asterisk-users] Recommend hard phone which supports IAX2?

[Gonzalo Servat]

On 7/25/06, Stephen Bosch <[EMAIL PROTECTED]> wrote:

Hi:

I'm setting up a branch office, but I don't want to trunk from the main
office because I don't want to introduce any more latency. Also, the
office will have only a single extension, so I can't justify the expense
of a second Asterisk server for it.

SIP is a pain when going through firewalls, and I'm worried about the
latency that would come with using an IPsec tunnel between the two
sites, so I'm looking for an IAX2 supporting hard phone, and want to
hear recommendations from people who have had direct experience with such.

What are the best IAX2 hard phones?


Hi,

You could just get an IAXy and connect an analog phone onto it?

Regards,
Gonzalo.
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Re: [asterisk-users] Regular expression problem

[Gonzalo Servat]

On 7/24/06, Benjamin Stocker <[EMAIL PROTECTED]> wrote:

Hi!

What's wrong with this?

exten => s,1,Set(myvar="nothing")
exten => s,2,Set(myvar = $["${CALLERID(num)}" : "([a-z]+)"])
exten => s,3,NoOp(${myvar})


Try removing the spaces on either side of the "=" symbol.

Regards,
Gonzalo
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Re: [asterisk-users] OT: Project Management & Collaboration Software

[Gonzalo Servat]

On 7/20/06, calvis <[EMAIL PROTECTED]> wrote:


We are looking at various software packages that do Project Management &
Collaboration.  Since I value the opinions of this list I would be
interested in how others are dealing with Project Management &
Collaboration.   By collaboration I mean the sharing of emails, contacts,
tasks, and files among team members.

If you wish you can send me a post off-list about the solution you are
using.


I think you just brought the acronym "OT" to a new level ! :-)

Anyway, there has been a LOT of talk about Zimbra lately. People have
been saying real good things about it so it's probably worth a look.

Cheers,
Gonzalo
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Re: [asterisk-users] re:Simple But important question (for me)

[Gonzalo Servat]

On 7/19/06, Matthew Warren <[EMAIL PROTECTED]> wrote:

We build custom scripts for Asterisk.  We can build this for you, for
reletivly inexpensive.  But you will need to contact me thru email at
mwarren "at" procomconsulting "dot" com   .. This is a commercial app you
need but requesting on a non commercial group.


Actually it is a non-commercial solution he needs and you offered a
commercial one in a non commercial group. ;-)

Cheers
Gonzalo
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Re: [Asterisk-Users] Question setting up a "bat phone" extension.

[Gonzalo Servat]

On 6/11/06, James Harper <[EMAIL PROTECTED]> wrote:
[.snip.]

My dialplan in the pap2 is:

(<:0>S0)

Which causes it to dial a '0' to asterisk as soon as I gets picked up.
In my asterisk dialplan it then does a DISA to another context, which
means Asterisk is doing all the dialplan stuff. For what I want in a
dialplan, I could have configured it in the pap2 but I didn't want to
learn it. I think I'm at that age where everything new I learn means
something else gets overwritten :)

[.snip.]

This is pretty cool! Thanks James. Now I just keep the Asterisk
dialplan configured and can leave the PAP2 dialplan untouched. The
only functionality I'd loose is the ability to use the *xx codes for
the PAP2. right?

Cheers,
Gonzalo
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Re: [asterisk-users] FW: $3,000 server

[Gonzalo Servat]

On 7/12/06, Alex Robar <[EMAIL PROTECTED]> wrote:

His work ethic is fine... He either couldn't do the job you wanted, or he
didn't want to, so he sent you to someone else. If you paid Greg instead of
NuFone, then that's really tough shit for you for not following the
instructions Jeremy gave you. Furthermore, Jeremy gets to wash his hands of
the issue because you never paid him... If I hire a tech to go out to your
site and install a server, but you pay the tech instead of my company, then
you're dealing with the tech, not me. If something breaks, you don't get to
come crying to me because you circumvented my company's payment system and I
never saw a dime.

Jeremy/NuFone has every right to send lawyers after you for saying the
things you're saying. He's right to say what you're doing is extortion. I
certainly hope he wins his case... Grow up.


I completely and wholeheartedly agree 100% with what Alex just said. I
certainly hope this thread has come to an end. I've seen far more
interesting threads stop at maybe 5 or 6 replies, and this one which
is dead boring and filled with lawyer threats has gone way too far.
Granted I'm not helping by replying, but I had to say Alex did a good
job at summirizing the issue and let's hope this thread is now closed.

Cheers,
Gonzalo.
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Re: [asterisk-users] Re: $3,000 server

[Gonzalo Servat]

On 7/12/06, Jeremy McNamara <[EMAIL PROTECTED]> wrote:

Michael Workman wrote:
>  Well you told me to talk to him
> And he said he worked for NuFone so its not My Fault Period...
>
> Its your Fault for pushing me to Him and You allowing him to say he works
> for NuFone...
> If Anything You should Go After Greg for the Money he Ripped off from US...
>
> AND WE HAVE MSN HISTORY OF CHAT FROM GREG SAYING WE PAID NUFONE FOR WORK


If you did not pay [EMAIL PROTECTED] via PayPal thus you did not pay
NuFone Inc.

I purposely replied to you off-list to resolve this situation. The only
reason to post your reply to asterisk-users was to publicly defame my
company.


He is not doing a very good job, Jeremy. For starters I sincerely
doubt he actually paid anyone $3,000. He sounds like somebody just
trying to defame NuFone for the fun of it. Also, the more he writes,
the more I think he's an idiot (much like Robert). He clearly shows
his lack of intelligence in his e-mails. I wouldn't pay anyone $3,000
without first having written confirmation of the work that will be
performed, who and where the person works, and other accompanying info
to avoid being "ripped off".

Regards,
Gonzalo.
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Re: [Asterisk-Users] Time Based Goto Ifs Act Strange?

[Gonzalo Servat]

On 6/21/06, Matt <[EMAIL PROTECTED]> wrote:

Hi,
I'm still in the process of debugging this, but I have a gotoif
statement that looks like this:

exten => 26,1,GotoIfTime(7:00-18:00|mon-fri|*|*?ext-queues,210,1)
exten => 26,n,Goto(ext-local,${VM_PREFIX}127,1)

[..snip..]

Hi Matt,

Are you sure the date/time is set correctly on the Asterisk machine?
Reason I ask is that you said it worked fine the first time, then you
rang at 18:05 and it went to queue 210. Maybe the machine's clock was
17:59 or something when you rang. You rang right back which went to
VM_PREFIX. Highly coincidential and unlikely, but maybe you rang just
before 18:00 and again just after 18:00, as far as the Asterisk
machine's time is concerned.

Also, try using | as the separator instead of ",". ie:

exten => 26,1,GotoIfTime(7:00-18:00|mon-fri|*|*?ext-queues|210|1)

I doubt it makes any difference otherwise it would have never worked,
but you never know. The show application gotoiftime says the separator
is "|".

HTH,
Gonzalo.
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Re: [Asterisk-Users] AEL2

[Gonzalo Servat]

On 6/9/06, Joshua Colp <[EMAIL PROTECTED]> wrote:
[..snip..]


I'd just like to note that AEL2 was brought over into Asterisk trunk
(what will become 1.4) and the old AEL removed. That's where most
development is taking place on AEL2, and why you don't see patches on
the bug tracker.


Hi Joshua,

I was just reading the bug report and noticed it has been merged.
Awesome news! I'm still using 1.2.x so sticking to AEL for now, but
I'm going to quickly move to AEL2 as soon as I upgrade to 1.4!
(whenever it comes out)

Regards,
Gonzalo.
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Re: [Asterisk-Users] AEL2

[Gonzalo Servat]

Hi Doug,

On 6/9/06, Doug Crompton <[EMAIL PROTECTED]> wrote:

Replying to myself here... I got the latest 1.2 head via svn. Did a patch
diff'ed it to latest AEL2 (as described at:
http://voip-info.org/wiki/view/Asterisk+AEL2

Patched it. All went fine. On compile I get the following error

[..snip..]

This probably won't help you much, but you'll find one big bug report
in the Digium Bug Database used for AEL2 where you'll see lots of
input from the AEL2 guy. He's a good guy and real helpful. Infact,
here's the URL:

http://bugs.digium.com/view.php?id=6021

Hope this helps!

Regards,
Gonzalo.
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Re: [Asterisk-Users] WCTDM-24xxp woes

[Gonzalo Servat]

On 6/4/06, Andrew D Kirch <[EMAIL PROTECTED]> wrote:

I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N
Nforce based board).  I am using a Digium wctdm24xxp to terminate 8 (2x
quad) FXO lines.  Using ztmonitor (From zapata) I cannot see that there is
any registerable incoming volume from these lines.  I've been running them
at rxgain = 25 (zapata.conf) to make the audio audible, however this
creates poor call quality issues (static and distortion) on most calls,
and audio garble in voicemails.   Fxotune fails for every line with "Could
not fill input buffer"
I've tried changing PCI slots, played with echo settings, and done
everything else I can think of to make this card play nice to no avial.

Anyone with solutions or ideas, your input will be greatfully appreciated.


Hi,

I'm using one of these cards with 3 quad modules (4 x FXO, 8 x FXS)
and I didn't have to touch the rxgain/txgain (0.0 for both). This is
how it should be set. If you need to adjust these settings, then you
have another problem. I know you tried changing PCI slots, but have
you looked at /proc/interrupts to see if the card is sharing an IRQ
with another device?

Regards,
Gonzalo.
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Re: [Asterisk-Users] Monitor application and e-mailing attachment

[Gonzalo Servat]

On 6/4/06, Attilla De Groot <[EMAIL PROTECTED]> wrote:

Hi all,


I'm trying to make a context that will monitor a call and when it's
completed it would e-mail the wav to a specified mail adres.

So I made a standard context that records a call, like this:

exten => _*31*00[1-9].,1,Setvar(CALLFILENAME=CALL-${EXTEN:4}-$
{TIMESTAMP})
exten => _*31*00[1-9].,2,Monitor(wav,${CALLFILENAME},m})

[..snip..]

Not sure if this is the problem or if you made a typo when sending the
list email, but you seem to have put an extra } by accident:

exten => _*31*00[1-9].,2,Monitor(wav,${CALLFILENAME},m})

.. should be:

exten => _*31*00[1-9].,2,Monitor(wav,${CALLFILENAME},m)

Regards,
Gonzalo
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[Asterisk-Users] Asterisk connecting to a proprietry PBX

[Gonzalo Servat]

Hi guys,

I'm interconnecting an Asterisk box with a Lucent Definity PBX by
means of FXO/FXS ports on a TDM2400 card. Everything works well,
except for one little thing. Every now and then somebody (from an
Asterisk extension) will call another extension on the Lucent Definity
PBX and they hit their voicemail. They caller leaves their message (or
not) and hangup, BUT the Lucent sometimes doesn't detect the hangup
and the channel never gets dropped. The call goes on and on and on (I
often have calls hanged for 20 hours or more). The problem is that
with the limited FXO/FXS ports we have, I often have people complain
they can't dial somebody, but it's simply because there are hanged
channels.

I did a ChanSpy on the hanged channels and all there is is dead
silence. I guess Asterisk is not going to drop the channel until it
hears the busy pattern, which I guess makes sense as you wouldn't want
to be hanged up on if there is any silence (obviously).  I was
wondering if there is some way I can configure Asterisk to hang up
calls if there is more than X minutes silence in a channel. For now, I
think the quick (and dirty, and not very nice) solution is to set an
AbsoluteTimeout of 1 hour on the calls, as 95% of calls are less than
one hour and will ensure there are no hanged chans longer than 1 hour
in the system.

Another idea I had in mind is to, maybe, set the voicemail system
(Audix) to cut messages after X minutes. I don't know for sure if the
(Asterisk) caller is leaving a message or not, as most voicemail
systems have a message size limit and drop the call anyway. Strange,
huh?

Any suggestions? opinions? sympathizing words? :-)

Thanks in advance guys.

Cheers,
Gonzalo
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[Asterisk-Users] PAP2/Sipura XML Provisioning File

[Gonzalo Servat]

Hi All,

I have a number of SPAX00X units (spa1001, 2002, etc) and about 30 odd
PAP2-NA units all hooked up to Asterisk. As you can imagine, setting
them up took a while, and changing settings on them also takes a
while. In order to prepare for future deployments, I'd like to use XML
provisioning (or any kind of remote provisioning). I figured since
Sipura/Cisco won't release the utility to create the file unless
you're a bigtime reseller, my only option is to use a XML file.

Does anyone have Sipura/Linksys ATAs sample XML files?

Thanks in advance for any help.

Regards,
Gonzalo.
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Re: [Asterisk-Users] Hi...Please help me

[Gonzalo Servat]

On 4/24/06, Crazy Boy <[EMAIL PROTECTED]> wrote:
> Hi Friends,
>
[..snip..]
> ---> Employee 1 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 2 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 3 PC (Softphone i.e., Headphones with Mic)
> --->--
> --->--
> ---> Employee 10 PC (Softphone i.e., Headphones with
> Mic)
>
> and vice versa.
>
> How can I implement this? Is it possible to implement this using "Asterisk"
> software? If It can be implemented using "Asterisk" software, What softwares
> I should install in Server and Employee PC's? Is there any need of buying
> extra hardware?
[..snip..]

It can be done with Asterisk. For the server side, you would need to
install Asterisk on your Fedora 5 box, Zaptel and lots of Wiki
reading.

I don't recommend using softphones for your employee PCs. It looks
like an attractive solution at first (from a cost perspective) but in
reality it's not very practical (at least that was my experience).
Buying 5 x 2 port ATAs will cost you around $300-$350 which is not
really expensive considering the kind of powerful PBX you will have at
your disposal. I would have suggested some Digium hardware for the FXS
(extensions) but I think it will be a lot more expensive (for 10
extensions) than the ATAs solution. You could also look into a channel
bank, but again it will be more expensive than the 5 ATAs. As for the
FXO (incoming/outgoing PSTN) I recommend buying Digium hardware
(TDM400P).

Hope this helps, and good luck!

Regards,
Gonzalo.
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Re: [Asterisk-Users] Sipura SP3000 question

[Gonzalo Servat]

On 4/23/06, Roshan Sembacuttiaratchy <[EMAIL PROTECTED]> wrote:
> I use the following dialplan within the Sipura:
>
> ([2-79]11<:@gw0>|999<:@gw0>|112<:@gw0>|0[12]x.|[*x]xx.<:@gw0>|<#9,:>[*x]x.|**)
[..snip..]

Is this @stuff something new in the SPA3000 dialplan syntax? I have
SPA-200x ATAs and I never saw any mention of this in the manual, which
makes sense if it's a SPA3k new dialplan feature.

Cheers,
Gonzalo.
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Re: [Asterisk-Users] SPA-941 Selective DND

[Gonzalo Servat]

On 2/27/06, Darren Ellis <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I have a request from a customer that I'm not sure how to implement.
> They have a Snom-360 as receptionist phone and SPA-941 for all other
> phones.  They use the SPA-941 DND function when they are away from their
> desks, which happens often due to the nature of their business.
>
> They would like to have the SPA-941 accept internal calls while DND is
> set.  If any of you know how to make this happen, I'd very much
> appreciate your help.
>
> The paging feature is not what they want, and the SPA-041 ignores the
> answer-after=0 SIP header when DND is on anyway.

IF the SPA-941 doesn't support the selective DND feature, the only
solution that comes to mind is to use server-side DNDs. ie. *11 (for
example) to turn DND on and *12 to turn it off.
In Asterisk, configure *11 to set a DND variable (database put DND ext
yes/no). When somebody calls the extension, if DND is "yes" in the
Asterisk internal DB for the extension, and the call is from a local
channel, ring the SPA-941, otherwise send to voicemail. Make sense?

Hope this helps.

Regards,
Gonzalo.
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Re: [Asterisk-Users] Best ATA for general residential deployment??

[Gonzalo Servat]

On 2/22/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
[snip]
> The information needed for XML provisioning is openly available from
> sipura/linksys. The actual linksys provisioning tools may be under some
> license but the XML provisioning syntax is not. It is actually
> ridiculously simple.
[snip]

The only XML info I found on the Sipura site was for the SPA 841.
Nothing on the ATAs. Have looked on the Linksys site too and no info
at all on it. Do you have a link?

Cheers,
Gonzalo.
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Re: [Asterisk-Users] Best ATA for general residential deployment??

[Gonzalo Servat]

On 2/22/06, Darrell Long <[EMAIL PROTECTED]> wrote:
> Correct. The XML works fine. If you need an example for the 2002, I will
> see if I can strip the information directly related to our company off
> and send it to you.

Hi Darrell,

I would really appreciate it if you could send me the XML file
(offlist), of course remove any company sensitive info. I have the
SPA841 sample XML file and while it's a good base to start from, it
has many SPA841 specific settings, so it would be better to get a copy
of your 2002 one. This same XML file should work with a Linksys
PAP2-NA, right? (it's basically the same device as a SPA-2002). If you
don't know whether it works with one, I'll soon let you all know :)

Thanks,
Gonzalo
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Re: [Asterisk-Users] Best ATA for general residential deployment??

[Gonzalo Servat]

On 2/22/06, Matt <[EMAIL PROTECTED]> wrote:
> Try the Sipura SPA-2002.. at good prices from VoipSupply.com
>
> We have been using those now with 0 problems.  We remote provision
> them from our office here.  Once a minute (time configurable) each
> device checks in with us to check out its configuration file and see
> if it needs updates.   The devices run around $60 a piece, so they are
> pretty cheap as well.

RE the remote provisioning, did you have to pay some sort of license
fee to get access to the tools to generate the remote provisioning
configurations and instructions on setting it all up?

I have 12 x PAP2-NA/SPA-2002 and changing one setting means going
around and changing all the settings for each line of the 12 ATAs,
that's 24 configuration changes in total - a real PITA. If you know of
a way to obtain the tools to do the remote provisioning, I'd be
grateful!

Thanks,
Gonzalo.
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Re: [Asterisk-Users] GotoIf number exists in file. How can i do this?

[Gonzalo Servat]

On 2/8/06, Arne Morten Johansen <[EMAIL PROTECTED]> wrote:
>
>
>
> Oh. So how can I do this?
>
> If I write something in PHP, how do I make it output to an Asterisk
> variabel? I need to set a variable in asterisk to TRUE or FALSE based on the
> result of the PHP-script.

You can find the answer to this question together with a few example
PHP scripts at:

http://www.voip-info.org/wiki-Asterisk+AGI+php

Cheers,
Gonzalo
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Re: [Asterisk-Users] change languages from an IVR

[Gonzalo Servat]

On 2/6/06, Mark Phillips <[EMAIL PROTECTED]> wrote:
> A customer of mine wants an IVR where the first 3 choices are
>
> 1 English
> 2 Spanish
> 3 French
>
> I can build the IVR but how do I get the system prompts to then speak
> the selected langauge. For example, a caller has selected Spanish and so
> is routed to the Spanish part of the IVR. At some point he breaks out of
> the IVR to leave a VM. How does the system know to continue offering him
> Spanish?

Maybe once they've selected the language, set their default language? ie:

exten => 1,1,Set(LANGUAGE()=en)
exten => 1,2,...

exten => 2,1,Set(LANGUAGE()=es)
exten => 2,2,...

exten => 3,1,Set(LANGUAGE()=fr)
exten => 3,2,...

Hope this helps.

Cheers,
Gonzalo
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Re: [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount

[Gonzalo Servat]

On 2/6/06, Joseph Tanner <[EMAIL PROTECTED]> wrote:
> Funny funny.  In this day of free (after rebate) PAP2s, a free (again,
> I assumed after rebate) IP phone seemed plausible.  BTW, check
> walmart.com, they do indeed sell ip phones.
>
> I guess I'll just have to use one of my free DTA310s or my free PAP2 instead.

... and even if they *did* indeed give away free phones, which is
unimaginable as they're in the business of MAKING money by SELLING, do
you really think people are going to come here and tell the world
about it?

Bit gullible, aren't ya... ;-)
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Re: [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount

[Gonzalo Servat]

On 2/6/06, Joseph Tanner <[EMAIL PROTECTED]> wrote:
> > well I've heard that there are "open source" IP phones given away for free
> > in WALMART, I'm seriously thinking to get couple of 'em!!
>
> What phone would this be?  I didn't notice any, but there's 5-6
> Wal-Marts within an hour's drive, I'd love to try to find some.  Never
> can have too many.  Are they regular IP phones that connect via
> ethernet, or do they plug in via usb?  I wouldn't want any usb ones
> for myself, but my dad could use one.

Oh they're giving away both types! and if you hurry, you get a free
Asterisk box to go with it! go go go!
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Re: [Asterisk-Users] Nested MySQL Commands

[Gonzalo Servat]

On 1/13/06, Mike Fedyk <[EMAIL PROTECTED]> wrote:

[..snip..]

> Also an ugly hack would be to call the perl bytecode instead of the text
> script.  That would allow for the ease of AGI (everything is cleaned up
> when the process exits) with lower overhead.
>
> FastAGI is of course what you want for production, but this can help in
> a pinch.

Also, just a suggestion that was mentioned to me by somebody (whose
nick I can't remember, sorry!) on the #asterisk chan, when using AGI
to do the sort of thing the original poster is after, it's a good idea
to set some "dial" variable and let the AGI exit and Dial from
Asterisk using this variable. This way the AGI exits and doesn't stay
running for the duration of the call.

Regards,
Gonzalo
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Re: [Asterisk-Users] Emailed voicemail messages not being deleted

[Gonzalo Servat]

On 11/28/05, Dustin Wenz <[EMAIL PROTECTED]> wrote:
> According to the Asterisk wiki, adding the delete=yes option to a
> voicemail definition should automatically delete messages after they
> are emailed. This is the format that I'm using:
> 101 => ,First Last,[EMAIL PROTECTED],attach=yes|delete=yes
[snip]

Try:

101 => ,First Last,[EMAIL PROTECTED],,attach=yes|delete=yes

(notice the extra comma after the email address)

I believe the setting that goes in between the empty commas is the
pager email address

Hope this helps.

Cheers,
Gonzalo
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[Asterisk-Users] Distinctive Ring Detection not working

[Gonzalo Servat]

Hi there.

I'm having a strange issue with the distinctive ring detection in
Asterisk (I have a FXO card).
It certainly seems to be enabled as I can see the Asterisk console
spitting out the cadences (same cadence every time: 0,0,0) but the
problem is that it is not waiting 2 seconds after "Starting switch on
Zap/1-1" like it used to, long enough to determine the cadences,
presumably the reason why it is always 0,0,0 as it hasn't had enough
time to detect the ring pattern.

My zapata.conf looks like the following:

[trunkgroups]
[channels]
language=es
context=incoming-landline
signalling=fxs_ks
usedistinctiveringdetection=yes
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
busydetect=yes
busycount=8
dring1=334,146,0
dring1context=secondnumber
channel => 1

I've looked through some of the chan_zap.c code to try and increase
the wait period, but after making a couple of attempts at fixing it
decided to leave it alone before I break something :-)

Another thing I've noticed is that if I *don't* add a dring pattern
for "0,0,0", when a call comes in, it tries to find the dring pattern
for 0,0,0, fails to do so, so it tries to go to context ",s,1" (notice
the missing context name as the first argument), fails to do so and it
supposedly hangs up the chan, then detects the ringing again (it's
still the same call, only in its 5th ring by now) and successfully
detects a pattern different to 0,0,0. This is the only way to have it
"somewhat" working, although it's pretty unreliable. It's coming up
with quite a few different patterns, still, I shouldn't have to do it
this way. A lot of people hang up after the 4th or 5th ring.

Does anyone have any ideas on this?

Any suggestions would be greatly appreciated.

Cheers,
Gonzalo
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Re: [Asterisk-Users] Sipura 2000 Dial Plan

[Gonzalo Servat]

On 9/28/05, Michael Blood <[EMAIL PROTECTED]> wrote:
>
> Anybody ever run into a case where the Sipura Dial Plan will not work with
> the S0 option to immediately connect?
>
> My Dial plan reads
> (*xx|[3469]11S0|0|00|[2-9]xxS0|1xxx[2-9]xxS0)
>
> and I can dial ONLY then numbers in the dial plan so I know that it works.
>
> For some reason when I dial 5551212 1212121212
> It does not dial for a while and then it dials 555 1212

Hi Michael,

I think the problem lies in the S0 suffix that you're adding to the
sequence. Last time I played with the Sipura dial plans I thought S0
and L0 were set dial-plan wide, and not on an individual sequence
basis. ie:

S:0 ( *xx | etc )

(short timeout: 0)

... or:

L:0,S:0 ( *xx | etc)

(long timeout: 0, short timeout: 0)

Check the Appendix in the SPA2000 User Manual to see the format of the
dial plan.

HTH,
Gonzalo
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Re: [Asterisk-Users] Zaptel won't compile under Fedora Core 4

[Gonzalo Servat]

On 7/12/05, Eric Bullen <[EMAIL PROTECTED]> wrote:
> I hope someone can offer me some help with this. Basically, the current CVS
> version of Zaptel will not compile under Fedora Core 4. I have closely
> followed the directions in
> http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
> using the versions given in the FC4 distro with no luck.  Here's the output
> when I run "make linux26". Any help would be great. TIA.

[...snip...]  

>  In file included from /asterisk_source/zaptel/zaptel.c:40:
>  /asterisk_source/zaptel/zconfig.h:10:27: error:
> linux/version.h: No such file or directory

Try installing kernel-source and/or glibc-kernheaders RPMs from the FC4 CDs.

Regards,
Gonzalo
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Re: [Asterisk-Users] Gizmo: Skype done right?

[Gonzalo Servat]

On 7/5/05, Dana Olson <[EMAIL PROTECTED]> wrote:
> I  think they were hoping that the client would connect to Asterisk,
> which makes it kinda useless, really.. But connecting Asterisk to the
> Gizmo network is handy.

Given that it's a fairly new program, we have to wait a while before
it's mature enough to be of any use. For example, I installed it to
check it out, and it crashed as soon as a call came in. This could be
reproduced every time I was about to receive a call :-)

GS
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Re: [Asterisk-Users] Wacko Distinctive Ring Patterns being detected??

[Gonzalo Servat]

On 5/29/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Sat, May 28, 2005 at 06:34:23AM +1000, Gonzalo Servat wrote:
[snip]
> > If Asterisk allowed me to configure up to 10 ringing patterns, I could
> > probably cover most of the ringing patterns being detected, but
> > unfortunately there is a limit of 3 which means 50% (or more) of the
> > calls are coming in under a distinctive ring pattern not configured in
> > Asterisk, and hence going to the default context.
> 
> Is there any deeper reason for that limitation, other than "it didn't
> bother anybody enough"?

I wonder that myself, but I have no idea why the limit is imposed. Any
Asterisk developers willing to answer that for us?

> > Does anyone have any suggestions/ideas/etc on how to resolve this issue?
> 
> Could you post here some ring patterns you get? A distinctive ring can
> identify a pattern that is "similar" enough to an existing pattern.

You're right, some that were not defined were close enough to the
ringing pattern and did match, but even with 3 popular distinctive
rings defined there were still calls that were coming up with a new
distinctive ring pattern and not getting matched by the defined dring
patterns.

Some of the ones I frequently saw were:

334,147,0
383,195,0
334,0,0
336,348,0
334,146,0

All the above patterns are for the same distinctive ring number.

Regards,
Gonzalo
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Re: [Asterisk-Users] Wacko Distinctive Ring Patterns being detected??

[Gonzalo Servat]

On 5/28/05, Jay Milk <[EMAIL PROTECTED]> wrote:
> Could you configure your "normal ring" to be recognized as a distinctive
> ring and go into a different context?  That would essentially allow you
> to distinguish between the calls.

Excellent suggestion Jay! Thanks!! I changed the default context (if
no dring match) to go to the context I configured for this "second"
number and made a distinctive ring match on 0,0,0 to go to the main
number context, and works beatifully.

Thanks for taking the time to reply and for your suggestion!

Regards,
Gonzalo
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[Asterisk-Users] Wacko Distinctive Ring Patterns being detected??

[Gonzalo Servat]

Hi All,

I've recently got a "second" number installed on my PSTN line,
trusting the Asterisk distinctive ring detection would work as
expected. It appeared to work fine at the start, as the second number
generated a different ring pattern to 0,0,0 (in the console) only to
realise that almost every phone call to this "second" number generated
a different ring pattern. Sometimes it might detect the same 2
patterns, but this is a rare case.
If Asterisk allowed me to configure up to 10 ringing patterns, I could
probably cover most of the ringing patterns being detected, but
unfortunately there is a limit of 3 which means 50% (or more) of the
calls are coming in under a distinctive ring pattern not configured in
Asterisk, and hence going to the default context.

Does anyone have any suggestions/ideas/etc on how to resolve this issue?

Thanks in advance guys.
Gonzalo
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Re: [Asterisk-Users] Zaptel Compile on a virtual dedicated host.

[Gonzalo Servat]

On Apr 6, 2005 10:34 PM, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:





















Anyone have any ideas on where I can find the right kernel
source?  I have look at rpmfind.net and
google'd with no avail!Hi,

You're never going to find the kernel source. The reason for this is
that your VPS is running under Virtuozzo, which is a commercial
software package designed to create virtual servers under one physical
server, sharing a common kernel. This means the kernel cannot be
upgraded or in any way modified.

Regards,
Gonzalo


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Re: [Asterisk-Users] Asterisk based Call Accounting software - 1strelease

[Gonzalo Servat]

On Apr 8, 2005 12:48 PM, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote:
Call Accounting is such an important issue for me it is literally a make orbreak component, without it I will not be able to deploy Asterisk at ourresort. If I have to use a windows computer to download and run the clientend of the software, so be it. At least the software will work and I willhave a solution. I think you should be more appreciative they areaccommodating Asterisk and less dogmatic about platform issues.
 
I agree Chris. Someone bothers to contribute to the Asterisk community, and you get the Firefox/Mozilla zealots who have nothing better to say than "I'm not using your software because your site doesn't work with Firefox". If the software is useful to you, and you have to make a little effort to get it (who cares if there are no screenshots!?), then be it. If it's not useful to you, why even bother making an issue about which browser the site is designed to work with?
 
My $0.02.
 
Regards,
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[Asterisk-Users] OT: "No authority found" connecting to Freshtel

[Gonzalo Servat]

Hi,

Has anyone else experienced problems as of the last couple of months
when outbound calling through Freshtel?

I've started getting a "No authority found" error. I've tried
contacting them, and they seem to have some serious communication
issues with their IT team, infact I think they have serious issues in
their IT team full stop. First they can't find my account in their DB,
and I keep being promised they will look into my problem and get back
to me, and of course they never do so I had no choice but to ask the
list just incase it's something on my end, which I seriously doubt as
I've triple checked everything.

Thanks in advance for any help.

Gonzalo
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Re: [Asterisk-Users] Call termination database

[Gonzalo Servat]

On Thu, 17 Feb 2005 10:29:54 +, Alistair Cunningham
<[EMAIL PROTECTED]> wrote:
> I've been considering doing a web based database system, where you can
> post your termination offerings or wanted, then search by location,
> price, minimum volumes, etc.

(snip)

Great idea Alistair. Would certainly come in handy right about now :)
I've been looking for a DID provider in Buenos Aires, Argentina, and
found only one so far with rather expensive pricing. Anyhow, I
digress.  I say go for it if you have the time. Obviously things like
location and pricing are a must. Pricing should be a required field
otherwise it's not so informative if users have to go and find out for
themselves.
I'd suggest a feedback feature too, for every provider, where users
can post their experiences or whatever they consider useful info.
 
Anyhow, good luck! Look forward to seeing it.

Gonzalo
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Re: [Asterisk-Users] Fw: Gift for Mark Spencer

[Gonzalo Servat]

On Tue, 2004-11-23 at 19:06 -0600, Joe Greco wrote:
[..snip..]
>
> On the flip side, senders of spam should not expect recipients to go 
> to much (or any) trouble on their behalf, especially given the current
> spam environment on the 'net.  They - not hackerwaCker - blew the 
> surprise by sending the message to recipients unknown.

I'll just summarise all you said into one conclusion which remains the
same as to what Steven said: hackerwanker is a moron.

:-)

Regards,
Gonzalo

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Re: [Asterisk-Users] Wonderful Success with PAP2-NA

[Gonzalo Servat]

On Tue, 2004-10-19 at 16:56 -0500, Matthew Boehm wrote:
> No. The link you gave is for a PAP2  NOT a PAP2NA. There is a HUGE OH MY GOD
> difference between the two model numbers.

What is this huge OH MY GOD difference between the two? (apart from the
-NA). I've googled and can't seem to find any site that lists the
difference(s).

Regards,
Gonzalo

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RE: [Asterisk-Users] New Open Source Project: Asterisk Management Portal

[Gonzalo Servat]

> > Salutations,
> > 
> > In hopes of accelerating the adoption of Asterisk and changing the 
> > landscape of the small business marketplace, we are contributing our 
> > administration interface to a new project that aims to bundle 
> > best-of-breed applications to produce a "canned" (but fully 
> > functional) 
> > turnkey small business phone system.
> > 
> > Details of the project can be found here:
> > 
> http://amp.voxbox.ca

After looking at the screenshots, I must say it looks very promising.
Great work, and thank you for contributing back to the community!

Regards,
Gonzalo

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Re: [Asterisk-Users] New Project - IP Phone Sources

[Gonzalo Servat]

Hi Stewart,

Nice project! Something I'd certainly love to be doing myself. Anyway,
the following replies I've made to your questions are based on my
experience and past research. There may be better/cheaper alternatives.
In any case, I hope it helps:

On Fri, 2004-10-15 at 12:05 -0400, Stewart M. Ives wrote:
> Hello,
[snip]
> 
> Question: If I just want to provide IP Telephony within the school and have no
> outside connections to the local phone system I suspect I can install Asterisk
> on a RH Linux server and plug in a bunch of IP Telephones on the network,
> config it all and it will work.  The only cost to the school would be the IP
> Telephones.  Correct??  I know it would involve a bit more configuration and
> planning as I have stated but basically is the idea correct??
>
> Question:  What phones or types of phones should I be looking at.  I suspect
> there are new ones coming out every day.  I'm just interested in the most
> basic phone to plug into the network.  Nothing fancy, basic, basic, basic.  I
> also know I can use soft phones but do not want to go there as it makes just
> another application we have to be responsible for on the desktop.
> 
> Many thanks in advance.

Pretty much. You have the following options as far as I can see (and I'm
sure there's more):

1) FXS Adapter - The IAXy[1] is a nice (and cute) device which allows
you to connect a single analog telephone and provide VoIP connectivity
using IAX to your Asterisk server. Buying the device helps support
Asterisk. The only catch is that it only supports one analog phone.
Keeping price in consideration, the only other device I would recommend
is the Sipura SPA-2000 which supports 2 analog telephones per device
(you would need one SPA-2000 per 2 classrooms (one analog phone per
classroom))

2) Digium TDM40B[2] (includes the TDM400P card plus the 4 FXS modules):
This configuration provides 4 x FXS (analog telephone) ports on a single
half-length PCI card. I just checked the Digium site and they're selling
the TDM40B for $305 (works out to be around $76 per telephone).
Certainly the best way of doing it, IMHO. Keep in mind with this
solution you would need telephone wiring FROM the Asterisk server where
the TDM40B lives to all the classrooms. With the IAXy or the SPA-2000
you just need telephone wiring from the unit itself to each classroom
it's providing VoIP to.
Great thing about this solution is that you can mix and match. If, for
instance, the school decided to get a telephone line hooked up to the
system, you can buy a FXO module and swap it for an unused FXS module,
or configure it however you want.

3) VoIP Telephones: Cheapest is the infamous Grandstream[3] BudgeTone
(AKA BarbieTone). Well, actually, I shouldn't say infamous since I've
not had a problem with them myself, but you'll find many reports from
other users on the mailing list archives about the myriad of problems
you can have with them. If you already have a network connection going
into each classroom, this (or the FXS adapters) may be the best option.

Hope this helps!

Best regards,
Gonzalo

[1] http://www.digium.com/index.php?menu=iaxy
[2] http://www.digium.com/index.php?menu=wildcard_tdm400p2
[3] http://www.grandstream.com/y-bt100.htm

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Re: [Asterisk-Users] voicebox

[Gonzalo Servat]

On Wed, 2004-09-15 at 16:10 +0200, wrote:
> Hello!
> I have been googling a lot and asked wiki a few times now, but i cant find
> a howto for setting up a voicebox.
> Any link/hint would be great!

I'd hate to refer you to the Wiki but the answer is in there :) (you did mean a 
voicemail box, right?)

   http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20voicemail.conf

It was a simple search for "voicemail" in www.voip-info.org so I'm not sure what 
freaky search terms you were using.

HTH.

Regards,
Gonzalo

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RE: [Asterisk-Users] How to uninstall Asterisk?

[Gonzalo Servat]

Correcto, I think it's also In My Humble Opinion too.

Gonzalo

P/D: Como andas Seba... :)



On Wed, 2004-07-14 at 11:45 -0300, Sebastian Nocetti wrote:
> IN MY HONEST OPINION... IMHO 
> 
> I am right?
> 
> 
> -Mensaje original-
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] En nombre de ruixun wu
> Enviado el: Miércoles, 14 de Julio de 2004 11:07 a.m.
> Para: [EMAIL PROTECTED]
> Asunto: Re: [Asterisk-Users] How to uninstall Asterisk?
> 
> hi Gus and Roger,
>Thanks for you reply. I choose no load the chan_oh323. The asterisk now
> can start again. :)
>And Gus, could you tell me what's the meaning of IMHO? I can't find the
> topic about IMHO in WIFI.
> 
> Thanks a lot!
> Best Regards
> Rui
> 
> 
> 
> --- CW_ASN <[EMAIL PROTECTED]> wrote: > > Hi,
> > >After I install openh323, the asterisk cann't
> > work
> > > anymore. Asterisk failed in loading chan_oh323. I cann't deleted the 
> > > openh323 package, so the only
> > thing
> > > I can do is to reinstall Asterisk. I checked out
> > the
> > > asterisk and make install Astersik without
> > installed
> > > openh323, but when I started Asterisk, Asterisk
> > still
> > > loaded openh323 and failed.
> > >Does anyone know how to uninstall Asterisk?
> > 
> > If you don't like to load a channel or module, you can choose for two
> > methods:
> > - You can delete it. The channels and apps are located in 
> > /usr/lib/asterisk/modules.
> > - You can choose to not load when asterisk loads.
> > Use modules.conf, set
> > noload => foo.so
> > 
> > At least is strage...  I'm using chan_oh323 without failures, and 
> > IMHO, it's more stable and powerful than others. I'm not wish to start 
> > a war, it's just my opinion.
> > 
> > Regards,
> > 
> > Gus
> > 
> > 
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> >
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> 
> __
> Post your free ad now! http://personals.yahoo.ca
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RE: [Asterisk-Users] asterisk to asterisk config

[Gonzalo Servat]

On 9/07/2004 6:06 AM +0100, Kevin Walsh wrote:
Eugen Cristea [EMAIL PROTECTED] wrote:
Find local movie times and trailers on Yahoo! Movies.
http://au.movies.yahoo.com
What does Yahoo have to do with it?
Have you considered trimming your quotes?  Clearly not.
Have you considered maybe his webmail provider (Yahoo) is automatically 
inserting the advertisement footer? Clearly not ;)

Gonzalo
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Re: [Asterisk-Users] wake-up call script in wiki

[Gonzalo Servat]

On 9/07/2004 10:21 AM +0700, Isianto Istiadi wrote:
Dear guys,
I'm searching the wake-up call script in wiki, found one, but I have no
idea how to use it. Can you give some direction how to install it?
Thanks
I presume you're talking about this wake up call script: 


Stick the following in cron:
* * * * * root /path/to/run_wakeups.sh
/path/to/run_wakeups.sh contains:
= cut ==
#!/bin/bash
PENDING=/tmp/wakeups
OUTGOING=/var/spool/asterisk/outgoing
SLEEP=5
TIME=$(/bin/date +%H%M)
for fn in $PENDING/$TIME.*.call
do
if test -r $fn
then
 /bin/mv -f $fn $OUTGOING/
 sleep $SLEEP
fi
done
= cut ==
The following is my wakeup.agi. Changes to the original version are: some 
debugging functionality (as I was troubleshooting an issue where it would 
read out the wrong time when the script tells you what time the wake up 
call was set to), and it also creates the /tmp/wakeups directory if it 
doesn't already exist. I suggest using the one on the voip-info.org page 
first, and if you decide to use my version then use at your own risk :)

= cut ==
#!/usr/bin/perl
use Asterisk::AGI;
use Date::Manip;
use strict;
#
# Settings:
my $pending_dir = '/tmp/wakeups';
unless (-d '/tmp/wakeups') {
   mkdir('/tmp/wakeups');
}
my $local_context = 'default';
# values for the call file:
my $maxretries = 60;
my $retrytime = 30;
my $waittime =  35;
my $debug = 1;
#my $application = 'MusicOnHold';
my $application = 'Playback';
my $data = 'wake-up';
my $callerid = 'Wakeup Call Service <297>';
#
my ($sec,$min,$hour,$mday,$mon,$year,$wday,$yday,$isdst) = localtime(time);
if ($debug) {
   my $log = '/tmp/wakeup.log';
   unlink($log);
   open (DBG,">>$log") or die "Cannot open debug file: $!";
   print DBG "\n" . "-" x 50 . "\n";
   print DBG "Logging started: " . join('/', $mday, $mon, $year) . " " 
. join(':', $hour, $min, $sec) . "\n";
   print DBG "-" x 50 . "\n";
}

my $agi = new Asterisk::AGI;
my %stuff = $agi->ReadParse;# MUST DO THIS! -- (add this to 
constructor!)

# this says "1 to create, 2 to confirm, 3 to cancel"
my $func = $agi->get_data('wakeup-menu', 2, 1);
exit if $func == -1;
my ($caller) = $stuff{callerid} =~ /<(\d+)>/;
if ($func == 1)
{
my $time = $agi->get_data('time', 15000, 4);
exit if $func == -1;
if ($time =~ /^(\d{2})(\d{2})$/)
{
 my $hour = $1 * 1;
 my $min = $2;
 print DBG 'HOUR entered: ' . $hour . "\n" if $debug;
 print DBG 'MINUTE entered: ' . $min . "\n" if $debug;
 if ($hour > 0 && $hour <= 12 && $min < 60)
 {
  my $time;
#   $agi->stream_file('pls-enter');
#   $agi->stream_file('digits/1');
#   $agi->stream_file('for');
#   $agi->stream_file('digits/a-m');
#   $agi->stream_file('or');
#   $agi->stream_file('digits/2');
#   $agi->stream_file('for');
#   my $ampm = $agi->get_data('digits/p-m', 15000, 1);
  my $ampm = $agi->get_data('am-or-pm', 15000, 1);
  exit if $ampm == -1;
  if ($ampm == 1)
  {
   $time = ParseDate(sprintf("%s:%02s AM", $hour, $min));
   print DBG 'TYPE entered: AM' . "\n" if $debug;
   print DBG '$time is set to: ' . $time . "\n" if $debug;
  }
  elsif ($ampm == 2)
  {
   $time = ParseDate(sprintf("%s:%02s PM", $hour, $min));
   print DBG 'TYPE entered: PM' . "\n" if $debug;
   print DBG '$time is set to: ' . $time . "\n" if $debug;
  }
  else
  {
   $agi->stream_file('vm-sorry');
  }
  if ($time)
  {
   my $h = UnixDate($time, "%I") * 1;
   my $m = UnixDate($time, "%M");
   my $a = UnixDate($time, "%p");
   foreach my $fn (<$pending_dir/*.$caller.call>)
   {
unlink $fn;
   }
   my $filename = sprintf("%s/%04s.%s.call", $pending_dir, UnixDate($time, 
"%H%M"), $caller);

   open(FILE, ">$filename");
   printf FILE q{#
Channel: Local/[EMAIL PROTECTED]
MaxRetries: %s
RetryTime: %s
WaitTime: %s
Application: %s
Data: %s
Callerid: %s
},
$caller, $local_context,
$maxretries,
$retrytime,
$waittime,
$application,
$data,
$callerid,
;
   close(FILE);
   # say "Your wakeup call"
   $agi->stream_file('has-been-set-to');
   print DBG 'UnixDate $time translates to ' . UnixDate($time, "%o") . 
"\n" if $debug;
   print DBG 'localtime (UnixDate $time) translates to ' . 
localtime(UnixDate($time, "%o")) . "\n" if $debug;
   $agi->exec('SayUnixTime', sprintf("%s||IMp", UnixDate($time, "%o")));

   $agi->stream_file('for');
   $agi->stream_file('extension');
   $agi->say_digits($caller);
   $agi->stream_file('auth-thankyou');
  }
 }
 else
 {
  $agi->stream_file('vm-sorry');
 }
}
else
{
 $agi->stream_file('vm-sorry');
}
}
elsif ($func == 2)
{
my ($fn) = <$pending_dir/*.$caller.call>;
if ($fn)
{
 my ($time) = $fn =~ /\/(\d{4})\.\d+\.call/;
 $time =~ s/(\d\d)(\d\d)/$1:$2/;
 $agi->stream_file('is-set-to');
 $agi->exec('SayUnixTime', sprintf("%s||IMp", UnixDate(ParseDate($time),
 "%s")));
}
else
{
 $agi->stream_file('is-not-set');
}
$agi->stream_file('auth-thankyou');
}
elsif ($func == 3)
{
foreach my $fn (<$pending_dir/*.

Re: [Asterisk-Users] Setting up your own menu like voice mail

[Gonzalo Servat]

On Sat, 2004-06-26 at 10:09 +0100, Dee Lowndes wrote:
> Hi all,
> 
>   Anyone know where/how I can setup my own menu to work like the
> voicemailmain menu.
> 
> e.g.
> 
> extension.conf
> 
> exten => 888,1,mymenusystem
> exten => 888,2,Goto(s,6)
> 
> then somewhere mymenusystem plays message and give options to goto exten
> 1, 2, 3 etc

http://www.voip-info.org/wiki-Asterisk+config+extensions.conf

The wiki has a *lot* of good info. Use it.

Regards,
Gonzalo

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Re: [Asterisk-Users] How to let users change Voice Mail password in Asterisk

[Gonzalo Servat]

On Thu, 2004-06-17 at 08:20 -0700, Deepak Malhotra wrote:
> Hello
>  
> Any idea or code on How to allow users to change their voice mail
> password over the Phone. 
> The only way io know is to change in voicemail.conf file and restart
> asterisk.

Try dialing your voicemail extension, enter your password, then press 0,
then press 4.  Follow the prompts.

HTH.

Regards,
Gonzalo

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RE: [Asterisk-Users] Asterisk-Users List Etiquette

[Gonzalo Servat]

On Thu, 2004-06-17 at 09:21 -0400, Troy Settle wrote:
<..snip..>
>
> However, my preference is for top posting.  The reason, is that in order to
> read my message here, you had to scroll through ~70 lines of previous
> discussion.  Stuff that you've /already/ read since you've been following
> this thread.
>
<..snip..}

Sorry to butt into this thread, but I think this is where you went
wrong.  There was absolutely no need to quote 70+ lines of text to say
what you had to say.  You're supposed to quote the relevant bits (as I
did with this email), not the entire thread.

Regards,
Gonzalo

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[Asterisk-Users] Choppy sound ONLY when a voicemail is left

[Gonzalo Servat]

Hi All,

Whenever a call comes in via the ISDN and somebody leaves a voicemail,
the sound file recorded is very choppy. If I actually take the call, the
sound is not choppy so it's obviously something to do with the Asterisk
box itself having to do the recording. Perhaps the sound card drivers?
I'm using the stock i810_audio (OSS) drivers on Fedora Core 1.

If I call from a local VoIP client to the Asterisk box and leave a
voicemail, the sound is NOT choppy. 

No errors on the Asterisk console.

Thanks in advance.

Regards,
Gonzalo

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Re: [Asterisk-Users] Automating calls

[Gonzalo Servat]

On Thu, 2004-06-10 at 16:27 +0100, Simon wrote:
> Hello
> 
> I have heard that i can put a file in a certain directory to get * to
> initiate a call.
> 
> Is this true ? if so where would i look ?

The rumours are true! You would look in the ever-so-helpful Wiki:

 http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out

HTH.

Best regards,
Gonzalo

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[Asterisk-Users] Echo problems using AVM Fritz!PCI Card

[Gonzalo Servat]

Hi All,

I'm running the latest asterisk CVS code (from experimental), and
hardware-wise an AVM Fritz!PCI Card, together with chan_capi 0.3.3. 
The problem is that any inbound/outbound calls result in echo on MY end
(the asterisk end). I've played with the echo settings in capi.conf
(mainly turning on echocancel and echosquelch, also tried playing with
rxgain/txgain) to no avail. The only setting that has helped (somewhat)
so far is enabling echosquelch. The echo disappears but a new problem
arises. When the person on the other end starts to talk, the first bit
is chopped off, and the last bit (before they go quiet) so it almost
sounds as though it's doing voice detection and transmitting only when
it detects voice. Also, if the other person is talking and i start to
talk, they get cut off immediately so this isn't a practical workaround.

Any help will be muchly appreciated.

Regards,
Gonzalo

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Re: [Asterisk-Users] Can I do this ...

[Gonzalo Servat]

On Wed, 2004-05-26 at 14:42 +1000, Simon Brown wrote:
> Can I do this with * ???
> 
> S,1,answer call
> S,2,play "thanks for calling, we'll be with you soon"
> S,3,play music while caller waits and ring nominated extensions at same time
> S,101,if not answered go to voicemail
> 
> I can't find a way to play music and ring extensions at the same time.
> 
> Any help would be greatly appreciated.

Hi Simon,

Look into the "m" option for the Dial command:

'm' — provide hold music to the calling party until answered.

.. quoted from:

http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Dial

HTH.

Regards,
Gonzalo

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Re: [Asterisk-Users] NetJet and RAS

[Gonzalo Servat]

On Mon, 2004-05-24 at 09:57 +1000, Andrew Yager wrote:
> Thanks! That's good to know. Please excuse my ignorance - if we have 
> two telstra ISDN2 lines, which card should I get?

A somewhat reasonably priced ISDN card that works with Asterisk and is
sold in Australia is the AVM Fritz:

   http://www.avm.de/en/partners/distributors/AUS/index.html

Hope this helps.

Regards,
Gonzalo

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Re: [Asterisk-Users] compile error

[Gonzalo Servat]

On Mon, 2004-02-02 at 23:19, jjj3 jjj3 wrote:
> I'm trying to compile the last * CVS version and I got this error:
> 
> gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes 
> -Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE 
>   -O6 -march=i686  -DZAPTEL_OPTIMIZATIONS 
> -DASTERISK_VERSION=\"CVS-09/10/03-18:47:18\" -DINSTALL_PREFIX=\"\" 
> -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" 
> -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" 
> -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" 
> -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" 
> -DASTMODDIR=\"/usr/lib/asterisk/modules\" 
> -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN 
> -DNEW_PRI_HANGUP-DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC   -c -o 
> res_crypto.o res_crypto.c
> gcc -shared -Xlinker -x -o res_crypto.so res_crypto.o -lssl -lcrypto
> /usr/bin/ld: cannot find -lssl
> collect2: ld returned 1 exit status
> make[1]: *** [res_crypto.so] Error 1
> make[1]: Leaving directory `/usr/src/asterisk/res'
> make: *** [subdirs] Error 1
> 
> 
> The openssl is already installed and all the .h files are in 
> /usr/include/openssl.
> 
> What does it can be?

Check /etc/ld.so.conf, ensure the path to your SSL libraries is in
there. If it's not, add it and run "ldconfig".

HTH.

Regards,
Gonzalo

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Re: [Asterisk-Users] Compile Problem

[Gonzalo Servat]

Hi,

On Tue, 2003-12-23 at 12:12, [EMAIL PROTECTED] wrote:
>  
[...]

> I am trying to compile the asterisk and if fails at the end
> on:
>  
> make[1]: Entering directory `/usr/src/asterisk-0.5.0/pbx'
> gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o
> `gtk-config --libs gthread`
> 
> /usr/lib/gcc-lib/i486-slackware-linux/3.3.2/../../../../i486-slackware-linux/bin/ld: 
> cannot find -lXext
> collect2: ld returned 1 exit status
> make[1]: *** [pbx_gtkconsole.so] Error 1
> make[1]: Leaving directory `/usr/src/asterisk-0.5.0/pbx'
> make: *** [subdirs] Error 1
>  
> Anyone know what is wrong? Linpri and zaptel compiled just
> fine. This is linux slackware 2.4.23 all the latest from 9.1
> slackware distrib fresh system install.

I'm not sure where you'd find the following file in Slackware, but in
RedHat:

  /usr/X11R6/lib/libXext.so.6

.. is part of the XFree86-libs RPM. Find the corresponding tgz, install
it and then try to compile again. It should get past that error.

HTH,
Gonzalo

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Re: [Asterisk-Users] AVM ISDN Fritz!Card USB works

[Gonzalo Servat]

On Tue, 2003-12-16 at 10:34, Michiel Betel wrote:
> Is case anyone wants to know... The Fritz! USB ISDN box works fine with 
> Asterisk!
> I'm running CAPI 0.3.0 and love it, because the mini ITX server I have 
> only takes one PCI slot which is now filled with a 4 port Digium card.

Hi Michiel,

Thanks for the report. My local distributor sell the AVM Fritz!Card USB
for only $5 more than the PCI one so it's well worth considering if it
works perfectly with Asterisk.

Regards,
Gonzalo

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RE: [Asterisk-Users] FXO Cards in Australia

[Gonzalo Servat]

I've spoken to Guy. I suggested he takes a look at:

  http://www.junghanns.net

.. for the CAPI Channel driver, but after speaking to a few more people
I began to understand how they all link together and the CAPI channel
driver is not going to help. He knows about the echo problems now so the
ball is in his court.

As I understand it, there needs to be some Linux CAPI drivers written
for the NetJet otherwise the only way to use the card is with
isdn4linux. Correct?

Regards,
Gonzalo


On Tue, 2003-11-18 at 14:03, Matthew Enger wrote:
> Hello,
> 
> Let us know how you go, be better if one person contacts him then all of
> us:)
> 
> Thanks,
>   Matthew Enger
>   [EMAIL PROTECTED]
> 
> On Tue, 2003-11-18 at 01:10, Gonzalo Servat wrote:
> > I'll be speaking to Guy tomorrow about this. Guy is certainly a helpful
> > & friendly guy and I'm sure he'll be keen to hear about these echo
> > problems.
> > 
> > Regards,
> > Gonzalo
> > 
> > 
> > On Tue, 2003-11-18 at 00:48, Bryan Nolen wrote:
> > > Re: these problems with the NetJet Cards: have people spoken with Traverse
> > > about them? I have found them to be most helpful with any problems (mainly
> > > with the Pulsar PCI ADSL cards)
> > > 
> > > Try talking to [EMAIL PROTECTED] ?
> > > 
> > > -Bryan
> > > 
> > > > -Original Message-
> > > > From: [EMAIL PROTECTED] 
> > > > [mailto:[EMAIL PROTECTED] On Behalf Of 
> > > > Gonzalo Servat
> > > > Sent: Tuesday, 18 November 2003 12:18 AM
> > > > To: [EMAIL PROTECTED]
> > > > Subject: RE: [Asterisk-Users] FXO Cards in Australia
> > > > 
> > > > 
> > > > On Mon, 2003-11-17 at 23:53, Adam Goryachev wrote:
> > > > > Yes, echo problems do still exist, I would suggest testing it before
> > > > > going live.
> > > > 
> > > > Yeah, so I've heard.
> > > > 
> > > > > A couple of points to note:
> > > > > 1) Using soft phones seems to compound the issue
> > > > 
> > > > So the echo problems are not so bad when using software phones?
> > > > 
> > > > > 2) A faster CPU seems to help (I upgraded from a PII300 to a Athlon
> > > > > 2200)
> > > > > 3) When dialling in/out over the ISDN DTMF won't work (at least I
> > > > > haven't seen the patch which purportedly allows it to work) 
> > > > when you use
> > > > > the isdn4linux patch.
> > > > 
> > > > This is specific to the NetJet card once again, right? Time to go
> > > > hunting for the patch...
> > > > 
> > > > > 4) Without the above kernel patch you will hear DTMF tones 
> > > > instead of
> > > > > the other persons voice when they talk. They don't hear the tones or
> > > > > notice anything wrong.
> > > > 
> > > > Hmm, not good. Since we want to run a small IVR the DTMF 
> > > > tones are kinda
> > > > needed.
> > > > 
> > > > > In short, if you can live with the above problems, then you 
> > > > can get away
> > > > > with it, from what I know now, I would suggest getting a chan_capi
> > > > > capable device, though I haven't tried that yet.
> > > > 
> > > > The NetJet is supposedly CAPI capable. Have you tried installing this?
> > > > --> http://www.junghanns.net/asterisk/page1.html
> > > > 
> > > > > I am about to switch from a netjet card to a TE4xxP card as soon as
> > > > > possible, I have a OnRamp 10 being installed tomorrow. This 
> > > > is largely
> > > > > to increase the number of incoming lines, but partly to resolve the
> > > > > above issues, and also partly to try to resolve long 
> > > > running reliability
> > > > > issues which may in fact be related to the TDM400P anyway. 
> > > > In which case
> > > > > I will be looking for a T1 channel bank some time soon :(
> > > > 
> > > > Argh, the fun never stops :)
> > > > 
> > > > > PS, I have a brand new Traverse Netjet card available (it 
> > > > was to be used
> > > > > for a dial-up ISDN internet account) which is no longer needed.
> > > > 
> > > > How much do you want for it? If you can confirm whether the 
> > > > capi channel
> > > > driver works with it and reduces the echo problem, I'll be interested.
> > > > 
> > > > Thanks for your help.
> > > > 
> > > > Regards,
> > > > Gonzalo
> > > > 
> > > > ___
> > > > Asterisk-Users mailing list
> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > 
> > > 
> > > 
> > > ___
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Willoughby North, NSW, 2068

Phone: +61 (02) 9499 2452
Fax:   +61 (02) 9499 2618
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RE: [Asterisk-Users] FXO Cards in Australia

[Gonzalo Servat]

I'll be speaking to Guy tomorrow about this. Guy is certainly a helpful
& friendly guy and I'm sure he'll be keen to hear about these echo
problems.

Regards,
Gonzalo


On Tue, 2003-11-18 at 00:48, Bryan Nolen wrote:
> Re: these problems with the NetJet Cards: have people spoken with Traverse
> about them? I have found them to be most helpful with any problems (mainly
> with the Pulsar PCI ADSL cards)
> 
> Try talking to [EMAIL PROTECTED] ?
> 
> -Bryan
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] On Behalf Of 
> > Gonzalo Servat
> > Sent: Tuesday, 18 November 2003 12:18 AM
> > To: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] FXO Cards in Australia
> > 
> > 
> > On Mon, 2003-11-17 at 23:53, Adam Goryachev wrote:
> > > Yes, echo problems do still exist, I would suggest testing it before
> > > going live.
> > 
> > Yeah, so I've heard.
> > 
> > > A couple of points to note:
> > > 1) Using soft phones seems to compound the issue
> > 
> > So the echo problems are not so bad when using software phones?
> > 
> > > 2) A faster CPU seems to help (I upgraded from a PII300 to a Athlon
> > > 2200)
> > > 3) When dialling in/out over the ISDN DTMF won't work (at least I
> > > haven't seen the patch which purportedly allows it to work) 
> > when you use
> > > the isdn4linux patch.
> > 
> > This is specific to the NetJet card once again, right? Time to go
> > hunting for the patch...
> > 
> > > 4) Without the above kernel patch you will hear DTMF tones 
> > instead of
> > > the other persons voice when they talk. They don't hear the tones or
> > > notice anything wrong.
> > 
> > Hmm, not good. Since we want to run a small IVR the DTMF 
> > tones are kinda
> > needed.
> > 
> > > In short, if you can live with the above problems, then you 
> > can get away
> > > with it, from what I know now, I would suggest getting a chan_capi
> > > capable device, though I haven't tried that yet.
> > 
> > The NetJet is supposedly CAPI capable. Have you tried installing this?
> > --> http://www.junghanns.net/asterisk/page1.html
> > 
> > > I am about to switch from a netjet card to a TE4xxP card as soon as
> > > possible, I have a OnRamp 10 being installed tomorrow. This 
> > is largely
> > > to increase the number of incoming lines, but partly to resolve the
> > > above issues, and also partly to try to resolve long 
> > running reliability
> > > issues which may in fact be related to the TDM400P anyway. 
> > In which case
> > > I will be looking for a T1 channel bank some time soon :(
> > 
> > Argh, the fun never stops :)
> > 
> > > PS, I have a brand new Traverse Netjet card available (it 
> > was to be used
> > > for a dial-up ISDN internet account) which is no longer needed.
> > 
> > How much do you want for it? If you can confirm whether the 
> > capi channel
> > driver works with it and reduces the echo problem, I'll be interested.
> > 
> > Thanks for your help.
> > 
> > Regards,
> > Gonzalo
> > 
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
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=
WEBTASTIC
ABN 30 087 960 619
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Phone: +61 (02) 9499 2452
Fax:   +61 (02) 9499 2618
Email: [EMAIL PROTECTED]
Web:   http://www.webtastic.com.au
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RE: [Asterisk-Users] FXO Cards in Australia

[Gonzalo Servat]

On Mon, 2003-11-17 at 23:53, Adam Goryachev wrote:
> Yes, echo problems do still exist, I would suggest testing it before
> going live.

Yeah, so I've heard.

> A couple of points to note:
> 1) Using soft phones seems to compound the issue

So the echo problems are not so bad when using software phones?

> 2) A faster CPU seems to help (I upgraded from a PII300 to a Athlon
> 2200)
> 3) When dialling in/out over the ISDN DTMF won't work (at least I
> haven't seen the patch which purportedly allows it to work) when you use
> the isdn4linux patch.

This is specific to the NetJet card once again, right? Time to go
hunting for the patch...

> 4) Without the above kernel patch you will hear DTMF tones instead of
> the other persons voice when they talk. They don't hear the tones or
> notice anything wrong.

Hmm, not good. Since we want to run a small IVR the DTMF tones are kinda
needed.

> In short, if you can live with the above problems, then you can get away
> with it, from what I know now, I would suggest getting a chan_capi
> capable device, though I haven't tried that yet.

The NetJet is supposedly CAPI capable. Have you tried installing this?
--> http://www.junghanns.net/asterisk/page1.html

> I am about to switch from a netjet card to a TE4xxP card as soon as
> possible, I have a OnRamp 10 being installed tomorrow. This is largely
> to increase the number of incoming lines, but partly to resolve the
> above issues, and also partly to try to resolve long running reliability
> issues which may in fact be related to the TDM400P anyway. In which case
> I will be looking for a T1 channel bank some time soon :(

Argh, the fun never stops :)

> PS, I have a brand new Traverse Netjet card available (it was to be used
> for a dial-up ISDN internet account) which is no longer needed.

How much do you want for it? If you can confirm whether the capi channel
driver works with it and reduces the echo problem, I'll be interested.

Thanks for your help.

Regards,
Gonzalo

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Re: [Asterisk-Users] FXO Cards in Australia

[Gonzalo Servat]

On Mon, 2003-11-17 at 16:00, Anthony Wood wrote:
> ISDN (telstra Onramp 2) is very similar in price to standard telstra lines.
> The only problem is you can't have ADSL & ISDN on the same line.
> 
> We upgraded from 2 analogue lines to 2 digital (i.e. 4 channels) for $250.

I was a bit turned off by the $300+ installation cost. I just rang
Telstra and its infact $190 if you already have a telephone line, which
I do. Awesome!

How come you 4 channels if you only have 2 digital lines? I thought it
was one channel per line. I was told by the Telstra rep that I need a
OnRamp2 which is 2 channels, 2 lines.

> But they Telstra'd up the installation so we asked for (and got) the $250 waived.

Typical (about Telstra'ing the installation, not the setup fee
discount!)

> It's worth thinking about it because of the Advantages of Digital signalling when
> using voice:
> 
> Know which number was dialed
> Know callerid early
> Know when the other end has hung up
> Better voice quality
> 
> Using Analogue with Asterisk seems to be filled with Kludges to detect hangups,
> busy, etc.  With ISDN, the exchange does that for you.

Yeah, we're now looking at it again. Local calls are pretty cheap too as
long as you don't talk for too long.

You mentioned echo problems with the NetJet cards. Is this still the
case or was it last time you tried that it that had echo problems? I did
a Google search and didn't find much on the echo problems with them.

Thanks again for the good info.

Regards,
Gonzalo

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[Asterisk-Users] Distinctive Ring

[Gonzalo Servat]

Hi All,

I was wondering what the status of distinctive ring support in Asterisk
is? I had a google search & read and Mark Spencer wrote some support for
it.

Is distinctive ring different in every country or is it pretty standard?

And for my final question, does the Wildcard FXO card support
distinctive ring?

Essentially what I'm trying to do is route incoming calls with ring #1
to, say, 2 SIP clients and incoming calls with ring #2 to 1 SIP client,
but somehow label incoming calls so the SIP client knows whether the
call was for ring #1 or ring #2.

Thanks in advance.

Regards,
Gonzalo

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Re: [Asterisk-Users] FXO Cards in Australia

[Gonzalo Servat]

On Mon, 2003-11-17 at 12:20, Anthony Wood wrote:

> I have spoken to a number of Australian users who are successfully using:
> 
> X100P
> NetJet (echo issues)
> AVM Fritz!Card
> 
> I hope to add myself to their number shortly, since we have recieved our Fritz!es
> 
> Also [EMAIL PROTECTED] seems to be having some success with the VoiceTronix 
> openline4.
> 
> All these cards are legal except the X100P.

Thanks very much Anthony. VoiceTronix cards are a little out of my
budget, the NatJet & AVM cards are for ISDN (and we need standard
analogue).

> PS: You are a SLUG member, no?

I'm a SLUG active mailing list user, not a financial member - yet. :)

Regards,
Gonzalo

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[Asterisk-Users] FXO Cards in Australia

[Gonzalo Servat]

Hi All,

This topic has come up before in the Asterisk mailing list many times,
so I know that a lot of people have given up in waiting for a FXO card
to be approved by the Australian telecommunications authority. My
question is: all legalities aside - is anyone using a FXO card in
Australia successfully?

Thanks in advance.

Regards,
Gonzalo

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