Re: [asterisk-users] What do you use? .conf or AEL?

[Louis-David Mitterrand]

On Tue, Feb 10, 2009 at 01:56:16PM -0800, Mik Cheez wrote:
> I use them both; my legacy dialplan is all .conf and new stuff is .ael. 
>   I find AEL to be the better option when jumping around, but that's 
> just my opinion.

But isn't AEL just converted into .conf language anyway? Or has this
evolved with 1.4.x ?

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Re: [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions

[Steve Totaro]

I go with what I know is solid, Adit or Adtran channel banks and T1
ports.  I personally like Adtran but that is just preference.

The Xorcom device is USB correct?  I just have a personal block on
putting anything mission critical on a USB port or hub.  I would have
to lab it up and really test it before feeling comfortable.  That
being said, I do not here people complaining or even really asking
questions and I know the devices are out there, so maybe it is time to
check it out.

Citel was a total flop for trying to replace a Definty G3.  Nothing
but problems, loud pops, screeching, hissing, dropped calls.  Citel
tech support was one step up from SwitchVox if you have a "real"
problem that doesn't fit in a script, downright hostile.  Once bitten,
twice shy, but then again, it was replacing a digital PBX, not analog
and that was many years ago (Citel, not SwitchVox.  SwitchVox was good
many years ago)

I had great success with Quintum Tenor AX boxen but that is almost
certainly overkill for your application.  They are basically high
density ATAs with more features than you can believe.  Great for
configuring and shipping to a remote location.

Just my experience and thoughts.

Thanks,
Steve T

On 2/12/09, Stephen Davies  wrote:
> Everybody is talking about other products.  But yes, the Xorcom will
> handle all ports active, supports a high density connector at the
> back, looks just like standard Zap/Dahdi ports to Asterisk, rack
> mounts nicely and much less $$ than the other solutions.
>
> Steve
>
> On 2/10/09, Erick Perez  wrote:
>> Hi, I am looking to connect 66 analog phones to an asterisk box. I was
>> thinking of a Xorcom astribank 32port (2 of them and another 8 port).
>> this is because the phones have no near connection to an ip network,
>> so replacing the phones in favor of  voip phones+network cabling is
>> kinda out of the question.
>>
>> In your experience, will these units support all the phones talking at
>> the same time with other units on the astribank, as well as to the
>> pbx, pstn, etc? The asterisk pbx will be a server-class Hp Proliant
>> unit (potentially a dl320). i must make sure the astribanks will not
>> die when fully utilized.
>>
>> other hardware suggestions for this task will be nice.
>>
>> thanks,
>>
>>
>> --
>> 
>> Erick
>> 

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Re: [asterisk-users] Invalid Extension

[Dovid Bender]

Do you have extension  ontext 059*162*178*122*78600051 in your 
extensions.conf under the default context ?

- Original Message - 
From: "Philipp Kempgen" 
To: "Asterisk Users" 
Sent: Monday, February 02, 2009 10:40 PM
Subject: Re: [asterisk-users] Invalid Extension


David @ULC schrieb:
> vicidialnow*CLI>

> -- Executing AGI("SIP/66.54.140.46-b7800468",
> "agi-VDAD_ALL_inbound.agi|CIDL
>
> OOKUPRC-LB-SALESLINE-936998-Closer-park--999-1--
>
>  ---TESTCAMP") in new stack

> Feb  2 14:53:09 NOTICE[18377]: chan_local.c:526 local_alloc: No such
> extension/c
> ontext 059*162*178*122*78600...@default creating local channel
> Feb  2 14:53:09 NOTICE[18377]: channel.c:2514 __ast_request_and_dial: 
> Unable
> to
>request channel Local/059*162*178*122*78600...@default

Ugh.

> When I call my DID, it get answered at my end but at other end , customer
> hears Its an INVALID extension and line get hang up.

> What could be the reason ?

Could be a bug in ViciDial or in your specific setup.

http://astguiclient.sourceforge.net/vicidial.html
http://www.eflo.net/vicidial.php


   Philipp Kempgen

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Re: [asterisk-users] How to make the Asterisk-GUI workwithDAHDI..please??

[Dovid Bender]

What is Zap mirroring ?

- Original Message - 
From: "Danny Nicholas" 
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 

Sent: Monday, February 09, 2009 10:09 PM
Subject: Re: [asterisk-users] How to make the Asterisk-GUI 
workwithDAHDI..please??


> Assuming you're still in the 1.4 set, enable the Zap "mirroring".  I'm not
> sure the fine folks at Digium have accounted for DAHDI in the current
> interfaces (but I'm probably wrong).
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Enrique
> Sent: Monday, February 09, 2009 12:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] How to make the Asterisk-GUI work
> withDAHDI..please??
>
> How to make the Asterisk-GUI work with DAHDI..please??
>
>
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Re: [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions

[Stephen Davies]

Everybody is talking about other products.  But yes, the Xorcom will
handle all ports active, supports a high density connector at the
back, looks just like standard Zap/Dahdi ports to Asterisk, rack
mounts nicely and much less $$ than the other solutions.

Steve

On 2/10/09, Erick Perez  wrote:
> Hi, I am looking to connect 66 analog phones to an asterisk box. I was
> thinking of a Xorcom astribank 32port (2 of them and another 8 port).
> this is because the phones have no near connection to an ip network,
> so replacing the phones in favor of  voip phones+network cabling is
> kinda out of the question.
>
> In your experience, will these units support all the phones talking at
> the same time with other units on the astribank, as well as to the
> pbx, pstn, etc? The asterisk pbx will be a server-class Hp Proliant
> unit (potentially a dl320). i must make sure the astribanks will not
> die when fully utilized.
>
> other hardware suggestions for this task will be nice.
>
> thanks,
>
>
> --
> 
> Erick
> 
>
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Re: [asterisk-users] Strange dialplan matching issue

[Stephen Davies]

Hi,

As others have mentioned, the 'n' is a pattern char.

I have a system that uses similar tricks to yours.  What I did about
this issue was to change the pattern match chars to be upper case
only.  Drop me a line if you want the patch.

Regards,
Steve

On 2/12/09, Chris Bagnall  wrote:
> Greetings list,
>
> Wondering if anyone has come across this strange dialplan pattern matching
> issue before:
>
> I have a context defined as follows (the plus simply implies it follows on
> from an existing context in another #include - which, yes, has been included
> first):
> [privatedundi](+)
> exten => _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)
>
> When dialling hilton-202 from another box via IAX2, I get:
> NOTICE[3727]: chan_iax2.c:8085 socket_process: Rejected connect attempt from
> , request 'hilton-...@privatedundi' does not exist
>
> Changing the context to read as follows solves the problem immediately:
> [privatedundi](+)
> exten => hilton-201,1,Goto(hilton,${EXTEN:7},1)
> exten => hilton-202,1,Goto(hilton,${EXTEN:7},1)
> exten => hilton-203,1,Goto(hilton,${EXTEN:7},1)
>
> Dialling hilton-202 now works every time.
>
> The *really* strange thing is that I have a number of similar pattern
> matches, and all the others work fine, it's just this one that doesn't.
>
> The box in question is running 1.4.22, but I have had a similar issue in the
> past with a 1.2 box, so it does not appear to be version specific.
>
> Any thoughts?
>
> TIA.
>
> Regards,
>
> Chris
>
>
>
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Re: [asterisk-users] Strange dialplan matching issue

[Nabeel Jafferali]

BTW I just did some quick experimentation. Example 1 did not work, example 2 
did work. So that's a solution to your issue.

Example 1:

exten => 999,1,Goto(nabeel,1)
exten => _nabeel,1,Goto(800,1)

Example 2:

exten => 999,1,Goto(nabeel,1)
exten => _[n]abeel,1,Goto(800,1)

--
Nabeel Jafferali
X2 Networks


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nabeel Jafferali
Sent: February-11-09 11:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Strange dialplan matching issue

Asterisk is looking for hilto[1-9]-2[0-9][0-9], if you know what I mean?

--
Nabeel Jafferali
X2 Networks

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: February-11-09 8:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Strange dialplan matching issue

Greetings list,

Wondering if anyone has come across this strange dialplan pattern matching 
issue before:

I have a context defined as follows (the plus simply implies it follows on from 
an existing context in another #include - which, yes, has been included first):
[privatedundi](+)
exten => _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)

When dialling hilton-202 from another box via IAX2, I get:
NOTICE[3727]: chan_iax2.c:8085 socket_process: Rejected connect attempt from 
, request 'hilton-...@privatedundi' does not exist

Changing the context to read as follows solves the problem immediately:
[privatedundi](+)
exten => hilton-201,1,Goto(hilton,${EXTEN:7},1)
exten => hilton-202,1,Goto(hilton,${EXTEN:7},1)
exten => hilton-203,1,Goto(hilton,${EXTEN:7},1)

Dialling hilton-202 now works every time.

The *really* strange thing is that I have a number of similar pattern matches, 
and all the others work fine, it's just this one that doesn't.

The box in question is running 1.4.22, but I have had a similar issue in the 
past with a 1.2 box, so it does not appear to be version specific.

Any thoughts?

TIA.

Regards,

Chris



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Re: [asterisk-users] Strange dialplan matching issue

[Nabeel Jafferali]

Asterisk is looking for hilto[1-9]-2[0-9][0-9], if you know what I mean?

--
Nabeel Jafferali
X2 Networks

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: February-11-09 8:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Strange dialplan matching issue

Greetings list,

Wondering if anyone has come across this strange dialplan pattern matching 
issue before:

I have a context defined as follows (the plus simply implies it follows on from 
an existing context in another #include - which, yes, has been included first):
[privatedundi](+)
exten => _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)

When dialling hilton-202 from another box via IAX2, I get:
NOTICE[3727]: chan_iax2.c:8085 socket_process: Rejected connect attempt from 
, request 'hilton-...@privatedundi' does not exist

Changing the context to read as follows solves the problem immediately:
[privatedundi](+)
exten => hilton-201,1,Goto(hilton,${EXTEN:7},1)
exten => hilton-202,1,Goto(hilton,${EXTEN:7},1)
exten => hilton-203,1,Goto(hilton,${EXTEN:7},1)

Dialling hilton-202 now works every time.

The *really* strange thing is that I have a number of similar pattern matches, 
and all the others work fine, it's just this one that doesn't.

The box in question is running 1.4.22, but I have had a similar issue in the 
past with a 1.2 box, so it does not appear to be version specific.

Any thoughts?

TIA.

Regards,

Chris



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[asterisk-users] IDAP T1

[Lee, John (Sydney)]

What is IDAP-T1?  How different is it from normal T1?
Any chance I can get it to work with Digium 412P and Asterisk 1.4.* ?
If yes, what would zaptel.cof look like?  
Any difference from normal T1 config?

Thanks.

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Re: [asterisk-users] Strange dialplan matching issue

[Jose P. Espinal]

On extensions.conf.sample I see this:

; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
;   . - wildcard, matches anything remaining (e.g. _9011. matches
;   anything starting with 9011 excluding 9011 itself)
;   ! - wildcard, causes the matching process to complete as soon as
;   it can unambiguously determine that no other matches are possible

Maybe after using '_' Asterisk is waiting for one of the above pattern 
matching characters.

a. The 'hilton-' part of your dialplan might not being considered valid, 
and Asterisk *might* be trying to match the 'XX' part LITERALLY, and 
would be trying to reach extension '2XX'

exten => _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)


b. then, in:
exten => hilton-203,1,Goto(hilton,${EXTEN:7},1)

You provided the real extension number (after you take out the fist 7 
digits).

So, Asterisk reaches '203', etc.



Try only using valid pattern matching characters in your dialplan to see 
if it works.



Chris Bagnall wrote:
> Greetings list,
> 
> Wondering if anyone has come across this strange dialplan pattern matching 
> issue before:
> 
> I have a context defined as follows (the plus simply implies it follows on 
> from an existing context in another #include - which, yes, has been included 
> first):
> [privatedundi](+)
> exten => _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)
> 
> When dialling hilton-202 from another box via IAX2, I get:
> NOTICE[3727]: chan_iax2.c:8085 socket_process: Rejected connect attempt from 
> , request 'hilton-...@privatedundi' does not exist
> 
> Changing the context to read as follows solves the problem immediately:
> [privatedundi](+)
> exten => hilton-201,1,Goto(hilton,${EXTEN:7},1)
> exten => hilton-202,1,Goto(hilton,${EXTEN:7},1)
> exten => hilton-203,1,Goto(hilton,${EXTEN:7},1)
> 
> Dialling hilton-202 now works every time.
> 
> The *really* strange thing is that I have a number of similar pattern 
> matches, and all the others work fine, it's just this one that doesn't.
> 
> The box in question is running 1.4.22, but I have had a similar issue in the 
> past with a 1.2 box, so it does not appear to be version specific.
> 
> Any thoughts?
> 
> TIA.
> 
> Regards,
> 
> Chris
> 
> 
> 
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-- 
Jose P. Espinal
http://www.eSlackware.com
IRC: [OFTC|FreeNode]
Khratos @ #slackware | #asterisk/-doc/-bugs


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[asterisk-users] Strange dialplan matching issue

[Chris Bagnall]

Greetings list,

Wondering if anyone has come across this strange dialplan pattern matching 
issue before:

I have a context defined as follows (the plus simply implies it follows on from 
an existing context in another #include - which, yes, has been included first):
[privatedundi](+)
exten => _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)

When dialling hilton-202 from another box via IAX2, I get:
NOTICE[3727]: chan_iax2.c:8085 socket_process: Rejected connect attempt from 
, request 'hilton-...@privatedundi' does not exist

Changing the context to read as follows solves the problem immediately:
[privatedundi](+)
exten => hilton-201,1,Goto(hilton,${EXTEN:7},1)
exten => hilton-202,1,Goto(hilton,${EXTEN:7},1)
exten => hilton-203,1,Goto(hilton,${EXTEN:7},1)

Dialling hilton-202 now works every time.

The *really* strange thing is that I have a number of similar pattern matches, 
and all the others work fine, it's just this one that doesn't.

The box in question is running 1.4.22, but I have had a similar issue in the 
past with a 1.2 box, so it does not appear to be version specific.

Any thoughts?

TIA.

Regards,

Chris



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Re: [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions

[C F]

I use channel banks.
I like the Adit 600s. For your configuration you'd need 2 Adit 600
with 9 FXS cards and 3 T1 ports in the Asterisk box.
One side advantage is that you can mount the Adit 600 right next to
your cat3 wiring, then just use an existing cat3 to the Asterisk box.
I have seen lots of older buildings where the data stuff (servers,
switches etc.) are no where close to the Voice (cat3 wiring, PBX etc.)
and no cat5s between those 2 areas.

Also the Adit 600 is carrier grade equipment.


On Wed, Feb 11, 2009 at 10:28 AM, Heath Roberts  wrote:
> On Tue, Feb 10, 2009 at 12:23 PM, Erick Perez  wrote:
>> Hi, I am looking to connect 66 analog phones to an asterisk box.
>
>> other hardware suggestions for this task will be nice.
>
> Citel makes a box:
> http://www.citel.com/Products/Portico.asp
>
> If you're converting from Definity, Norstar, p-phone (Nortel Centrex),
> Meridian, or NEC, a google search found them for under $500 (for 24
> ports):
> http://www.neobits.com/digital_handset_converters_s263.html
>
> I've not used the Citel product, but have heard good things about them.
>
> Depending on what it is, you could also just leave the existing switch
> in place and connect with a PRI interface--let your PBX think the
> asterisk box is the telco.
> --
> Heath Roberts
> htrobe...@gmail.com
>
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[asterisk-users] Problem with AMI action userevent

[Jim Dickenson]

When action_userevent was rewritten to not use local variables there was an
omission. The buffer is not initialized each time so things keep getting
appended to the buffer.

In addition I would find it useful to have the ping action return the
timestamp. That way I do not have to have timestamp event enabled and when I
want to check the time sync with the asterisk server I can just to a ping
and get a timestamp back.

Here is a patch the current svn version of manager.c from the 1.6.0 branch


--- manager.c2009-02-11 15:16:16.0 -0800
+++ manager.c.mine2009-02-11 15:10:52.0 -0800
@@ -1089,9 +1089,13 @@
 
 static int action_ping(struct mansession *s, const struct message *m)
 {
+struct timeval now;
+
+now = ast_tvnow();
+
 astman_append(s, "Response: Success\r\n"
-"Ping: Pong\r\n"
-"\r\n");
+"Ping: Pong Timestamp: %ld.%06lu\r\n"
+"\r\n", now.tv_sec, (unsigned long) now.tv_usec);
 return 0;
 }
 
@@ -2462,6 +2466,8 @@
 const char *event = astman_get_header(m, "UserEvent");
 struct ast_str *body = ast_str_thread_get(&userevent_buf, 16);
 int x;
+/* better init stuff so  ast_str_append can be called */
+body->used = 0;
 for (x = 0; x < m->hdrcount; x++) {
 if (strncasecmp("UserEvent:", m->headers[x], strlen("UserEvent:")))
{
 ast_str_append(&body, 0, "%s\r\n", m->headers[x]);

-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/


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Re: [asterisk-users] call picking and transfers

[Philipp Kempgen]

Jeff LaCoursiere schrieb:
> Working on some niche requests from one of my hotel clients.  asterisk 
> 1.4.20-1 on CentOS, Polycom 501s.
> 
> The first request is for the Polycom's screen to show the CID of the 
> inbound caller when a call pick is executed, so the picker knows if the 
> call is internal or external.  I have already "worked around" this issue 
> by using ALERT info to give seperate ring tones for outside and inside, 
> but they are used to their old Nortel switch which apparently showed the 
> CID immediately after the pick, and they then knew how to answer the 
> phone.
> 
> The second is to show CID information on the screen when a call has been 
> answered by the front desk, then a blind transfer sent to an internal 
> phone.  Today they simply see "Front Desk" and there is no indication of 
> who the actual caller is to distinguish internal staff, internal guest 
> room, or outside caller.
> 
> Has anyone attacked these things with Polycom that might share their 
> approach?

These bugs cover the functionality you need I guess:

http://bugs.digium.com/view.php?id=5014
http://bugs.digium.com/view.php?id=13827
http://bugs.digium.com/view.php?id=8824

However none of the patches are likely to be merged into 1.4.

   Philipp Kempgen

-- 
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[asterisk-users] call picking and transfers

[Jeff LaCoursiere]

Howdy,

Working on some niche requests from one of my hotel clients.  asterisk 
1.4.20-1 on CentOS, Polycom 501s.

The first request is for the Polycom's screen to show the CID of the 
inbound caller when a call pick is executed, so the picker knows if the 
call is internal or external.  I have already "worked around" this issue 
by using ALERT info to give seperate ring tones for outside and inside, 
but they are used to their old Nortel switch which apparently showed the 
CID immediately after the pick, and they then knew how to answer the 
phone.

The second is to show CID information on the screen when a call has been 
answered by the front desk, then a blind transfer sent to an internal 
phone.  Today they simply see "Front Desk" and there is no indication of 
who the actual caller is to distinguish internal staff, internal guest 
room, or outside caller.

Has anyone attacked these things with Polycom that might share their 
approach?

Thanks!

j



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Re: [asterisk-users] What do you use? .conf or AEL?

[Tzafrir Cohen]

On Wed, Feb 11, 2009 at 12:14:33PM -0600, Matthew Nicholson wrote:
> Of course you should be using Lua.
> 
> More seriously, use whatever works best for you.  If you have time,
> evaluate all three alternatives and pick the one you like the best.  If
> you don't have the time, I wouldn't put a lot of effort into switching
> to AEL or Lua based dialplans.
> 
> There are advantages to using both AEL and Lua over conventional ".conf"
> dialplan code (and some disadvantages).  Mainly, complex AEL and Lua
> dialplans are easier to read, write, and maintain then complex
> traditional dialplans.  At the same time, traditional ".conf" dialplans
> are not going away anytime soon, and you do not lose any functionality
> vs AEL and Lua.  So trying to pick one over another is very similar to
> the problem of picking one programming language over another (e.g.  "Is
> python better then ruby?").

AEL is basically a very simple dialplan compiler. That is: things there
translate pretty much directly to the internal dialplan structure (That
is very close in spirit to what you write in a .conf file).

Lua works differently (e.g: no "priority").

Does this help?

Anybody here misses res_perl and/or res_js?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] What do you use? .conf or AEL?

[Tilghman Lesher]

On Wednesday 11 February 2009 13:39:58 Philipp Kempgen wrote:
> Tilghman Lesher schrieb:
> > On Wednesday 11 February 2009 13:07:16 Gordon Henderson wrote:
> >> On Wed, 11 Feb 2009, Tilghman Lesher wrote:
> >> > My viewpoint is that you should work on separation of your application
> >> > code versus data, so that other than new development, your dialplan
> >> > should be completely static and never need changing (other than, like
> >> > I said, new development).
> >>
> >> But what would you call "new development"? Say I have a site who has
> >> many extensions and they then wanted to create a call-group - ie. one
> >> new extension, ring multiple phones?
> >>
> >> In my world, they go to the web interface, create the extension, tick a
> >> selection of existing extensions and the code then writes out a new
> >> segment of dialplan to create the new extension, issues an extensions
> >> reload command to asterisk and off it goes...
> >
> > I'd have a range of extensions, when dialled, it goes to the database,
> > retrieves the list of channels, and dials those channels.  The web
> > frontend would look exactly the same, but the data would go directly into
> > a database, not taking an extra step to go into a dialplan, then reload
> > the text file.
>
> How do you define the hints (for BLF, directed pickup, group
> pickup etc.)?

As of 1.6, you can have pattern-match hints that query a database for the
actual set of channels.

-- 
Tilghman

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Re: [asterisk-users] What do you use? .conf or AEL?

[Philipp Kempgen]

Tilghman Lesher schrieb:
> On Wednesday 11 February 2009 13:07:16 Gordon Henderson wrote:
>> On Wed, 11 Feb 2009, Tilghman Lesher wrote:

>> > My viewpoint is that you should work on separation of your application
>> > code versus data, so that other than new development, your dialplan
>> > should be completely static and never need changing (other than, like I
>> > said, new development).

>> But what would you call "new development"? Say I have a site who has many
>> extensions and they then wanted to create a call-group - ie. one new
>> extension, ring multiple phones?
>>
>> In my world, they go to the web interface, create the extension, tick a
>> selection of existing extensions and the code then writes out a new
>> segment of dialplan to create the new extension, issues an extensions
>> reload command to asterisk and off it goes...

> I'd have a range of extensions, when dialled, it goes to the database,
> retrieves the list of channels, and dials those channels.  The web frontend
> would look exactly the same, but the data would go directly into a database,
> not taking an extra step to go into a dialplan, then reload the text file.

How do you define the hints (for BLF, directed pickup, group
pickup etc.)?


   Philipp Kempgen

-- 
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Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] What do you use? .conf or AEL?

[Tilghman Lesher]

On Wednesday 11 February 2009 13:07:16 Gordon Henderson wrote:
> On Wed, 11 Feb 2009, Tilghman Lesher wrote:
> > On Wednesday 11 February 2009 08:22:39 Gordon Henderson wrote:
> >> For my appication, I get on OK with pure "dialplan". I have a fully
> >> featured PBX system which runs on nothing more than dialplan, and I'm
> >> happy with it. I do have something "higher level" that generates some of
> >> the dialplan for me, but I still had to write that dialplan in the first
> >> place and I was happy to do it in pure "dialplan".
> >
> > My viewpoint is that you should work on separation of your application
> > code versus data, so that other than new development, your dialplan
> > should be completely static and never need changing (other than, like I
> > said, new development).
>
> One mans program is another mans data
>
> But what would you call "new development"? Say I have a site who has many
> extensions and they then wanted to create a call-group - ie. one new
> extension, ring multiple phones?
>
> In my world, they go to the web interface, create the extension, tick a
> selection of existing extensions and the code then writes out a new
> segment of dialplan to create the new extension, issues an extensions
> reload command to asterisk and off it goes... The dialplan was static
> before, and is static after, it's just that I wrote some php to write
> dialplan based on user input...
>
> How do others do it?

I'd have a range of extensions, when dialled, it goes to the database,
retrieves the list of channels, and dials those channels.  The web frontend
would look exactly the same, but the data would go directly into a database,
not taking an extra step to go into a dialplan, then reload the text file.
The advantage is that I'd never have the possibility of two people colliding
on the regeneration of a text file.  While it is possible for two people to
select the same number when defining new extensions, that can be very
easily worked around, given that database updates are atomic.

-- 
Tilghman

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Re: [asterisk-users] The download link, why server down?

[Tilghman Lesher]

On Wednesday 11 February 2009 13:08:54 bilal ghayyad wrote:
> Hi All;
>
> Why the below does not work? Since about 10 days?
>
> wget http://ftp.digium.com/pub/zaptel/zaptel-1.2-current.tar.gz

The server isn't down.  The name of the server has been downloads.digium.com
for quite some time, and the 1.2 version of zaptel is not current, so of
course there is no symlink there.

-- 
Tilghman

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[asterisk-users] [OT] Re: What do you use? .conf or AEL?

[Philipp Kempgen]

Matthew Nicholson schrieb:
> Of course you should be using Lua.

I'd like to file a bug report.

Incited by all this extensions.* euphoria I'm trying to use
extensions.js but Asterisk doesn't even try to read the file.
The logs don't give any helpful information about the problem.
Reproducibility: always.

Expected behavior: Asterisk should read my extensions.js file
and run the JavaScript code.

This is a production system (Asterisk 1.0) so please send the
bug fix asap.  ;-)


   Philipp Kempgen

-- 
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Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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[asterisk-users] The download link, why server down?

[bilal ghayyad]

Hi All;

Why the below does not work? Since about 10 days?

wget http://ftp.digium.com/pub/zaptel/zaptel-1.2-current.tar.gz

Regards
Bilal


  

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Re: [asterisk-users] What do you use? .conf or AEL?

[Gordon Henderson]

On Wed, 11 Feb 2009, Tilghman Lesher wrote:

> On Wednesday 11 February 2009 08:22:39 Gordon Henderson wrote:
>> For my appication, I get on OK with pure "dialplan". I have a fully
>> featured PBX system which runs on nothing more than dialplan, and I'm
>> happy with it. I do have something "higher level" that generates some of
>> the dialplan for me, but I still had to write that dialplan in the first
>> place and I was happy to do it in pure "dialplan".
>
> My viewpoint is that you should work on separation of your application code
> versus data, so that other than new development, your dialplan should be
> completely static and never need changing (other than, like I said, new
> development).

One mans program is another mans data

But what would you call "new development"? Say I have a site who has many 
extensions and they then wanted to create a call-group - ie. one new 
extension, ring multiple phones?

In my world, they go to the web interface, create the extension, tick a 
selection of existing extensions and the code then writes out a new 
segment of dialplan to create the new extension, issues an extensions 
reload command to asterisk and off it goes... The dialplan was static 
before, and is static after, it's just that I wrote some php to write 
dialplan based on user input...

How do others do it?

Gordon

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Re: [asterisk-users] What do you use? .conf or AEL?

[Terry Wilson]

> Of course you should be using Lua.

I really have to try that sometime

> tAt the same time, traditional ".conf" dialplans
> are not going away anytime soon, and you do not lose any functionality
> vs AEL and Lua.

My reason for sticking with .conf files so far?  "dialplan show" -- it  
is easier (perhaps because I am used to it) to debug when what I see  
on the CLI matches 1 to 1 with what my dialplan shows.

>  So trying to pick one over another is very similar to
> the problem of picking one programming language over another (e.g.   
> "Is
> python better then ruby?").

That isn't hard, it's python!
(or ruby)


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Re: [asterisk-users] asterisk across a firewall

[Gordon Henderson]

On Wed, 11 Feb 2009, Erick Perez wrote:

> Excuse my ignorance but if i have an asterisk in a LAN, and i have
> users in their homes/internet (dozens), in order to correctly connect
> those users across my firewall, what is the technology that i need to
> buy, called?
> secure border gateway?
> session controller?
> secure gateway?
> the audiocodes site seems to have many names for the same thing...but
> i better ask here and learn before i make a big mistake.
>
> my customer has a dumb firewall (not SIP aware) that will not replace.
> he wants another box to do the magic.

I have many customers like that, and "working from home" is gaining 
momenting where I live...

So the scenario (if I interpret it correctly): Asterisk at HQ is behind a 
NAT firewall with remote users (who themselves may be behing a NAT 
firewall)

HQ needs a static IP address on the outside and plenty of bandwidth.

The dumb router at HQ needs to port-forward external port 5060 and 
1-2 into the asterisk box (you can limit this range - see 
rtp.conf) Most dumb routers can port-forward.

Asterisk needs to know it's LAN and extneral ip address - sip.conf, 
externip= and localnet=

remote extensions need nat=yes in sip.conf

and that's basically it.

If the remote extensions are themselves behind a NAT firewall, then the 
easiest way to get them through it is by using a stun server - ether run 
your own, or use someone elses... Do not do any port-forwarding at the 
remote users sites.

Yes, you can fiddle about with proxies, gateways, etc. but keep it simple 
to start with and I have many installations doing it this way and it "just 
works". One day I'm sure I'll trip up, but until then...

Pitfalls - the same with all VoIP - bandwidth, espeically outgoing b/w 
from HQ. Broken NAT gateways, and routers which have SIP ALGs built in 
which are also broken. (Turn them off!)

Routers with broken SIP ALG are the biggest PITA to work round.

Gordon

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Re: [asterisk-users] asterisk across a firewall

[Alex Balashov]

It all depends on how much money you want to spend and how scalable you
want your platform to be, as well as your level of comfort with open source
technology stacks vs. proprietary vendor gear.

You could pull this off with a SIP proxy like Kamailio/OpenSIPS and
Mediaproxy if you wanted.  And up from there.

On Wed, 11 Feb 2009 13:21:06 -0500, Erick Perez  wrote:
> Excuse my ignorance but if i have an asterisk in a LAN, and i have
> users in their homes/internet (dozens), in order to correctly connect
> those users across my firewall, what is the technology that i need to
> buy, called?
> secure border gateway?
> session controller?
> secure gateway?
> the audiocodes site seems to have many names for the same thing...but
> i better ask here and learn before i make a big mistake.
> 
> my customer has a dumb firewall (not SIP aware) that will not replace.
> he wants another box to do the magic.
> 
> --
> 
> Erick Perez
> Cel +(507) 6675-5083
> 
> 
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] asterisk across a firewall

[Tim Nelson]

OpenVPN?

--Tim

- "Erick Perez"  wrote:

> Excuse my ignorance but if i have an asterisk in a LAN, and i have
> users in their homes/internet (dozens), in order to correctly connect
> those users across my firewall, what is the technology that i need to
> buy, called?
> secure border gateway?
> session controller?
> secure gateway?
> the audiocodes site seems to have many names for the same thing...but
> i better ask here and learn before i make a big mistake.
> 
> my customer has a dumb firewall (not SIP aware) that will not
> replace.
> he wants another box to do the magic.
> 
> -- 
> 
> Erick Perez
> Cel +(507) 6675-5083
> 
> 
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>http://lists.digium.com/mailman/listinfo/asterisk-users

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[asterisk-users] Intercom/Doorbell Integration

[Gavin Lewandowski]

Hi,

I'm trying to integrate the following into my Asterisk environment.

BPT Targa Single button audio panel 
(http://www.bpt.co.uk/entry-control/pictures/albums/targha/pages/HSC1ST%20Targha%20Entry%20Panel%20Mounted_JPG.htm),
 
linked to an IT200 interface unit 
(http://www.norbain.co.uk/support//manuals/ref:M412CA0D98C67B/) page 6

I have been told by the engineer that the Targa will generate a tone to 
ring an attached phone to the IT200 interface unit.

However, I do not wish to attach a phone to the IT200 and wish to 
connect it into my Asterisk box.

The problem here, is the Asterisk box is located in a data center and I 
have an MPLS link to it - so we cannot connect the IT200 to a Zap card 
or similar.

I also have a Linksys SPA3102 ATA and I was thinking if we could connect 
the IT200 to this, and get the SPA to call an extension on the Asterisk 
box to make a phone ring when someone presses the door bell.

Many thanks for any hints,

Gavin



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T:+44 1454 31   F:+44 1454 32
M:+44 7939 607840   E:ga...@lewandowski.org.uk
SIP/MSN:ga...@lewandowski.org.uk

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[asterisk-users] asterisk across a firewall

[Erick Perez]

Excuse my ignorance but if i have an asterisk in a LAN, and i have
users in their homes/internet (dozens), in order to correctly connect
those users across my firewall, what is the technology that i need to
buy, called?
secure border gateway?
session controller?
secure gateway?
the audiocodes site seems to have many names for the same thing...but
i better ask here and learn before i make a big mistake.

my customer has a dumb firewall (not SIP aware) that will not replace.
he wants another box to do the magic.

-- 

Erick Perez
Cel +(507) 6675-5083


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Re: [asterisk-users] What do you use? .conf or AEL?

[Matthew Nicholson]

Of course you should be using Lua.

More seriously, use whatever works best for you.  If you have time,
evaluate all three alternatives and pick the one you like the best.  If
you don't have the time, I wouldn't put a lot of effort into switching
to AEL or Lua based dialplans.

There are advantages to using both AEL and Lua over conventional ".conf"
dialplan code (and some disadvantages).  Mainly, complex AEL and Lua
dialplans are easier to read, write, and maintain then complex
traditional dialplans.  At the same time, traditional ".conf" dialplans
are not going away anytime soon, and you do not lose any functionality
vs AEL and Lua.  So trying to pick one over another is very similar to
the problem of picking one programming language over another (e.g.  "Is
python better then ruby?").

On Wed, 2009-02-11 at 12:58 -0500, C F wrote:
> Of course you should be using .conf
> 
> On Tue, Feb 10, 2009 at 2:28 AM, Lee, John (Sydney)
>  wrote:
> > Of course you should be using AEL.
> >
> >> -Original Message-
> >> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> >> boun...@lists.digium.com] On Behalf Of Alan Lord (News)
> >> Sent: Tuesday, 10 February 2009 6:24 PM
> >> To: asterisk-users@lists.digium.com
> >> Subject: [asterisk-users] What do you use? .conf or AEL?
> >>
> >> Hi all,
> >>
> >> I built my first asterisk using the traditional (?) .conf files and
> >> constructs.
> >>
> >> I recall reading books at the time about AEL but it seemed "new" and
> >> untested so I left it alone.  Now, I'm interested to poll the audience
> >> here to see if I should look into using AEL instead (or in addition
> > to)
> >> for future work.
> >>
> >> TIA
> >>
> >>
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> >> To UNSUBSCRIBE or update options visit:
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> >
> >
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> >
> 
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-- 
Matthew Nicholson
Digium, Inc. | Software Developer


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[asterisk-users] ChanSpy problem

[Jim Dickenson]

I have an extension defined like this:

exten => do_monitor,1,Answer()
exten => do_monitor,n,NoOp(Just got '${CfMC_ActionID}')
exten => do_monitor,n,ChanSpy(${CfMC_WhoHear},qX)
exten => do_monitor,n,Hangup()


I use an AMI packet like this:

Action: Originate
Channel: Agent/1001
Exten: do_monitor
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=callE1334
Variable: CfMC_WhoHear=SIP/dickenson
ActionID: callE1324
Async: true


It seems as if the X option to the ChanSpy application causes asterisk to
get into a loop executing priorities 1-3 of the extension until I press a 1
to go to extension 1 in the same context as the do_monitor extension is in.

While asterisk is in this loop I hear no audio from the spied upon channel.

I am running Asterisk SVN-branch-1.6.0-r174845 due to problems in 1.6.0.5
that have been fixed in the current source code.

I thought the X option, added in 1.6, would resolve I problem I submitted a
patch for a few days ago to 1.4.23.1.

It might resolve the problem but I can tell as this loop problem occurred
during my testing.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/




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Re: [asterisk-users] What do you use? .conf or AEL?

[C F]

Of course you should be using .conf

On Tue, Feb 10, 2009 at 2:28 AM, Lee, John (Sydney)
 wrote:
> Of course you should be using AEL.
>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>> boun...@lists.digium.com] On Behalf Of Alan Lord (News)
>> Sent: Tuesday, 10 February 2009 6:24 PM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] What do you use? .conf or AEL?
>>
>> Hi all,
>>
>> I built my first asterisk using the traditional (?) .conf files and
>> constructs.
>>
>> I recall reading books at the time about AEL but it seemed "new" and
>> untested so I left it alone.  Now, I'm interested to poll the audience
>> here to see if I should look into using AEL instead (or in addition
> to)
>> for future work.
>>
>> TIA
>>
>>
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>
>
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Re: [asterisk-users] Billing and Soft Switch.

[Alex Balashov]

David @ULC wrote:

> Looking for a Free VOIP Billing and Soft Switch.
> 
> Any suggestions ?

I'm looking to put the milk back in the cow.

If you have the skinny on that, maybe we can swap suggestions.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] DTMF tones mid conversation

[F6HQZ]

Hi men,

Resolved for one of my customers by upgrading Asterisk/Libpri/Zaptel.
I don't remember what wer the versions, sorry.
Check and advise us the results, please.

Best Regards,
Francois

No virus found in this outgoing message.
Checked by AVG - www.avg.com 
Version: 8.0.233 / Virus Database: 270.10.19/1941 - Release Date: 02/09/09 
06:50:00


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Re: [asterisk-users] Billing and Soft Switch.

[Steve Howes]

On 11 Feb 2009, at 14:22, David @ULC wrote:
> Looking for a Free VOIP Billing and Soft Switch.

And you are asking an Asterisk list... Asterisk? Billing is probably  
best doing a custom job..

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Re: [asterisk-users] What do you use? .conf or AEL?

[Tilghman Lesher]

On Wednesday 11 February 2009 08:22:39 Gordon Henderson wrote:
> For my appication, I get on OK with pure "dialplan". I have a fully
> featured PBX system which runs on nothing more than dialplan, and I'm
> happy with it. I do have something "higher level" that generates some of
> the dialplan for me, but I still had to write that dialplan in the first
> place and I was happy to do it in pure "dialplan".

My viewpoint is that you should work on separation of your application code
versus data, so that other than new development, your dialplan should be
completely static and never need changing (other than, like I said, new
development).

-- 
Tilghman

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Re: [asterisk-users] Aastra phone crashes with Asterisk 1.6

[Steve Davies]

2009/2/11 OCG Technical Support :
> Don't expect too much from Aastra.  In our previous dealings trying to
> report serious bugs (like phone lockup/crash) to Aastra, they didn't want
> the details, or they simply gave us canned answers which did no good.
> (Superficial tech support)
>
> We've moved away from Aastra for new installs, but we still have to support
> old customers with Aastra
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
> Kempgen
> Sent: February 11, 2009 12:45 AM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] Aastra phone crashes with Asterisk 1.6
>
> Carlos Chavez schrieb:
>>   I upgraded my office server from 1.4.22 to 1.6.0.5 on the weekend
> and
>> after some testing there seems to be a compatibility problem when using
>> Aastra phones.
>
>> If I dial any of those phones the
>> call will drop after a minute or so and the phone will crash.
>
> I'm not saying it's not an Asterisk problem. Maybe something in
> the SIP signaling/RTP is broken.
>
> However it's definitely an Aastra problem. No matter how broken
> the signaling -- that's no excuse for crashing. So make sure to
> report the issue to Aastra as well.
>
>
>   Philipp Kempgen

Aastra prefer you to handle support through the person who sold you
the phone. Not unreasonable I suppose...

Assuming you understand IP networking and SIP reasonably well, I would
suggest capturing a trace of all UDP traffic between Asterisk and the
phone, using tcpdump, and then looking at it in Wireshark's VoIP
decoder as this will at least indicate which party is hanging up, and
will perhaps reveal the reason. If it is after exactly 60 seconds
every time, it is possibly some kind of timeout setting.

I would also trawl through all of the non-default settings on the
Aastras in case one of them is causing the issue.

Regards,
Steve

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Re: [asterisk-users] Aastra phone crashes with Asterisk 1.6

[OCG Technical Support]

Don't expect too much from Aastra.  In our previous dealings trying to
report serious bugs (like phone lockup/crash) to Aastra, they didn't want
the details, or they simply gave us canned answers which did no good.
(Superficial tech support)

We've moved away from Aastra for new installs, but we still have to support
old customers with Aastra


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: February 11, 2009 12:45 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Aastra phone crashes with Asterisk 1.6

Carlos Chavez schrieb:
>   I upgraded my office server from 1.4.22 to 1.6.0.5 on the weekend
and
> after some testing there seems to be a compatibility problem when using
> Aastra phones.

> If I dial any of those phones the
> call will drop after a minute or so and the phone will crash.

I'm not saying it's not an Asterisk problem. Maybe something in
the SIP signaling/RTP is broken.

However it's definitely an Aastra problem. No matter how broken
the signaling -- that's no excuse for crashing. So make sure to
report the issue to Aastra as well.


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions

[Heath Roberts]

On Tue, Feb 10, 2009 at 12:23 PM, Erick Perez  wrote:
> Hi, I am looking to connect 66 analog phones to an asterisk box.

> other hardware suggestions for this task will be nice.

Citel makes a box:
http://www.citel.com/Products/Portico.asp

If you're converting from Definity, Norstar, p-phone (Nortel Centrex),
Meridian, or NEC, a google search found them for under $500 (for 24
ports):
http://www.neobits.com/digital_handset_converters_s263.html

I've not used the Citel product, but have heard good things about them.

Depending on what it is, you could also just leave the existing switch
in place and connect with a PRI interface--let your PBX think the
asterisk box is the telco.
-- 
Heath Roberts
htrobe...@gmail.com

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Re: [asterisk-users] DTMF tones mid conversation

[Paulo Santos]

Andrew Thomas wrote:

> I seem to have a problem of intermittent DTMF tones being played during
> a conversation.

I'm having the same problem, but in my case, it's every 1 minute and at 
the start of the call.

I wonder if it has anything to do with echo cancellation.
I've only noticed when using a Zap channel, but I'll run some more tests.

asterisk 1.4.17 / addons 1.4.7 / zaptel 1.4.12.1 / mISDN 1.1.8

Paulo Santos

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[asterisk-users] Billing and Soft Switch.

[David @ULC]

Looking for a Free VOIP Billing and Soft Switch.
Any suggestions ?
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Re: [asterisk-users] Billing and Soft Switch.

[Philipp Kempgen]

David @ULC schrieb:
> Looking for a Free VOIP Billing and Soft Switch.

"soft switch" includes back-to-back user agents (Asterisk) I guess?


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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[asterisk-users] DTMF tones mid conversation

[Andrew Thomas]

Hi helpers,

I seem to have a problem of intermittent DTMF tones being played during
a conversation.

Eg: Extn 100 takes an inbound call and all is fine.  Except, at an
undetermined time the person on extn 100 will here a DTMF tone for no
apparent reason (it's not the caller pressing buttons).  The caller
doesn't hear the tone - only the called person.  The call itself
progresses normally.

I am using PRI --> * --> various SIP hard phones.  But I have also heard
it on BRI as well.

Any ideas where to start looking to cure this?

Cheers
Andy


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[asterisk-users] Expressions, CUT(), ... (was: Re: call forward all except the extension it is forwarded to)

[Philipp Kempgen]

Vieri schrieb:

> I'm using Set(RETURN_EXT=${BLINDTRANSFER:4:4})
> but that assumes that I have only 4-digit extensions

Well, skip the length argument (the second ":4").

> and all have the same prefix (SIP/).
> Is there a more "portable" way?

Set(RETURN_EXT=CUT(BLINDTRANSFER,/,2));


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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[asterisk-users] Looking for 'remote Asterisk hands' support in Mexico

[Jeff Verheyen]

Hello,


We are looking for someone who can act as our 'remote Asterisk hands' in
Mexico. One of our customers has opened an new office in the 'Col. San
Rafael, Delg. Azcapotzalco' region. We need someone (or organisation) to
perform the onsite installation and connections for an Asterisk server
(and maybe other servers too).

If you're able and interested to help us with the Mexico part of our
Asterisk project feel free to contact me at: jeff.verhe...@ampersant.be


Kind regards,


Jeff




-- 
Ampersant bvba - Jeff Verheyen
 Dr. Jacobsstraat 1 - 2570 DUFFEL - Belgium
 Tel: +32 15 32 36 19 - Fax: +32 15 32 37 90
 Email: jeff.verhe...@ampersant.be
 Website: http://www.ampersant.be/


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Re: [asterisk-users] What do you use? .conf or AEL?

[Gordon Henderson]

On Wed, 11 Feb 2009, Philipp Kempgen wrote:

> Gordon Henderson schrieb:
>> On Wed, 11 Feb 2009, Alan Lord (News) wrote:
>>
>>> That was quite an interesting set of responses. I didn't get any
>>> impression that there is a strong preference either way.
>>
>> I asked the same question some time back too... Got a few replies, and now
>> (as then), all my systems are 100% .conf (or dialplan code, whatever you
>> want to call it)
>>
>> John Lees reply did rather irriate me - because he gave no explanation, or
>> justification for it - hence my own terse reply!
>>
>> I have some huge dialplans with 1000's of lines of code in them, but most
>> of it is actually computer generated - as I said in a post some time back:
>> "PHP is my AEL"... I'll code something by hand, then get PHP to generate
>> multiple instances of it, rewriting my extensions.conf file every time.
>>
>> I haven't yet had to resort to AEL or AGI and personally I'm all for
>> keeping the "core" as simple as possible - less code to go wrong and all
>> that.
>>
>> This to me seems OK to me, as once a system is installed, the changes to
>> it are infrequent, and a reload takes a fraction of a second.
>
> So what's so irritating about AEL then? You are using a higher
> level language to generate extensions.conf as well.

I didn't say anything was irritating about AEL. I said I was irritated by 
John Lee's terse reply which I quote: "Of course you should be using AEL" 
which he gave with no justification at all.

My view is that you should use what you need to use for your application.

For my appication, I get on OK with pure "dialplan". I have a fully 
featured PBX system which runs on nothing more than dialplan, and I'm 
happy with it. I do have something "higher level" that generates some of 
the dialplan for me, but I still had to write that dialplan in the first 
place and I was happy to do it in pure "dialplan".

Gordon

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Re: [asterisk-users] Asterisk AGX addons compile issues

[Andrew Thomas]

svn co https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addons agx-ast-addons


./build_sh from the trunk.

 
-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: 10 February 2009 18:35
To: mich...@networkstuff.co.nz; Asterisk Users Mailing List - Non-Commercial 
Discussion
Subject: Re: [asterisk-users] Asterisk AGX addons compile issues


2008/12/18 Michael 
Has anyone seen this before, and know what is happening?

u...@host:~/asterisk/agx-ast-addons# ./build.sh
-- Configuring done
-- Generating done
-- Build files have been written to: /root/asterisk/agx-ast-addons
[ 11%] Building C object CMakeFiles/app_devstate.dir/app_devstate.o
Linking C shared module dist/app_devstate.so
[ 11%] Built target app_devstate
[ 22%] Building C object
CMakeFiles/app_nv_backgrounddetect.dir/app_nv_backgrounddetect.o
Linking C shared module dist/app_nv_backgrounddetect.so
[ 22%] Built target app_nv_backgrounddetect
[ 33%] Building C object CMakeFiles/app_nv_faxdetect.dir/app_nv_faxdetect.o
Linking C shared module dist/app_nv_faxdetect.so
[ 33%] Built target app_nv_faxdetect
[ 44%] Building C object CMakeFiles/app_pickup2.dir/app_pickup2.o
Linking C shared module dist/app_pickup2.so
[ 44%] Built target app_pickup2
[ 55%] Building C object CMakeFiles/app_rxfax.dir/app_rxfax.o
cc1: warnings being treated as errors
/root/asterisk/agx-ast-addons/app_rxfax.c: In function 'phase_e_handler':
/root/asterisk/agx-ast-addons/app_rxfax.c:126: warning: implicit declaration
of function 't30_get_local_ident'
/root/asterisk/agx-ast-addons/app_rxfax.c:127: warning: implicit declaration
of function 't30_get_far_ident'
/root/asterisk/agx-ast-addons/app_rxfax.c: In function 'rxfax_exec':
/root/asterisk/agx-ast-addons/app_rxfax.c:380: warning: implicit declaration
of function 't30_set_local_ident'
/root/asterisk/agx-ast-addons/app_rxfax.c:383: warning: implicit declaration
of function 't30_set_header_info'
/root/asterisk/agx-ast-addons/app_rxfax.c:385: warning: passing argument 2
of 't30_set_phase_b_handler' from incompatible pointer type
/root/asterisk/agx-ast-addons/app_rxfax.c:386: warning: passing argument 2
of 't30_set_phase_d_handler' from incompatible pointer type
make[2]: *** [CMakeFiles/app_rxfax.dir/app_rxfax.o] Error 1
make[1]: *** [CMakeFiles/app_rxfax.dir/all] Error 2
make: *** [all] Error 2
u...@host:~/asterisk/agx-ast-addons#

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Hi,

I think the trick is to download trunk version from svn (see voip-info.org for 
instrcution).

Regards 


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Re: [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions

[Jeff LaCoursiere]

I'm a big fan of Audicodes MP-124.  Very stable and a full feature set, 
and it has amphenol connectors so you can tie directly to 66-blocks for 
cabling your phones.

j

On Tue, 10 Feb 2009, Erick Perez wrote:

> Hi, I am looking to connect 66 analog phones to an asterisk box. I was
> thinking of a Xorcom astribank 32port (2 of them and another 8 port).
> this is because the phones have no near connection to an ip network,
> so replacing the phones in favor of  voip phones+network cabling is
> kinda out of the question.
>
> In your experience, will these units support all the phones talking at
> the same time with other units on the astribank, as well as to the
> pbx, pstn, etc? The asterisk pbx will be a server-class Hp Proliant
> unit (potentially a dl320). i must make sure the astribanks will not
> die when fully utilized.
>
> other hardware suggestions for this task will be nice.
>
> thanks,
>
>
> -- 
> 
> Erick
> 
>
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Re: [asterisk-users] call forward all except the extension it is forwarded to

[Vieri]

--- On Wed, 2/11/09, Vieri  wrote:

> > >> I would like to know if I can set Call
> Forwarding
> > on an extension but allow direct calls from the
> extension it
> > is forwarded to.
> > >> 
> > >> Example:
> > >> Extension 100 sets call forwarding (all) to
> > extension 101.
> > >> All calls to 100 are immediately forwarded to
> 101
> > as expected.
> > >> However, if 101 tries to transfer a call to
> 100 or
> > tries to call 100 directly, it sounds "busy"
> > because it obviously goes into a loop where 101 tries
> to
> > call itself.
> > >> How do I set an "exception" so that
> 101
> > can actually call 100 even if the latter is forwarded
> to
> > 101?
> > 
> > > Use the "if" statement.
> > > 
> > > 100 => {
> > >   if ("${CALLERID(num)}" =
> "101") {
> > >   // call 100
> > >   Dial(SIP/100);
> > >   }
> > >   else {
> > >   // forward to 101
> > >   Dial(Local/101);
> > >   }
> > > }
> > 
> > Or use Caller ID matching:
> > 
> > 100/101 => {
> > // caller is 101
> > Dial(SIP/100);
> > }
> > 100 => {
> > // for other callers
> > Dial(Local/101);
> > }
> 
> I hacked the dialparties.agi script in freepbx and now it
> works for direct calls but fails for blind transfers,
> probably because I need to check another variable such as
> BLINDTRANSFER. Attended transfers seem to work.

What is the "best" way to get the extension number of the user who is making a 
blind transfer?
I'm using Set(RETURN_EXT=${BLINDTRANSFER:4:4})
but that assumes that I have only 4-digit extensions and all have the same 
prefix (SIP/).
Is there a more "portable" way?



  

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Re: [asterisk-users] What do you use? .conf or AEL?

[Philipp Kempgen]

Gordon Henderson schrieb:
> On Wed, 11 Feb 2009, Alan Lord (News) wrote:
> 
>> That was quite an interesting set of responses. I didn't get any
>> impression that there is a strong preference either way.
> 
> I asked the same question some time back too... Got a few replies, and now 
> (as then), all my systems are 100% .conf (or dialplan code, whatever you 
> want to call it)
> 
> John Lees reply did rather irriate me - because he gave no explanation, or 
> justification for it - hence my own terse reply!
> 
> I have some huge dialplans with 1000's of lines of code in them, but most 
> of it is actually computer generated - as I said in a post some time back: 
> "PHP is my AEL"... I'll code something by hand, then get PHP to generate 
> multiple instances of it, rewriting my extensions.conf file every time.
> 
> I haven't yet had to resort to AEL or AGI and personally I'm all for 
> keeping the "core" as simple as possible - less code to go wrong and all 
> that.
> 
> This to me seems OK to me, as once a system is installed, the changes to 
> it are infrequent, and a reload takes a fraction of a second.

So what's so irritating about AEL then? You are using a higher
level language to generate extensions.conf as well.


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] call forward all except the extension it is forwarded to

[Vieri]

--- On Wed, 2/11/09, Philipp Kempgen  wrote:

> >> I would like to know if I can set Call Forwarding
> on an extension but allow direct calls from the extension it
> is forwarded to.
> >> 
> >> Example:
> >> Extension 100 sets call forwarding (all) to
> extension 101.
> >> All calls to 100 are immediately forwarded to 101
> as expected.
> >> However, if 101 tries to transfer a call to 100 or
> tries to call 100 directly, it sounds "busy"
> because it obviously goes into a loop where 101 tries to
> call itself.
> >> How do I set an "exception" so that 101
> can actually call 100 even if the latter is forwarded to
> 101?
> 
> > Use the "if" statement.
> > 
> > 100 => {
> > if ("${CALLERID(num)}" = "101") {
> > // call 100
> > Dial(SIP/100);
> > }
> > else {
> > // forward to 101
> > Dial(Local/101);
> > }
> > }
> 
> Or use Caller ID matching:
> 
> 100/101 => {
>   // caller is 101
>   Dial(SIP/100);
> }
> 100 => {
>   // for other callers
>   Dial(Local/101);
> }

I hacked the dialparties.agi script in freepbx and now it works for direct 
calls but fails for blind transfers, probably because I need to check another 
variable such as BLINDTRANSFER. Attended transfers seem to work.

Thanks




  

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Re: [asterisk-users] What do you use? .conf or AEL?

[Gordon Henderson]

On Wed, 11 Feb 2009, Alan Lord (News) wrote:

> That was quite an interesting set of responses. I didn't get any
> impression that there is a strong preference either way.

I asked the same question some time back too... Got a few replies, and now 
(as then), all my systems are 100% .conf (or dialplan code, whatever you 
want to call it)

John Lees reply did rather irriate me - because he gave no explanation, or 
justification for it - hence my own terse reply!

I have some huge dialplans with 1000's of lines of code in them, but most 
of it is actually computer generated - as I said in a post some time back: 
"PHP is my AEL"... I'll code something by hand, then get PHP to generate 
multiple instances of it, rewriting my extensions.conf file every time.

I haven't yet had to resort to AEL or AGI and personally I'm all for 
keeping the "core" as simple as possible - less code to go wrong and all 
that.

This to me seems OK to me, as once a system is installed, the changes to 
it are infrequent, and a reload takes a fraction of a second.

Good luck with whatever approach you choose!

Gordon

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Re: [asterisk-users] call forward all except the extension it is forwarded to

[Philipp Kempgen]

Philipp Kempgen schrieb:
> Vieri schrieb:
>> I would like to know if I can set Call Forwarding on an extension but allow 
>> direct calls from the extension it is forwarded to.
>> 
>> Example:
>> Extension 100 sets call forwarding (all) to extension 101.
>> All calls to 100 are immediately forwarded to 101 as expected.
>> However, if 101 tries to transfer a call to 100 or tries to call 100 
>> directly, it sounds "busy" because it obviously goes into a loop where 101 
>> tries to call itself.
>> How do I set an "exception" so that 101 can actually call 100 even if the 
>> latter is forwarded to 101?

> Use the "if" statement.
> 
> 100 => {
>   if ("${CALLERID(num)}" = "101") {
>   // call 100
>   Dial(SIP/100);
>   }
>   else {
>   // forward to 101
>   Dial(Local/101);
>   }
> }

Or use Caller ID matching:

100/101 => {
// caller is 101
Dial(SIP/100);
}
100 => {
// for other callers
Dial(Local/101);
}


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] call forward all except the extension it is forwarded to

[Vieri]

--- On Wed, 2/11/09, Philipp Kempgen  wrote:

> > I would like to know if I can set Call Forwarding on
> an extension but allow direct calls from the extension it is
> forwarded to.
> > 
> > Example:
> > 
> > Extension 100 sets call forwarding (all) to extension
> 101.
> > 
> > All calls to 100 are immediately forwarded to 101 as
> expected.
> > 
> > However, if 101 tries to transfer a call to 100 or
> tries to call 100 directly, it sounds "busy"
> because it obviously goes into a loop where 101 tries to
> call itself.
> > 
> > How do I set an "exception" so that 101 can
> actually call 100 even if the latter is forwarded to 101?
> > 
> > I'm using * 1.4
> 
> Use the "if" statement.

Thanks. I thought it would be that simple but I guess I´ll have to check out 
the freepbx code and see where I can stick in an "if".



  

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Re: [asterisk-users] Hangup extensions via CLI?

[Lenz Emilitri]

This is a bit of trickery, but could not resist :)

This will kill a channel that is connected to SIP/201

 asterisk -rx "soft hangup $(asterisk -rx 'show channels' | grep SIP/201 |
awk '{ print $1 '} )"

It basically calls *, gets the list of channels, filters them out to get the
channel name and hangs it up.

OK, using AMI and a real programming language and hadling multiple lines
would be better.

Thanks

l.

2009/2/9 Tim Nelson 

> Greetings list-
>
> I'd like the ability to hangup all calls for a particular extension from
> the system CLI. I understand this can probably be scripted using the AMI but
> I'm not familiar on how to do it. Help!
>
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332 x105
>
>
-- 
Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] call forward all except the extension it is forwarded to

[Philipp Kempgen]

Vieri schrieb:
> I would like to know if I can set Call Forwarding on an extension but allow 
> direct calls from the extension it is forwarded to.
> 
> Example:
> 
> Extension 100 sets call forwarding (all) to extension 101.
> 
> All calls to 100 are immediately forwarded to 101 as expected.
> 
> However, if 101 tries to transfer a call to 100 or tries to call 100 
> directly, it sounds "busy" because it obviously goes into a loop where 101 
> tries to call itself.
> 
> How do I set an "exception" so that 101 can actually call 100 even if the 
> latter is forwarded to 101?
> 
> I'm using * 1.4

Use the "if" statement.

100 => {
if ("${CALLERID(num)}" = "101") {
// call 100
Dial(SIP/100);
}
else {
// forward to 101
Dial(Local/101);
}
}

> and freepbx.

Don't know about FreePBX. Ask on their mailing-list/forum/...


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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[asterisk-users] call forward all except the extension it is forwarded to

[Vieri]

I would like to know if I can set Call Forwarding on an extension but allow 
direct calls from the extension it is forwarded to.

Example:

Extension 100 sets call forwarding (all) to extension 101.

All calls to 100 are immediately forwarded to 101 as expected.

However, if 101 tries to transfer a call to 100 or tries to call 100 directly, 
it sounds "busy" because it obviously goes into a loop where 101 tries to call 
itself.

How do I set an "exception" so that 101 can actually call 100 even if the 
latter is forwarded to 101?

I'm using * 1.4 and freepbx.

Thanks

Vieri



  

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Re: [asterisk-users] OPTIONS packets

[Alex Balashov]

Try manipulate the fromuser= parameter in sip.conf.

On Wed, 11 Feb 2009 11:43:13 +0200, michel freiha 
wrote:
> Hi all,
>  I need to register asterisk on an  OpenSIPS SIP Proxy...The Registration
> is
> OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP
> Proxy
> is not replying back...The issue is the UNKNOWN username that reside in
> the
> OPTIONS packet as you can see in the captured packets as you can see
> below:
> 
> 
>1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060
>2. OPTIONS sip:OPENSIPS_IP SIP/2.0.
>3. Via: SIP/2.0/UDP ASTERISK_IP:5060;branch=z9hG4bK2d757382;rport.
>4. From: "Unknown" ;tag=as5a8c9f3b.
>5. To: .
>6. Contact: .
>7. Call-ID: 7cad53ac17cc1ed971a3ffc674ce9...@asterisk_ip.
>8. CSeq: 102 OPTIONS.
>9. User-Agent: Asterisk PBX.
>10. Max-Forwards: 70.
>11. Date: Wed, 11 Feb 2009 09:17:37 GMT.
>12. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY.
>13. Supported: replaces.
>14. Content-Length: 0.
> 
> How i can fix this UNKNOWN username?
> 
> Regards
-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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[asterisk-users] OPTIONS packets

[michel freiha]

Hi all,
 I need to register asterisk on an  OpenSIPS SIP Proxy...The Registration is
OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy
is not replying back...The issue is the UNKNOWN username that reside in the
OPTIONS packet as you can see in the captured packets as you can see below:


   1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060
   2. OPTIONS sip:OPENSIPS_IP SIP/2.0.
   3. Via: SIP/2.0/UDP ASTERISK_IP:5060;branch=z9hG4bK2d757382;rport.
   4. From: "Unknown" ;tag=as5a8c9f3b.
   5. To: .
   6. Contact: .
   7. Call-ID: 7cad53ac17cc1ed971a3ffc674ce9...@asterisk_ip.
   8. CSeq: 102 OPTIONS.
   9. User-Agent: Asterisk PBX.
   10. Max-Forwards: 70.
   11. Date: Wed, 11 Feb 2009 09:17:37 GMT.
   12. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
   13. Supported: replaces.
   14. Content-Length: 0.

How i can fix this UNKNOWN username?

Regards
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Re: [asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems

[nik600]

On Wed, Feb 11, 2009 at 2:49 AM, Steven J. Douglas  wrote:
> Hi,
>
> Have you tried using "externip" in your sip.conf? By setting the correct
> "localnet", any SIP packets that goes elsewhere will use the value in
> "externip". This might solve your problem.
>
> Regards,
> Steve
>

yes i've done it.

The rtp traffic is redirect correctly but the SIP INVITE contains the
ip of the lan and not of the nat.

I'll try with SipAddHeader and then let you know...

thanks
-- 
/*/
nik600
http://www.kumbe.it

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Re: [asterisk-users] Is "a=fmtp:101 0-15" a legal option in SDP ?

[Johansson Olle E]

9 feb 2009 kl. 23.17 skrev Raj Jain:

> On Mon, Feb 9, 2009 at 4:43 PM, Olivier  wrote:
>>
>> Hi,
>>
>> My patton 4638 is sending :
>> v=0
>> o=MxSIP 0 46 IN IP4 192.168.100.52
>> s=SIP Call
>> c=IN IP4 192.168.100.52
>> t=0 0
>> m=audio 4984 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=sendrecv
>> m=image 4986 udptl t38
>> a=T38FaxUdpEC:t38UDPRedundancy
>> a=sendrecv
>>
>>
>> Asterisk (1.4.22.1) replies :
>> Got unsupported a:fmtp in SDP offer
>>
>> Shall I care ?
>
> This error is somewhat benign. Basically, the end-point is telling
> that it can receive RFC 2833 events in the range of 0-15 (DTMF tones)
> and Asterisk is ignoring that range.

To clarify: We're not ignoring DTMF. We're just not parsing the fmtp  
header.
This is a message while doing debug, so don't bother with it.

/O

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Re: [asterisk-users] unistim and transfer calls

[Ralf Träskman]

I have added t in dialplan
exten => 1234,1,Dial(USTM/2...@c,40,t)
so now i can transfer, but when the caller the extension I transfer to hangs up 
asterisk dumps an I have to start it up again.
Any thoughts?

/ralf

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ralf Träskman
Sent: den 10 februari 2009 16:00
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] unistim and transfer calls

Hi
When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make 
the transfer and it rings on the extension I transfer to, but when I accept the 
call, asterisk dumps. How can I get it to work? And how do I save the dump 
error?

Regards
/ralf


Ralf Träskman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
r...@adlibris.com 
www.adlibris.com
P Please consider the environment before printing this e-mail

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Re: [asterisk-users] What do you use? .conf or AEL?

[Alan Lord (News)]

That was quite an interesting set of responses. I didn't get any 
impression that there is a strong preference either way.


Thanks for all the replies.

Al

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