Re: [Asterisk-Users] Zaptel card AND Ztdummy together?

[Rod Bacon]

It's a Digium single-port job. No other timing sources aviailable (the * box IS 
the pbx).



qrss wrote:

What kind of card are they using?  Is there only 1 telco circuit?
If so, then I'm thinking their card should have detected the loss of
service and switched to it's internal clock. Do they have a secondary
clock source available across another circuit? Perhaps a tie line to a pbx
that can be configured as a secondary?

-Original Message-
From: Rod Bacon
Sent: Thu, June 23, 2005 12:03 am

I had a weird (unforeseen) situation today. We have a remote office with


an * server and ISDN 10 service. We connect to each other over an IAX
trunk


with G729.

Today, some of Sydney experienced a power surge which knocked out their


ISDN services. Without a clock source on their PRI card, my IAX calls to
them


resulted in one-way audio (they could hear me, but I not them).

Is it possible to load *both* the relevant card driver *and* ztdummy to


guard against this occurrance?


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[Asterisk-Users] OT: MAX TNT and PRI calling name (CNAM) facility message

[Kevin Blackham]

Does anyone have a MAX/APX with working ingress PRI calling name?

I recently acquired a MAX TNT on the cheap and it's integrating fine
except for one thing.  In the 11.0.0 release notes, it is stated that
ISDN calling name will, if present and permitted by presentation
flags, be added to the From: and Remote-Party-ID: headers of the
INVITE.  I'm not able to make this happen.  Pcap captures show it is
indeed in neither header, and I suspect the MAX is sending the INVITE
before it receives this data.  Debug traces show it does receive the
message, but due to limitations of the CLI, I cannot correlate whether
it's received before or after the INVITE is dispatched.  It works
great direct to Asterisk (of course) via TE410P on the same NI-2
spans.

My FACILITY message that contains the CNAM wanders in from 100 to
400ms after the initial SETUP.  I can't seem to find any way to get
the MAX to stall for a half-second before invoking the INVITE (if
that's even the issue).  Is my provider too slow?  Is there another
valid way for CNAM to be provided during the SETUP message, assuming
my provider can stall the call setup until the SS7 query is returned?
(google for Q.931 docs not helping me much there either)

I know this isn't the place for Ascend/Lucent MAX discussion, but
there doesn't seem to be anything active out there.  I'm looking for a
mail list/newsgroup/community if there is one still alive.
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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Armin Schindler]

On Thu, 23 Jun 2005, Massimo De Nadal wrote:
 Have you planned to integrate some echo cancel feature ?

Echo cancelling (if the card supports it) is already implemented.
As far as I know the Eicon Diva Server cards are the only cards supporting
echo cancel via onboard DSPs.

Armin

 Armin Schindler ha scritto:
  Hi all,
  
  I would like to announce the first release of the chan_capi
  channel driver on sourceforge.net
  
  The package is available for download with name  chan_capi-cm-0.5
  and is the current CVS HEAD.
  
  It is derived from the chan_capi-0.4.0PRE1 of kapejod.
  
  The main changes are:
  - complete rework
  - fix race-conditions
  - fix call state handling
  - rework of debug/verbose messages
  - added capiFax feature (provided by Frank Sautter)
  - auto-config (compile and work with Asterisk CVS-HEAD and older versions)
  - use with ELinOS cross-toolbox and project handling
  
  For the versioning, I have decided to use the name extention 'cm' to avoid
  confusion with kapejod's version.
  This first release is 0.5 (not 0.1) because the base is 0.4.0.
  Only the major and the minor number will be used. The exception to have a
  third number (patch-version) will be added for fixup-patches only.
  
  Feedback welcome.
  
  Armin
  
  PS: sorry for cross-posting.
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Re: [Asterisk-Users] Is anyone using VOIPREACH

[Luki]

 I have been trying to open an account with voipreach.net for over
 a week now and I have not gotten any response from them as yet.
 None of their phone numbers are working.
They didn't respond to my emails either... Tixter is right, forget
about them if they don't even care to reply to take your money.

--Luki
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[Asterisk-Users] ASTCC not making calls

[Juan Luis Moyano]

Hi, im trying to setup ASTCC but I'm getting it difficult. I've
correctly set up the mysql database astcc and added a brand, trunk,
route and a card as follows:

brands
+--+--+--+--+--++--+--+
| name | language | inc  | publishednum | did  | markup | days | fee  |
+--+--+--+--+--++--+--+
| FWD  | es   | 6| 4| 4|  0 | 30   |0 |
+--+--+--+--+--++--+--+

trunks
+--+--+-+
| name | tech | path|
+--+--+-+
| FWD  | IAX2 | 657XXX:[EMAIL PROTECTED] |
+--+--+-+

routes
+-+---++-+-+--+
| pattern | comment   | trunks | connectcost | includedseconds | cost |
+-+---++-+-+--+
| 4.  | FWD   | FWD|   0 |   0 |  150 |
+-+---++-+-+--+

-Added a card with $1 credit and using 'FWD' brand.

extensions.conf
---
[outbound-fwd]
;
exten = _4.,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:1})
exten = _4.,2,Hangup()

iax.conf

register = 657050:[EMAIL PROTECTED]


The problem is that when, for example, I dial '4612' i get:

-- Executing DeadAGI(IAX2/[EMAIL PROTECTED]/3, astcc.agi|21|612) in new 
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
-- Playing 'digits/1' (language 'en')
-- AGI Script astcc.agi completed, returning 0
-- Executing Hangup(IAX2/[EMAIL PROTECTED]/3, ) in new stack

and i hear allison saying I'm sorry that is not a recognized phone
number, goodbye.

Anyone knows what could be happening right here?

Many thanks in advance.

-- 
Juan Luis Moyano
[EMAIL PROTECTED]

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Re: [Asterisk-Users] indexing tables for dialing

[Luki]

Ypek,

 I would like to know how can I manage to implement a table which translates
 an extension number into a phone number. Let see an example:

There are many ways of doing this. You could map the extensions to
phones in extensions.conf, via the internal database or via an
external database, or via an AGI script.

Example:
1) Make database entries:
AGENTS/3201 = 411212,4251113131
AGENTS/3202 = 4251110011,8881114545,7871114545

2) Define dial plan entry:
exten = _,1,Macro(dialagent,${EXTEN})

[macro-dialagent]
exten = s,1,Set(DEST=${DB(AGENTS/${ARG1})})
exten = s,n,Set(N=1)
exten = s,n(loop),Cut(D=DEST,\,,${N})
exten = s,n,GotoIf($[${DEST} : ]?done)
exten = s,n,Dial(SIP/[EMAIL PROTECTED],15)
exten = s,n,GotoIf($[${DIALSTATUS} = ANSWER]?done)
exten = s,n,Set(N=$[${N} + 1])
exten = s,n,Goto(loop)
exten = s,n(done),NoOp(Dial Agent ${ARG1} at ${DEST} done)

This is somewhat paraphrased from my config, but I didn't test this
particular example. You certainly need to adapt it to your setup (like
define how to handle outgoing calls, etc.). This should get your
started. I believe CVS-HEAD is required for this, though.

--Luki
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[Asterisk-Users] Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy

[Robert Rozman]

Hi,

I'm pulling my hair down and getting bold :-) . I have Asterisk between
Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff
Asterisk)

I'm trying to do just plain transfer of call from pbx to ISDN through
Asterisk...

It seems like PBX hangsup, when call is progressing with no apparent reason.
I'd kindly ask for any advice or some working example for this

On isdn side I also have a problem. Asterisk quite often says that it cannot 
create ZAP channel, although partticular span is reported up and active. 
I've also tried to connect loop between NT and TE port and call doesn't get 
through


I'd really appreciate if anyone has any advice on this problem, or any 
experience or working example for italian ISDN and particular Panasonic 
PBX.


Thanks in advance,

regards,

Rob.


I'm getting this :
Jun 22 16:25:13 VERBOSE[5536]: -- Accepting overlap voice call from
'432575513' to '000' on channel 0/2, span 4
Jun 22 16:25:21 VERBOSE[5536]: -- Executing
Dial(Zap/11-1,
ZAP/g1/38670613063|60) in new stack
Jun 22 16:25:21 VERBOSE[5536]: -- Called g1/38670613063
Jun 22 16:25:32 DEBUG[5536]: Queuing frame from PRI_EVENT_PROCEEDING on
channel 0/2 span 1
Jun 22 16:25:32 VERBOSE[5536]: -- Zap/2-1 is making progress passing it
to Zap/11-1
Jun 22 16:25:32 DEBUG[5536]: Received AST_CONTROL_PROGRESS on Zap/11-1
Jun 22 16:25:32 DEBUG[5536]: Dunno what to do with control type 15
Jun 22 16:25:34 VERBOSE[5536]: -- Channel 0/2, span 4 got hangup
Jun 22 16:25:34 DEBUG[5536]: Set option AUDIO MODE, value: ON(1) on Zap/2-1
Jun 22 16:25:34 DEBUG[5536]: Hangup: channel: 2 index = 0, normal = 33,
callwait = -1, thirdcall = -1
Jun 22 16:25:34 DEBUG[5536]: Not yet hungup...  Calling hangup once with
icause, and clearing call
Jun 22 16:25:34 DEBUG[5536]: disabled echo cancellation on channel 2
Jun 22 16:25:34 DEBUG[5536]: Set option TDD MODE, value: OFF(0) on Zap/2-1
Jun 22 16:25:34 DEBUG[5536]: Updated conferencing on 2, with 0 conference
users
Jun 22 16:25:34 DEBUG[5536]: Set option AUDIO MODE, value: OFF(0) on Zap/2-1
Jun 22 16:25:34 DEBUG[5536]: disabled echo cancellation on channel 2
Jun 22 16:25:34 VERBOSE[5536]: -- Hungup 'Zap/2-1'
Jun 22 16:25:34 DEBUG[5536]: Exiting with DIALSTATUS=CANCEL.



I have zapata.conf:
[channels]

switchtype = euroisdn

pridialplan = dynamic
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00
usecallingpres=yes

callerid=asreceived
overlapdial=yes
usecallingpres=yes




echocancel = yes
echocancelwhenbridged = yes
echotraining = 100

;---
; p2p TE mode (for connecting ISDN lines in point-to-point mode)
signalling = bri_cpe
; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode)
;signalling = bri_cpe_ptmp

context=from-isdn
group = 1

; S/T port 1-4 (first quadBRI, or lower ports of an octoBRI)
channel = 1-2
channel = 4-5
channel = 7-8
;channel = 10-11

;---

; p2p NT mode (for connecting an ISDN PBX in point-to-point mode)
;signalling = bri_net
; p2p NT mode (for connecting an ISDN PBX in point-to-multipoint mode)
signalling = bri_net_ptmp

context=from-pbx
group = 2
;overlapdial=no

; S/T port 5-8 (second quadBRI, or upper ports of an octoBRI)
channel = 10-11
;channel = 13-14
;channel = 16-17
;channel = 19-20
;channel = 22-23

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[Asterisk-Users] flash panel only works with IP address

[Ohad.Levy]

Hi,



It
seems that my flash panel only works when I specify my ip address and not the
host name.

I've
tried quite a few things (change host file, dns resolve, proxying.) but couldnt
get it to work.

Anyone
knows how to solve this?



Thanks,

Ohad








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Re: [Asterisk-Users] ASTCC not making calls

[Juan Luis Moyano]

Sorry 4 a.m. I'm kind of tired and I slipped a password. :S
Already changed it. Sorry!

Juan Luis Moyano wrote:
 Hi, im trying to setup ASTCC but I'm getting it difficult. I've
 correctly set up the mysql database astcc and added a brand, trunk,
 route and a card as follows:
 
 brands
 +--+--+--+--+--++--+--+
 | name | language | inc  | publishednum | did  | markup | days | fee  |
 +--+--+--+--+--++--+--+
 | FWD  | es   | 6| 4| 4|  0 | 30   |0 |
 +--+--+--+--+--++--+--+
 
 trunks
 +--+--+-+
 | name | tech | path|
 +--+--+-+
 | FWD  | IAX2 | 657XXX:[EMAIL PROTECTED] |
 +--+--+-+
 
 routes
 +-+---++-+-+--+
 | pattern | comment   | trunks | connectcost | includedseconds | cost |
 +-+---++-+-+--+
 | 4.  | FWD   | FWD|   0 |   0 |  150 |
 +-+---++-+-+--+
 
 -Added a card with $1 credit and using 'FWD' brand.
 
 extensions.conf
 ---
 [outbound-fwd]
 ;
 exten = _4.,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:1})
 exten = _4.,2,Hangup()
 
 iax.conf
 
 register = 657050:[EMAIL PROTECTED]
 
 
 The problem is that when, for example, I dial '4612' i get:
 
 -- Executing DeadAGI(IAX2/[EMAIL PROTECTED]/3, astcc.agi|21|612) in new 
 stack
 -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
 -- Playing 'digits/1' (language 'en')
 -- AGI Script astcc.agi completed, returning 0
 -- Executing Hangup(IAX2/[EMAIL PROTECTED]/3, ) in new stack
 
 and i hear allison saying I'm sorry that is not a recognized phone
 number, goodbye.
 
 Anyone knows what could be happening right here?
 
 Many thanks in advance.
 

-- 
Juan Luis Moyano
[EMAIL PROTECTED]

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[Asterisk-Users] Configuration Cisco FXO with asterisk

[craz sead]

Hi all,

Thanks anyway for helping me to install h323 and it
work i think. my problem now ..i dunno the
configuration from cisco and oh323.conf coz i have
tried several time ans still get error message from
asterisk voip-h323...failed so falling back to
'exten' s.

did anyone here have the configuration at the cisco
and the asterisk also. actualy i am using huawei ar 28
router but i will learn from cisco configuration if
somebody have here.

thanks 

Roy wish



 
Yahoo! Sports 
Rekindle the Rivalries. Sign up for Fantasy Football 
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[Asterisk-Users] Routing calls by trunk?

[Rick]

I am running [EMAIL PROTECTED] and have a digium
tdm04b (4 fxo)



The problem we have is we have 3 incoming pstn lines that step down from the telco,
then a spare line and a fax line. The office is now looking to add a second
0800 (free dial in NZ) to terminate to the spare line and the fax line. However
we need to route the calls to two separate locations in the office.



I tried to fake the callerid
to route by that in zaptel-auto.conf by setting callerid= number here then routing
the number that way. And that worked perfectly until I did a reboot and then it
wouldnt even ring extentions properly.



Is there any way to route via the zap line or setting up
some other fake callerid or fake did route or
something to get around this problem? Or do we have to get a ISDN line and a fxs card?






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Re: [Asterisk-Users] Error on installing oh323 on asterisk

[craz sead]

try to clean and retry co configure strp by step dont
forget to mark your step. I already try and installed
my problem was the same with you so ...i tri  to use
another version. I tri 0.7.1 at the first time then
0.7.0 also have a same error but then work after i use
0.6.5. my machine runing RH9

good luck
roy
--- Charles Huang [EMAIL PROTECTED] wrote:

 I'm following the instruction from João Amaro from
 the
 page
 

http://lists.digium.com/pipermail/asterisk-users/2005-February/090752.html
 
 Everything went fine until I run the 'make' command
 under asterisk-oh323-0.6.5.  I got the error message
 
 chan_oh323.c:5220: too many arguments to function
 `ast_channel_register'
 
 I have attached the error message.  I'm running
 asterisk CVS HEAD version, would that be the cause
 of
 the problem?
 
 Any help would greatly appricated.
 
 Thanks,
 Charles
 
 
 
 # make
 for x in wrapper asterisk-driver; do make -C $x
 build
 || exit 1 ; done
 make[1]: Entering directory
 `/root/asterisk-oh323-0.6.5/wrapper'
 ./check_ver /root/pwlib pwlib
 ./check_ver /root/openh323 openh323
 ar rc liboh323wrap_s.a wrapper_misc.o
 asteriskaudio.o
 wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o
 wrapgkserver.o
 make[1]: Leaving directory
 `/root/asterisk-oh323-0.6.5/wrapper'
 make[1]: Entering directory
 `/root/asterisk-oh323-0.6.5/asterisk-driver'
 gcc -Wall -pipe -Wall -Wstrict-prototypes
 -Wmissing-prototypes -Wmissing-declarations
 -D_REENTRANT -D_GNU_SOURCE
 -I/usr/src/asterisk/include
 -I../wrapper -g -c -o chan_oh323.o chan_oh323.c
 chan_oh323.c:37:34: asterisk/channel_pvt.h: No such
 file or directory
 chan_oh323.c: In function `oh323_exception':
 chan_oh323.c:1145: dereferencing pointer to
 incomplete
 type
 chan_oh323.c: In function `oh323_indicate':
 chan_oh323.c:1326: dereferencing pointer to
 incomplete
 type
 chan_oh323.c: In function `oh323_digit':
 chan_oh323.c:1388: dereferencing pointer to
 incomplete
 type
 chan_oh323.c: In function `oh323_text':
 chan_oh323.c:1410: dereferencing pointer to
 incomplete
 type
 chan_oh323.c: In function `oh323_call':
 chan_oh323.c:1434: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:1453: structure has no member named
 `callerid'
 chan_oh323.c:1455: structure has no member named
 `callerid'
 chan_oh323.c:1457: structure has no member named
 `callerid'
 chan_oh323.c: In function `oh323_hangup':
 chan_oh323.c:1613: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:1721: dereferencing pointer to
 incomplete
 type
 chan_oh323.c: In function `oh323_read':
 chan_oh323.c:1749: dereferencing pointer to
 incomplete
 type
 chan_oh323.c: In function `oh323_write':
 chan_oh323.c:2050: dereferencing pointer to
 incomplete
 type
 chan_oh323.c: In function `oh323_answer':
 chan_oh323.c:2242: dereferencing pointer to
 incomplete
 type
 chan_oh323.c: In function `oh323_fixup':
 chan_oh323.c:2286: dereferencing pointer to
 incomplete
 type
 chan_oh323.c: In function `ast_oh323_new':
 chan_oh323.c:2518: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2527: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2529: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2536: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2537: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2538: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2539: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2540: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2541: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2542: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2543: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2544: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2545: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2546: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2547: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2548: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2549: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2550: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2551: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2552: dereferencing pointer to
 incomplete
 type
 chan_oh323.c:2579: structure has no member named
 `dnid'
 chan_oh323.c:2589: structure has no member named
 `callerid'
 chan_oh323.c:2590: structure has no member named
 `callerid'
 chan_oh323.c:2591: structure has no member named
 `callerid'
 chan_oh323.c:2596: structure has no member named
 `callerid'
 chan_oh323.c:2597: structure has no member named
 `callerid'
 chan_oh323.c:2598: structure has no member named
 `callerid'
 chan_oh323.c:2600: structure has no member named
 `callerid'
 chan_oh323.c:2605: structure has no member named
 `callerid'
 chan_oh323.c:2606: structure has no member named
 `callerid'
 chan_oh323.c:2608: structure has no member named
 `callerid'
 chan_oh323.c:2610: structure has no member named
 `callerid'
 chan_oh323.c:2614: structure 

[Asterisk-Users] SIP DID routing

[snacktime]

How do you get the called number on incoming SIP calls?  I've never
had multiple DID's via SIP from one provider before and somehow I
never realized that with IAX it just works, and SIP is different.

If I don't set an extension in the register command the incoming
invite has sip:[EMAIL PROTECTED] in the To field.  Now if I have multiple
DID's that I want routed to different extensions, what's the solution?
 Is there a SIP header that is normally used to pass the called number
in?

Hope that makes sense..

Chris
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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Sergio Chersovani]

Armin Schindler ha scritto:


Have you planned to integrate some echo cancel feature ?
   


Echo cancelling (if the card supports it) is already implemented.
 


I think he was talking about the software echo suppressor


As far as I know the Eicon Diva Server cards are the only cards supporting
echo cancel via onboard DSPs.
 


AVM active cards do not support it?

Sergio
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[Asterisk-Users] Loosing hair on connecting Panasonic PBX- *- Euroisdn Italy

[Justin Newman]

 Date: Thu, 23 Jun 2005 08:50:50 +0200
 From: Robert Rozman [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *
 - Euroisdn Italy

 I'm pulling my hair down and getting bold :-) . I have Asterisk
between
 Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff
 Asterisk)

Plenty of experience with the Panasonics, but not the EuroISDN. Contact
me offline if you have KXTD816 questions.

Regards,

Justin
[EMAIL PROTECTED]
Newman Telecom, Inc.

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Re: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy

[Peter Svensson]

On Thu, 23 Jun 2005, Robert Rozman wrote:

 I'm pulling my hair down and getting bold :-) . I have Asterisk between
 Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff
 Asterisk)
 
 I'm trying to do just plain transfer of call from pbx to ISDN through
 Asterisk...
 
 It seems like PBX hangsup, when call is progressing with no apparent reason.
 I'd kindly ask for any advice or some working example for this
 
 On isdn side I also have a problem. Asterisk quite often says that it cannot 
 create ZAP channel, although partticular span is reported up and active. 
 I've also tried to connect loop between NT and TE port and call doesn't get 
 through
 
 I'd really appreciate if anyone has any advice on this problem, or any 
 experience or working example for italian ISDN and particular Panasonic 
 PBX.

Look at the logs from a pri intense debug span X to see what causes the 
lines to be hung up. 

Make sure progress detection is disabled.

Peter


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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Elmar Haneke]

Have you planned to integrate some echo cancel feature ?


Besides the Eicon-CAPI feature there is an echosquelch in the driver.

Elmar
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[Asterisk-Users] Re: ZapRAS

[Daniel Nyström]

Daniel,
  we have the same problem when our PRI line drops and Zapras has to
reconnect.  You will also notice that the pppd process does not die
when Zapras does and the ppp connection cannot re-establish itself.
What we normally do is restart asterisk and then kill the pppd process
with the command:  killall -9 pppd.
How much of your resources does the pppd process take up when Zapras 
executes?

Maybe try making pppd a lower priority than asterisk.
Double check your configuration and make sure that you've done all
that you needed to in order to properly setup zapras.  go to digiums
website and look at documentation.

Thanks for your reply. Now I retried ZapRAS and, when not in panic, it 
only required an restart of Asterisk to recover the sound.

But the pppd process was gone and nothing was taking more than 0.2% CPU.
Seems like ZapRAS destroy something within Asterisk then?
Regarding ZapRAS, I don't even get it to work at all. More on that later.
How are your ZapRAS configured? Do you connect to an ISP through it?
I've tried the new app_pppd as well, and that one connects. But the 
connetion dies within a few second (just stop responding/working). See 
my other earlier posts about that.
I've gone through Digiums website more than once to get this to work, 
but there seems to be only a README-file in the ftp area for information.

Which page did you mean?
I've been trying for month to get this working. And alot of questions 
have been posted to the mailing list from me. Even the IRC channels has 
been visited.
But no one seems to have this problem, or are even using the 
ZapRAS/app_pppd.

I request any tips regarding ISDN dialup Internet access.
Thanks!
--
Daniel
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[Asterisk-Users] Cisco 7960 firmware upgrade promblems

[Patrick Lidstone (Personal E-mail)]

I have a second-hand 7960 which I am attempting to upgrade to use a SIP
image.

The phone currently has a firmware release which doesn't seem to be listed
in Cisco docs - P003AM30. On reboot, it finds the tftp server and requests
the firmware image listed in OX79XX.txt correctly, displaying Upgrading
Software on the screen. It then continues to re-request the same image from
the tftp server at 10s intervals indefinitely. What am I doing wrong?

Thanks
Patrick

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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Massimo De Nadal]

Yes, I know. I was meaning the software thing.
Diva server cancels echo via dsp only with new revisions boards (older 
boards are not able to run newer drivers with echo cancellation).

Fritz cards don't cancel echo anyway.
And echo squelch is only a trick that doesn't really solve the problem.

Is it possible to port zap echo cancelor to different channels like 
chan_capi ?



Armin Schindler ha scritto:


On Thu, 23 Jun 2005, Massimo De Nadal wrote:
 


Have you planned to integrate some echo cancel feature ?
   



Echo cancelling (if the card supports it) is already implemented.
As far as I know the Eicon Diva Server cards are the only cards supporting
echo cancel via onboard DSPs.

Armin

 




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[Asterisk-Users] Welltech 4 Port FXO - Asterisk

[Erik Espinoza]

Hey does anyone know how to configure the 4 port fxo to work with
Asterisk? I have the updated firmware. All ports register, however
incoming calls are never handled properly by the fxo. I even set
hotline.

Does anyone have any info, or perhaps a web site?
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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Massimo De Nadal]

Sergio Chersovani wrote:

As far as I know the Eicon Diva Server cards are the only cards 
supporting

echo cancel via onboard DSPs.
 


AVM active cards do not support it?


No.
Avm active cards are basically multi fritz boards running the same 
firmware onboard instead of  charging pc cpu.
They are surely more stable than fritz cards, but offer the same 
features (even with more channels).




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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Armin Schindler]

On Thu, 23 Jun 2005, Massimo De Nadal wrote:
 Yes, I know. I was meaning the software thing.
 Diva server cancels echo via dsp only with new revisions boards (older boards
 are not able to run newer drivers with echo cancellation).

Which boards don't support that? If DSPs on board, echo-cancel should be 
available.

 Fritz cards don't cancel echo anyway.
 And echo squelch is only a trick that doesn't really solve the problem.
 
 Is it possible to port zap echo cancelor to different channels like chan_capi
 ?

Yes, that should be possible. 
But I don't think a channel driver (and each channel driver) should do that 
on its own. Software echo cancelling belongs in a common part of Asterisk.

Armin
 
 Armin Schindler ha scritto:
 
  On Thu, 23 Jun 2005, Massimo De Nadal wrote:
  
  
   Have you planned to integrate some echo cancel feature ?
   
   
  
  Echo cancelling (if the card supports it) is already implemented.
  As far as I know the Eicon Diva Server cards are the only cards supporting
  echo cancel via onboard DSPs.
  
  Armin
  
  
  
 
 
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[Asterisk-Users] avm c2 correct configuration for two p2p lines

[Simone Cittadini]

I have an asterisk box connected to two isdn lines via an AVM c2 card,
the ISDN boxes have the 0227006XXX and 0227007XXX numbers, and are
configured both p2p, with the first one as file-leader.
(I don't know if file-leader is the correct term, it's a literal
translation from the italian term capofila, in other words 
0227006XXX is our real number, and when the 2 B channels are both used
it should forward the call to the second line)

What's the correct configuration for the capi channel/driver ?

this is my current :

/etc/capi.conf

# card  fileproto   io  irq mem cardnr  options
c2  c2.bin  DSS1-   -   -   1   p2p
c2  -   DSS1-   -   -   2

/etc/asterisk/capi.conf 
(surely wrong, it refers to a previous conf as two separate lines)

[general]
nationalprefix=0
internationalprefix=0039
rxgain=0.8
txgain=0.8

[interfaces]
msn=0227006XXX
incomingmsn=*
controller=1
softdtmf=1
context=incoming
echosquelch=1
mode=immediate
callgroup=1
devices=2

msn=0227007XXX
incomingmsn=*
controller=2
softdtmf=1
context=incoming
echosquelch=1
mode=immediate
callgroup=1
devices=2

Sorry if it's a basic question, but I can't find a detailed and complete
example googling, and being the *box a production machine I can't
neither procede with try and see if it works method.


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Re: [Asterisk-Users] MeetMe Problems

[Waldo Rubinstein]

Doing further tests, I discovered that I can successfully do MeetMe  
on both server B and server C, AS LONG AS all parties are SIP  
extensions registered on the same server (e.g. server B or server C).  
However, when I try to bring a call from server A into a MeetMe in  
server B or server C, that's when the problem shows up. Hope this  
helps anyone who can help me.


Thanks,
Waldo

On Jun 22, 2005, at 3:06 PM, Waldo Rubinstein wrote:

I decided to test a similar scenario against another machine  
(server C). This machine behaves in a similar way as server B. It  
is also running on Gentoo. When I try to transfer a call into a  
conference room, it fails. Below is the CLI output of an inbound  
call coming from server A into server C, ringing extension SIP/ 
3211. Once answered, I try transferring to MeetMe room 0211 which  
fails.


bacardi init.d # aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
== 
===

Connected to Asterisk 1.0.7 currently running on bacardi (pid = 10925)
Verbosity was 0 and is now 10
-- Accepting AUTHENTICATED call from 10.0.10.9, requested  
format = 4, actual format = 4
-- Executing Dial(IAX2/[EMAIL PROTECTED]/16384, SIP/3211) in  
new stack

-- Called 3211
-- SIP/3211-1bd8 is ringing
-- SIP/3211-1bd8 answered IAX2/[EMAIL PROTECTED]/16384
-- Started music on hold, class 'default', on IAX2/ 
[EMAIL PROTECTED]/16384

-- Executing MeetMe(SIP/3211-e3c6, 0211|qM) in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '0211'
-- Started music on hold, class 'default', on SIP/3211-e3c6
-- Stopped music on hold on SIP/3211-e3c6
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16384
Jun 22 14:59:40 WARNING[10952]: app_meetme.c:667 conf_run: Error  
getting conference

-- Hungup 'Zap/pseudo-1721629866'
  == Spawn extension (default, 0211, 1) exited non-zero on 'IAX2/ 
[EMAIL PROTECTED]/16384'

-- Hungup 'IAX2/[EMAIL PROTECTED]/16384'
-- Attempting native bridge of SIP/3211-e3c6ZOMBIE and SIP/ 
3211-1bd8
Jun 22 14:59:40 WARNING[10951]: rtp.c:1365 ast_rtp_bridge: Can't  
find native functions for channel 'SIP/3211-e3c6ZOMBIE'
Jun 22 14:59:40 WARNING[10951]: channel.c:2634 ast_channel_bridge:  
Private bridge between SIP/3211-e3c6ZOMBIE and SIP/3211-1bd8 failed
  == Spawn extension (default, 3211, 1) exited non-zero on 'SIP/ 
3211-e3c6ZOMBIE'


I don't know if it has anything to do with the ZOMBIE channel.  
lsmod shows that both zaptel and ztdummy are loaded. Any ideas?


Thanks,
Waldo

On Jun 22, 2005, at 10:41 AM, Waldo Rubinstein wrote:


Absolutely. Here is the CLI output. I made two attempts. First, I  
dialed inbound into an extension and then tried using meetme room  
0201 from Server B, which didn't work. Then I dialed inbound into  
the same extension and then tried using meetme room 0215 which  
resides in Server A. Note that all inbound calls come into Server  
A, for it has the Digium card.


SERVER A
=

gateway0:~# aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk CVS-Nv1-0-7-06/01/05-01:27:25, Copyright (C) 1999-2004  
Digium.

Written by Mark Spencer [EMAIL PROTECTED]
= 

Connected to Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 currently  
running on gateway0 (pid = 2653)

Verbosity is at least 10
-- Starting simple switch on 'Zap/1-1'
-- Executing NoOp(Zap/1-1, Inbound Call - 3211 - ) in new  
stack

-- Executing Dial(Zap/1-1, IAX2/corona/3211||r) in new stack
-- Called corona/3211
-- Call accepted by 10.0.10.13 (format ulaw)
-- Format for call is ulaw
-- IAX2/corona/16386 is ringing
-- IAX2/corona/16386 answered Zap/1-1
-- Hungup 'IAX2/corona/16386'
  == Spawn extension (inbound, 3211, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing NoOp(Zap/1-1, Inbound Call - 3211 - ) in new  
stack

-- Executing Dial(Zap/1-1, IAX2/corona/3211||r) in new stack
-- Called corona/3211
-- Call accepted by 10.0.10.13 (format ulaw)
-- Format for call is ulaw
-- IAX2/corona/16388 is ringing
-- IAX2/corona/16388 answered Zap/1-1
  == Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '0215'
-- Started music on hold, class 'default', on IAX2/ 
[EMAIL PROTECTED]/16390

-- Hungup 'Zap/31-1'
-- Hungup 'IAX2/corona/16388'
  == Spawn extension (inbound, 3211, 2) exited non-zero on 'Zap/1-1'
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16390
-- Hungup 'Zap/pseudo-1262753463'
  == Spawn extension (meetme, 0215, 1) exited non-zero on 'IAX2/ 
[EMAIL 

Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Massimo De Nadal]

Armin Schindler ha scritto:

Which boards don't support that? If DSPs on board, echo-cancel should be 
available.
 

I have in my hands right now  a DIVA Server BRI-2M-PCI (not the 2.0 
version) which own its dsp but doesn't echo cancel, due to old capi 
drivers which don't support this feature. Newer eicon drivers won't run 
on this board.


Yes, that should be possible. 
But I don't think a channel driver (and each channel driver) should do that 
on its own. Software echo cancelling belongs in a common part of Asterisk.


 

I strongly agree. But asterisk doesn't seem to work this way. Zap 
channel has it's own echo cancel engine. Other channels don't.

This is so sad :-(
Why not implement a really common echo cancel api usable from any channel ??



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[Asterisk-Users] Music on Hold Choppy

[Mahmoud Badran]

Hello all

i am using asterisk 1.07 with mpg123-0.59r but still i get very choppy sounds, any suggestions?





extensions.conf
---
exten = 444,1,WaitMusicOnHold(120)




modules.conf

[modules]
autoload=yes

load = chan_modem.so
load = res_musiconhold.so

noload = chan_alsa.so
noload = chan_oss.so

[global]
chan_modem.so=yes



musiconhold.conf
-
[classes]
default = quietmp3:/var/lib/asterisk/mohmp3






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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Armin Schindler]

On Thu, 23 Jun 2005, Massimo De Nadal wrote:
 Armin Schindler ha scritto:
 
  Which boards don't support that? If DSPs on board, echo-cancel should be
  available.
  
  
 I have in my hands right now  a DIVA Server BRI-2M-PCI (not the 2.0 version)
 which own its dsp but doesn't echo cancel, due to old capi drivers which don't
 support this feature. Newer eicon drivers won't run on this board.

Do you talk about the driver package from Eicon? What about the driver from
melware.net ?
 
  Yes, that should be possible. But I don't think a channel driver (and each
  channel driver) should do that on its own. Software echo cancelling
  belongs in a common part of Asterisk.
  
  
  
 I strongly agree. But asterisk doesn't seem to work this way. Zap channel has
 it's own echo cancel engine. Other channels don't.
 This is so sad :-(
 Why not implement a really common echo cancel api usable from any channel ??

Exactly!
I'm not familiar with the Asterisk API, but it could be some
plugin like res_* ... 

Maybe this belongs to the Asterisk-Dev list.

Armin
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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Klaus-Peter Junghanns]

  
   Yes, that should be possible. But I don't think a channel driver (and each
   channel driver) should do that on its own. Software echo cancelling
   belongs in a common part of Asterisk.
   
   
   
  I strongly agree. But asterisk doesn't seem to work this way. Zap channel 
  has
  it's own echo cancel engine. Other channels don't.
  This is so sad :-(
  Why not implement a really common echo cancel api usable from any channel ??
 
 Exactly!
 I'm not familiar with the Asterisk API, but it could be some
 plugin like res_* ... 
 
 Maybe this belongs to the Asterisk-Dev list.
 
 Armin

I strongly disagree. :-) You dont want to do echo cancelation in
userspace. Especially not on a non-realtime operating system.
To make echo cancelation work it has to be as close to the line
interface as possible. Also the frames have to be as small
as possible. This rules out capi pretty much.

best regards

Klaus
--
Klaus-Peter Junghanns

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[Asterisk-Users] Server Load/Capacity

[Waldo Rubinstein]

I'm trying to figure out how much call load I can put on a Dual Xeon  
2.4 Ghz Asterisk server acting strictly as an IAX2 call director, as  
show in the diagram below.


The idea is that I have N number of gateway asterisk servers  
connected to the PSTN using T1 Digium boards. Then, I have M number  
of servers where my agents and/or telephone extensions (whether they  
are IAX or SIP hard/soft phones). What I'm trying to accomplish is  
put a server in between these two groups of machines which will  
simply be able to intelligently route calls in either direction.  
This call director server will only use IAX2 (ulaw) to minimize any  
transcoding and alleviate load.


Under this scenario, does anyone have any idea how many calls this  
call director server may be able to handle/direct?


Thanks,
Waldo

Please see http://200.62.6.122/pastedGraphic.pdf for the diagram
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[Asterisk-Users] Least Cost Routing

[Daniel ANDRE]

Hello,

I am searching for a working solution for Least Cost Routing usable in 
France with asterisk. Does Anyone have any tip?


Regards,

Daniel ANDRE

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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Armin Schindler]

On Thu, 23 Jun 2005, Klaus-Peter Junghanns wrote:
Yes, that should be possible. But I don't think a channel driver (and 
each
channel driver) should do that on its own. Software echo cancelling
belongs in a common part of Asterisk.

   I strongly agree. But asterisk doesn't seem to work this way. Zap channel 
   has
   it's own echo cancel engine. Other channels don't.
   This is so sad :-(
   Why not implement a really common echo cancel api usable from any channel 
   ??
  
  Exactly!
  I'm not familiar with the Asterisk API, but it could be some
  plugin like res_* ... 
  
  Maybe this belongs to the Asterisk-Dev list.
  
  Armin
 
 I strongly disagree. :-) You dont want to do echo cancelation in
 userspace. Especially not on a non-realtime operating system.
 To make echo cancelation work it has to be as close to the line
 interface as possible. Also the frames have to be as small
 as possible. This rules out capi pretty much.

If you don't want echo-canceling in user-space, then neither Asterisk nor
any chan_* plugin should do it.

I don't know the zap channel code, but does the zap echo-cancel-code is 
inside a kernel module?
If yes, then I have to disagree here. Something like 'playing' with 
audio-data is nothing the kernel should be concerned with.
This belongs in user-space and if you need realtime, then you should use a 
realtime OS or use RT-scheduling. Just putting such a code into kernelspace 
is a bad idea.

So the correct way is either the hardware supports it or the 
application knows what to do with the data received, like DTMF.

Armin
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Re: [Asterisk-Users] Question on bridged calls

[Rich Adamson]

 If I connect to a provider using iax, and that provider connects to
 his provider using only sip,  the provider I am connecting to isn't
 going to be able to bridge the call and drop out of the media stream
 correct?

Correct.

 If I'm understanding how bridging works, you lose the ability to have
 the media stream going directly between the two endpoints of the call
 with most of the providers out there if you use iax, unless the
 provider has their own tdm network.

Correct. However, you can probably guess that most sip/iax providers
also use canreinvite=no anyway. Why? Because of the number of customers
that have some sort of inexpensive firewall/nat box that would cause
an audio failure several seconds into a call, driving their support
costs skyhigh. You've been around this list long enough to have seen
a high number of * implementors not even understand that, so how 
would you expect a less-technical itsp customer to understand that on
initial account setup?

 Is this correct or am I completely missing something?

You're also assuming that most itsp's use asterisk, and that is not a
valid assumption.


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Re: [Asterisk-Users] Server Load/Capacity

[tim panton]

On 23 Jun 2005, at 10:48, Waldo Rubinstein wrote:

I'm trying to figure out how much call load I can put on a Dual  
Xeon 2.4 Ghz Asterisk server acting strictly as an IAX2 call  
director, as show in the diagram below.


The idea is that I have N number of gateway asterisk servers  
connected to the PSTN using T1 Digium boards. Then, I have M number  
of servers where my agents and/or telephone extensions (whether  
they are IAX or SIP hard/soft phones). What I'm trying to  
accomplish is put a server in between these two groups of machines  
which will simply be able to intelligently route calls in either  
direction. This call director server will only use IAX2 (ulaw) to  
minimize any transcoding and alleviate load.


Under this scenario, does anyone have any idea how many calls this  
call director server may be able to handle/direct?




At Astricon the man from Signate showed some benchmark results which  
indicated
a 'stock' PC server could do 122 ulaw SIP passthrough calls at  
acceptable

call quality.
Their own-brand servers can do  2k (If I remember right).

However I think you should look into Dundi - used correctly with a
clear dialplan you may be able to get rid of the director and
have a cloud of dundi peers instead.

Tim.
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Re: [Asterisk-Users] Music on Hold Choppy

[jurczak]

do you have VAD enabled?

On Thu, 23 Jun 2005 12:23:15 +0300, Mahmoud Badran wrote
 Hello all
 
 i am using asterisk 1.07 with mpg123-0.59r but still i get very 
 choppy sounds, any suggestions?
 
 extensions.conf
 ---
 exten = 444,1,WaitMusicOnHold(120)
 
 
 
 modules.conf
 
 [modules]
 autoload=yes
 
 load = chan_modem.so
 load = res_musiconhold.so
 
 noload = chan_alsa.so
 noload = chan_oss.so
 
 [global]
 chan_modem.so=yes
 
 
 
 musiconhold.conf
 -
 [classes]
 default = quietmp3:/var/lib/asterisk/mohmp3
 




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Re: [Asterisk-Users] TDM400P Channel Group

[Rich Adamson]

 Shouldn't [Asterisk] be smart enough to go to Zap/4 as the only available
 port in the group [with a live trunk]?
   
 
 Adam Goryachev wrote:
 
 No, asterisk doesn't do dialtone detection.
 
 But this isn't an issue of dialtone detection but one of detecting 
 battery (a much easier task). Shouldn't (doesn't) Asterisk check for 
 battery on an FXO port before using the port? Shouldn't this be an 
 option? (Sorry but I haven't the time to look at the code not to mention 
 think about adding this if it isn't already there.)

Asterisk does not look for battery. Mark made a change to the code
about a year ago (I think shortly after the TDM card was released)
to detect it (since the TDM chipset has that capability). For whatever
reason, it created problems for some users and the change was
enclosed in ifdef's. I don't know if they remain in the code or yet
right now.


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RE: [Asterisk-Users] Cisco 7960 firmware upgrade promblems

[Geoff Manning]

 I have a second-hand 7960 which I am attempting to upgrade to 
 use a SIP
 image.
 
 The phone currently has a firmware release which doesn't seem 
 to be listed
 in Cisco docs - P003AM30. On reboot, it finds the tftp server 

Here's how I performed the upgrade:

Downgrade from the stock P003AM30 to POS30203

Upgrade to version 5.1 (first signed binary firmware)

Upgrade to version 7.1 * (most recent version? maybe 7.4?)

* When upgrading to 7.1 there is a typo in the OS79XX file, it will say
P00x change it to P0Sxgreat typo by Cisco.


Check the comments on this wiki page: 

http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx?page=Asterisk%20phon
e%20cisco%2079xxcomments_threshold=0comments_offset=0comments_sort_mode=c
ommentDate_desccomments_maxComments=10comments_parentId=353#threadId358

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Re: [Asterisk-Users] Asterisk in India?

[Alistair Cunningham]

Matt,

I've done it several times for customers in India using E1s with 
EuroISDN and:


loadzone = nl
defaultzone = nl

Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/


Matthew Gibson wrote:

Hi,

Is anyone successfully using Asterisk in India hooked up to the PSTN?

I have tried defaultzone=us and no tones would work at all when
calling the IVR,
but if i set defaultzone=uk most but not all of the buttons work.

Does anyone have any tips or tricks for getting TDM / PSTN connectivity
from asterisk
in India?

Tia,
Matt

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[Asterisk-Users] MGCP Groups

[Mark Johnson]

I am looking into using a Cisco T1 device that uses MGCP.  Asterisk is 
talking to it fine, but I am having a hard time figuring out how to 
handle channel grouping like Zap does.  With Zap, I can take channels 
1-23 and make a group g1 out of it and then simply dial Zap/g1.  Does 
MGCP have this type of functionality?  Everything I've tried points to no...


Thanks!

Mark
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[Asterisk-Users] Asterisk with failover and load balancing

[Mohamed A. Gombolaty]

Dear All,
I was searching voip-info for Failover and load balancing for
Asterisk, my goal here is to have a system where the SIP traffic is being
divided on five central servers with Asterisk on, and if an asterisk server
fails another asterisk server will assume it's place , from my readings
I have cited the following options:
1- SER + ASTERISK with Domain SRV
2- vovida Load balancer (I am not happy about this one it's old I can't
compile on new OSand it's mailing list is useless and development
seems to have stopped )
I hope any one could enlighten me with his experience if he has done
such a thing and which can be a better option or if there is something
I am still missing.
--
Thx
MAG

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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Klaus-Peter Junghanns]

Am Donnerstag, den 23.06.2005, 12:41 +0200 schrieb Armin Schindler:

  I strongly disagree. :-) You dont want to do echo cancelation in
  userspace. Especially not on a non-realtime operating system.
  To make echo cancelation work it has to be as close to the line
  interface as possible. Also the frames have to be as small
  as possible. This rules out capi pretty much.
 
 If you don't want echo-canceling in user-space, then neither Asterisk nor
 any chan_* plugin should do it.
 
 I don't know the zap channel code, but does the zap echo-cancel-code is 
 inside a kernel module?

Yes, sir.

 If yes, then I have to disagree here. Something like 'playing' with 
 audio-data is nothing the kernel should be concerned with.
 This belongs in user-space and if you need realtime, then you should use a 
 realtime OS or use RT-scheduling. Just putting such a code into kernelspace 
 is a bad idea.

What is so bad about playing with audio-data in kernel space?
If you play with echo cancelation in user space you will need
to de-jitter the audio first introducing more and more latency, so
your echo cancelation becomes way more computationally expensive.

 
 So the correct way is either the hardware supports it or the 
 application knows what to do with the data received, like DTMF.
 

Why should the application have to worry about things like echo
cancelation? Zaptel is not only used by Asterisk but also by other
projects. With EC in kernel space (which gets switched on and off
by userspace) there is no need to reinvent the EC-wheel for every
project.

Klaus


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Re: [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?

[Steve Underwood]

Peter Svensson wrote:


On Tue, 21 Jun 2005, Leandro Morgado wrote:

 


Steve Underwood wrote:
   


Robert Rozman wrote:
 


I'm getting unreliable dtmf recognition (it works fine for 4-5
digits, errors (duplicates) on more), when transferred inband from
gsm gateway to NT port of quadbri under bristuffed Asterisk.
   



We get these quite often. If there is any line noise asterisk will 
interpret it as the end of a digit and then detect the same digit again. 
We are connected to the pstn via isdn. The problem is with calls where the 
dtmf tones are a bit unclean, i.e. too much energy is in the overtones. 
Clean dtmf tones seem to be much more resistant to line noise.


Out other systems are more accepting of slightly off-spec dtmf tones.
 

People often make claims like this, but never seem able to back them up. 
They usually turn out to be a wildy set receive gain, wrong codecs, or 
something equally screwed up. Can you quote chapter and verse for a real 
problem?


The DTMF detector in * is one of the most robust around.

Regards,
Steve

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[Asterisk-Users] Zap lines-inbound,outbound calls intersect

[Colin E. McDonald]

I have been having a problem for a while where an internal user will be
calling out via SIP through the Asterisk box which has a TDM400 and when
they pick up they have an inbound caller on the line. The lines then
become bridged and they stay that way until you do a soft hangup on
one of the lines. Is anybody else running into this problem?


Thanks

Colin
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Re: [Asterisk-Users] FXS interfaces

[Rich Adamson]

  I got current stable release in CVS repository, and I think that Ok.
  See below:
  
  /var/log/messages 
  Jun 22 17:04:35 darthvaden kernel: PCI: Found IRQ 9 for device 02:09.0
  Jun 22 17:04:35 darthvaden kernel: PCI: Sharing IRQ 9 with 00:1f.5
  Jun 22 17:04:35 darthvaden kernel: Freshmaker version: 71
  Jun 22 17:04:35 darthvaden kernel: Freshmaker passed register test
  Jun 22 17:04:35 darthvaden kernel: Module 0: Installed -- AUTO FXS/DPO
  Jun 22 17:04:35 darthvaden kernel: Module 1: Installed -- AUTO FXS/DPO
  Jun 22 17:04:35 darthvaden kernel: Module 2: Installed -- AUTO FXO (FCC
  mode)
  Jun 22 17:04:35 darthvaden kernel: Module 3: Installed -- AUTO FXO (FCC
  mode)
  Jun 22 17:04:35 darthvaden kernel: Found a Wildcard TDM: Wildcard
  TDM400P REV E/F (4 modules)
  Jun 22 17:04:35 darthvaden kernel: Registered tone zone 0 (United States
  / North America)
 
 Congratulations.  What were you using prior your pull from CVS?  Maybe
 something old that didn't recognize the the TDM400P and its daughters?
  
  But all ports are green! 
 
 Really?  Maybe they aren't making the RED FXO cards anymore.  You should
 look at them carefully for p/n differences and not rely on colors.  The
 zapel driver tells you what you need to know too.
  
  p1 - green 
  p2 - green
  p3 - green
  p4 - green
 
 I wonder if Digium will update their website?  It's got a strong
 commitment to red FXO modules in the graphics.

I'm not the OP, but it would appear (based on postings within the last
week) that yet _another_ version of the TDM modules/card is being
sold by digium, and the Stable zaptel drivers (and probably many
cvs-head working systems) are using card drivers that don't support
these modules/card. 

Driver version handling still seems to be an area that could be
addressed far better then what it has been in the past.


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[Asterisk-Users] Management: Reload performace Realtime performance ?

[René Ott]

Hello,

I am interested in some management-performance issues:

1st Scenario:

A management tool (for example a webbased one) has the following process:

- write in database
- read with script (for example perl) data from db and write conf files
- reload asterisk

I was reading around in the mailing lists and people say reloading is 
stable. Now this tool has to manage 1000 clients so the conf files are 
quite big and reloading needs some time. What happens if a call comes in 
during that reload time ?
How is the performance in general of the process described above 
(assumed the used hardware is not under- and not overdimensioned), can 
such a tool easily handle 1000 clients ?

Does somebody use a similar tool with many clients ?

2nd Scenario

A management tool has the following process:

- write in database
- asterisk reads with realtime the conf

Somewhere in the mailing lists someone said that the realtime uses many 
database queries. If there are also 1000 clients to manage, this should 
lead to lots of database queries.
And again the questions, how is the performance in general of the 
realtime process (again normal hardware assumed) ? Can realtime handle 
1000 clients ?

Does somebody use it with lots of clients ?

Thanks in advance for the answers.

Best regards,
René
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Re: [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?

[Steve Underwood]

Leandro Morgado wrote:


Steve Underwood wrote:

 


Robert Rozman wrote:

   


Hi,

I'm getting unreliable dtmf recognition (it works fine for 4-5
digits, errors (duplicates) on more), when transferred inband from
gsm gateway to NT port of quadbri under bristuffed Asterisk.

Since Asterisk is claimed to have good dtmf recognizer, I suspect
there are some settings to workarouned... I've tried dtmf relax, but
didn't help, so I suspect gain settings

Is there any other possible cause of unreliable dtmf inband
recognition ? Where can I set gain on voice channel (I guess majority
of settings under bristuff in zaptel.conf are dummy) ?

Any other advice on this problem or similar experience ?

Thanks in advance,
 


I kind of amazed if works at all when getting DTMF out of a GSM phone.
You really shouldn't expect it to.
   



We have sucessfully read incoming DTMF from:

a) Nokia32 Analog GSM connected to TDM400 (had to use relaxdtmf with
chan_zap)
b) Ateus BRI ISDN GSM connected to AVM Fritz (had to patch chan_capi
0.3.5 to support relaxdtmf)


Question (I'm from a software eng. background, not telco):
What would be the reason for not receiving DTMF from a GSM
phone/gateway? Do you have the time to explain why? (I'm really
interested in learning :)
 

The low bit rates codecs used for GSM cannot carry DTMF without 
seriously corrupting it. To allow DTMF to be sent to things like IVRs 
the GSM protocol alows the handset to send a message to the basestation 
to tell it to send a DTMF digit to the wireline network. The timing of 
these digits is completely unrelated to the user pressing keys on the 
phone, so any input method based on DTMF timing does not work. There is 
usually no need for DTMF to be sent reliably from a basestation to a 
handset, so the GSM protocol makes no provision for it.


DTMF might or might not get through. The signals are quite distorted, 
but they are still present. Its pretty hit and miss. GSM 06.10 (the 
original GSM codec, which things like Asterisk also support) tends to 
work quite a bit of the time. The newer codecs (EFR, half-rate and AMR) 
tend not to.


Regards,
Steve

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RE: [Asterisk-Users] Zap lines-inbound,outbound calls intersect

[Betül Gözlükoğlu]

I think I have a similar problem...When I dial out using TDM400P,It sometimes 
stucks on and backs me as the line is busy although it is not...When I reboot 
the Asterisk box , it becomes ok..

-Original Message-
From: Colin E. McDonald [mailto:[EMAIL PROTECTED] 
Sent: Thursday, June 23, 2005 3:58 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Zap lines-inbound,outbound calls intersect

I have been having a problem for a while where an internal user will be
calling out via SIP through the Asterisk box which has a TDM400 and when
they pick up they have an inbound caller on the line. The lines then
become bridged and they stay that way until you do a soft hangup on
one of the lines. Is anybody else running into this problem?


Thanks

Colin
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HASSAN GROUP

HIGHTEX 2005
Ist. Uluslararasi Teknik Tekstiller ve Nonwowen Fuari
  bünyesinde sizleri agirlamaktan gurur duyar.
13-16 Temmuz 2005
  2.hol 28 no.  



Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun 
kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli 
olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi 
mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya 
kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj 
tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi 
vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve 
imha etmenizi rica ederiz. Tesekkürler - Hassangroup 

Important note : This e-mail transmission is intended only for the use of the 
individual or entity to which it is addressed, and may contain information that 
is privileged, confidential and that may not be made public by law or 
agreement. If the recipient of this message is not the intended recipient or 
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[Asterisk-Users] Help with Dial multiple channels simultanously

[Kib Eki]

Hi,

the following from extension.conf does not work correctly:

exten = 301, 1, Dial(SIP/455SIP/456, 15)

That is the console output:
   -- Executing Dial(mISDN/1/105, SIP/455SIP/456SIP/456| 10) in 
new stack

   -- Called 455
   -- Called 456
   -- SIP/455-46a8 is ringing
 == Spawn extension (incoming, 301, 1) exited non-zero on 'mISDN/1/105'

As you can see only the extension 455 is dialed.

What is wrong with my configuration?

Thank you very much,
Kib


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Re: [Asterisk-Users] FXS interfaces

[Alessandro]

On Wed, 2005-06-22 at 18:49, Mike M wrote:



Congratulations.  What were you using prior your pull from CVS?  Maybe
something old that didn't recognize the the TDM400P and its daughters?

 My prior went changing zaptel.conf like you said to:

fxoks=1,2 
fxsks=3,4 

 I got from CVS - azaptel libpri asterisk asterisk-addons asterisk-sounds in http://asterisk.org/index.php?menu=download

 

 But all ports are green! 

Really?  Maybe they aren't making the RED FXO cards anymore.  You should
look at them carefully for p/n differences and not rely on colors.  The
zapel driver tells you what you need to know too.

 Suppot Digium agreed all the ports are green as they are properly configured! This is the expected result.

 
 p1 - green 
 p2 - green
 p3 - green
 p4 - green

I wonder if Digium will update their website?  It's got a strong
commitment to red FXO modules in the graphics.


 Yes! Just do like http://asterisk.org/index.php?menu=download.

Thanks a lot!



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[Asterisk-Users] Legal Requirement for Digital PBX

[Roland Welker]

Hello,

Does anyone now, if there are any legal requirements for setups of
Digital (i.e. Linux/Asterisk) based PBXs in the UK? I am especially
interested, if a system does need to hang on a UPS?

Thanks,
Roland

Roland Welker
Moray Office Supplies
Edgar Road, Elgin, IV30 6YQ
T: +44/(0)1343/549869 F:+44/(0)1343/549300
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[Asterisk-Users] Asterisk @ Home setup Doc

[Paul]

As a newbie to Asterisk, I'm in love. 

There is no information discussing the way to use the FTP program vsftpd
which is need for phone configurations.  So far I've been able to add a user
with useradd and add that user to the VSFTPD.USER_LIST and now I can FTP to
my [EMAIL PROTECTED] server but need the layout to configure both Polycom phones
and Cisco phones if anyone can help?

Is there any way an outlook client from an XP station can use their contact
list to dial the phone.  I read in other posts that there is a way to force
a phone off hook and dial or dial and direct it to the phone, but can this
be integrated with MS outlook?

Can anyone direct me to samples of how to reconfigure the panel for operator
use as well as just regular desktop use (removing trunks, queues..)?

I have heard that Polycom phones have a problem with Asterisk - any truth to
this?

Thanks



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Re: [Asterisk-Users] Help with Dial multiple channels simultanously

[Asterisk]

Something is not quite right - your extensions.conf is specifying

Dial(SIP/455SIP/456, 15)

but the console is showing

Executing Dial(mISDN/1/105, SIP/455SIP/456SIP/456| 10)

note the extra SIP/456 (as in SIP/456SIP/456) and the 10 instead of the 
 15 in the extensions.conf.


Are you sure you've posted the correct extensions.conf ?

Julian


Kib Eki wrote:

Hi,

the following from extension.conf does not work correctly:

exten = 301, 1, Dial(SIP/455SIP/456, 15)

That is the console output:
   -- Executing Dial(mISDN/1/105, SIP/455SIP/456SIP/456| 10) in 
new stack

   -- Called 455
   -- Called 456
   -- SIP/455-46a8 is ringing
 == Spawn extension (incoming, 301, 1) exited non-zero on 'mISDN/1/105'

As you can see only the extension 455 is dialed.

What is wrong with my configuration?

Thank you very much,
Kib


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Re: [Asterisk-Users] FXS interfaces

[Alessandro]

Hi J.

On Wed, 2005-06-22 at 19:22, Jerry wrote:

Hi Alessandro,




I think he means the daughter card color, not the LED on the card slot.
What color are the actual daughter cards?

	You are correct! The actual daughter cards are green(FXS) and red(FXO).

Greetings!


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Re: [Asterisk-Users] Legal Requirement for Digital PBX

[Steve Underwood]

Roland Welker wrote:


Hello,

Does anyone now, if there are any legal requirements for setups of
Digital (i.e. Linux/Asterisk) based PBXs in the UK? I am especially
interested, if a system does need to hang on a UPS?
 


Unless things have changed, you need either:

   - 7 hours of UPS support or
   - an analogue line with any cheap old phone plugged in.

The analogue line could be used by the PBX when things are functioning 
normally, if a relay switches it to the phone on power failure.


Regards,
Steve


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RE: [Asterisk-Users] indexing tables for dialing

[Jay Milk]

Two approaches come to mind -- 1) Using DBPut/DBGet to associate a fixed
amount of phone-numbers with a given extension and dial, all from
extensions.conf, or 2) Using a small mySQL table and a short agi script
to accomplish the same thing.  The former solution has the advantage
that it's rather easy to implement and won't require any additional
components; the latter is more flexible and could allow maintenance of
the forward numbers by, say, a website.

 -Original Message-
 From: Ipek Zivane [mailto:[EMAIL PROTECTED] 
 Sent: Wednesday, June 22, 2005 6:50 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] indexing tables for dialing
 
 
 Hello
 
 I would like to know how can I manage to implement a table 
 which translates 
 an extension number into a phone number. Let see an example:
 
 If I dial an extension like 3021, Asterisk  has  to Dial an 
 agent (our 
 employees) located at San Francisco  using the following 
 telephone number: 
 415 541 . If it does not work we can also use his/her 
 mobile number.
 
 We need to manage more than 180 agents nationwide so I would 
 like to use a 
 table or data base to translate a large number of agent's telephones.
 
 The table looks like this:
 
 EXTPHONE1PHONE2   PHONE3
 
 3021  4155   415Y   510X
 2130  415Z510L
 3060  510X   XXX
 
 .
 .
    XXX XXX
 
 Thanks in advance for your help.
 
 Ypek
 
 _
 Sadece sohbet ile yetinmeyin - eglneceye de doymak için 
 Messenger'i tercih 
 edin! http://messenger.msn.com/?mkt=trDI=3490XAPID=2584
 
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Re: [Asterisk-Users] Re: Segfault on restart

[Alphonse Ogulla]

On 6/14/05, Tzafrir Cohen [EMAIL PROTECTED] wrote:
 Hi
 
 Has this been resolved?
 

Not as such but I noticed I get the error only when I run asterisk in
the foreground with the arguments -vvvc.

However I get no segfault error when asterisk restarts when running in
the background.

So in short, I do asterisk  then asterisk -vvvr to connect to the
running server. This way there is no segfault.

--
Rgds,
Alphonse Ogulla
Nairobi, Kenya
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[Asterisk-Users] Re: combining calls from 2 queues

[alan]

[EMAIL PROTECTED] wrote:

 We have 1 queue called helpdesk and are setting up a second one called isp.
 The helpdesk queue is for internal support calls and isp for our ISP customer
 calls.  Both of these queues will be directed to the same agents (helpdesk
 phone extensions).

 We want to have the separate queues for tracking purposes but the queued calls
 need to be ordered and answered as if there was only one queue.  For example,
 if there are 3 calls in the helpdesk queue and 1 call in the isp queue, if a
 new call comes in, no matter which queue, it should be 5th in the queue.

I am not sure if exactly what you want to do is possible. However, you
may be able to get both the tracking you need and the queue control
you need. If you use a single queue, but use the CDR Account Code to
store the helpdesk or isp designation, then you can get most of
what you want.

I don't think account code shows up in the queue log, so if you're using
queue logs for tracking, then this might not work for you.

Alan

 e.g.,
 helpdesk 3 calls
1   Zap/23-1
2   Zap/12-1
4   Zap/20-1

 isp  2 calls
3   Zap/22-1
5   Zap/18-1

 Is this type of setup possible and if so, what needs to be done in the config
 files to accomplish this?

 Thanks
 Ron Bergin
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[Asterisk-Users] AGI to monitor conenction quality

[Chris Mason (Lists)]

I need an AGI to monitor the quality of two connections and return a 
yes/no based on packet loss, connectivy, provider being there, so I can 
rollover the dial plan and dial the next available method. We have two 
internet connections, two providers, and PSTN for backup.


My main concern is to make sure the call gets connected one way or the 
other, cost being secondary but also important. I am not looking to 
determine the quality of the VOIP provider, just the network.

Has anyone got some code?

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


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Re: [Asterisk-Users] Legal Requirement for Digital PBX

[Roland Welker]

On Thu, 2005-06-23 at 21:43 +0800, Steve Underwood wrote:
 Roland Welker wrote:
 
 Hello,
 
 Does anyone now, if there are any legal requirements for setups of
 Digital (i.e. Linux/Asterisk) based PBXs in the UK? I am especially
 interested, if a system does need to hang on a UPS?
   
 
 Unless things have changed, you need either:
 
 - 7 hours of UPS support or
 - an analogue line with any cheap old phone plugged in.
 
 The analogue line could be used by the PBX when things are functioning 
 normally, if a relay switches it to the phone on power failure.

Thank you very much for this information. This is exactly what I needed.
And I presume, this applies as well to traditional systems using
proprietary protocols?

Regards,
Roland

 Regards,
 Steve
 
 
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-- 
Roland Welker
Moray Office Supplies
Edgar Road, Elgin, IV30 6YQ
T: +44/(0)1343/549869 F:+44/(0)1343/549300
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Re: [Asterisk-Users] SIP DID routing

[denis]

Hi Chris.

You have been facing the same problem of mine. I was encouraged to use the
CVS HEAD version that includes an application called SIP_HEADER. With
SIP_HEADER we can handle SIP Headers fields.

If you get success on it, please let me know. I will do the same.

Regards,

Deniss Galvãao.

 How do you get the called number on incoming SIP calls?  I've never
 had multiple DID's via SIP from one provider before and somehow I
 never realized that with IAX it just works, and SIP is different.

 If I don't set an extension in the register command the incoming
 invite has sip:[EMAIL PROTECTED] in the To field.  Now if I have multiple
 DID's that I want routed to different extensions, what's the solution?
  Is there a SIP header that is normally used to pass the called number
 in?

 Hope that makes sense..

 Chris
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[Asterisk-Users] Polycom display variable

[Kib Eki]

Hi,

does anyone know what Asterisk variable must be set to manipulate the 
line under From:-line with a polycom 500 ip phone?


Thanks + regards,
Kib

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[Asterisk-Users] BRI signalling Morocco

[igil]

Hello,

I have got a new project about an asterisk instalation in a Morocco travel Company.
I thought to use a quadBRI card to connect it to the Morocco PSTN, but I do not know exactly witch kind of port signalling should I use in Morocco.

Another thing is that I do not know exactly how the ISDN works in Morocco, witch mode use TE or NT, and if it works in Point to Point or Point to Multipoint.

Any clue will be welcomed.

Thanks for your time.

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Re: [Asterisk-Users] Help with Dial multiple channels simultanously

[Kib Eki]

yes, you are right - the extension.conf wasn't the same as debug output

but it is solved anyway. There was just a missing registration for the 
extension 456


Thanks

Asterisk wrote:


Something is not quite right - your extensions.conf is specifying

Dial(SIP/455SIP/456, 15)

but the console is showing

Executing Dial(mISDN/1/105, SIP/455SIP/456SIP/456| 10)

note the extra SIP/456 (as in SIP/456SIP/456) and the 10 instead of 
the  15 in the extensions.conf.


Are you sure you've posted the correct extensions.conf ?

Julian


Kib Eki wrote:


Hi,

the following from extension.conf does not work correctly:

exten = 301, 1, Dial(SIP/455SIP/456, 15)

That is the console output:
   -- Executing Dial(mISDN/1/105, SIP/455SIP/456SIP/456| 10) in 
new stack

   -- Called 455
   -- Called 456
   -- SIP/455-46a8 is ringing
 == Spawn extension (incoming, 301, 1) exited non-zero on 'mISDN/1/105'

As you can see only the extension 455 is dialed.

What is wrong with my configuration?

Thank you very much,
Kib


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[Asterisk-Users] This cpu usage doesn't seem right.

[Matthew Boehm]

Perhaps my deffinition of multi-threaded is skewed/wrong...

I've got asterisk HEAD running on a 4 proc machine.

I'm using top as my guide (and yes I know top sucks, but what else do I 
use?).


I just watched asterisk hit 63% cpu usage for about 5 seconds. There 
were 5/5 G729 licenses in use and 6 calls up during those 5 seconds.


CPU #2 had an idle of 39% and CPU #3 had 98%. CPU's #1 and #4 were both 
100% idle.


This kinda tells me that our max number of calls we can handle is about 
10-15? That doesn't seem right. That's not even a full T1.


I'm running an SMP kernel and asterisk is suposedly multi-threaded so 
why isn't the load being shared better across the procs?


Is there some compiler flag I missed to make asterisk better for 
multi-procs?


We want to be able to put in a quad span card but it seems asterisk 
can't even handle a single span at this rate.


Ideas? Comments?

Thanks,
Matthew

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[Asterisk-Users] dialtone conf.of Turkey for ata186 sip

[Betül Gözlükoğlu]

Hi;

Does anybody knows what is the recommended
settings for Turkey
to configure dialtone of ata186 sip version 3.1.1



Thanks in advance

Betul







HASSAN GROUP
HIGHTEX 2005
Ist. Uluslararasi Teknik Tekstiller ve Nonwowen Fuari bünyesinde sizleri agirlamaktan gurur duyar.
13-16 Temmuz 2005 2.hol 28 no.

Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup 
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RES: [Asterisk-Users] MFC R2 - Can this problem be solved??????????

[j_amorim]

Hello Steve, 

Wich will be the version I will need to install to solve this problem?? 

Is this version already finished 

Best Regards, 

Jônatas Amorim 
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Re: [Asterisk-Users] combining calls from 2 queues

[Johann]

My company is facing a similar situation.  The agents/queue system in 
Asterisk 1.0.x is badly designed to meet such needs.  Temporary I am 
working around the problem by giving each employee that answers a call 
one AgentID.  I then set them up as callback agents.  They are then 
members of both queues.  This will at least prevent the brain dead ACD 
from sending them multiple calls for each queue.  The distributing 
strategies however only look at their status with the current queue.  
There are issues with the wrapuptime as well and possibly some race 
conditions.  The users have reported some odd behavior that I haven't 
been able to exactly duplicate(ie some agents waiting minutes to get a 
call that is on hold while they are available).


For example, using leastrecent for both queues.  An agent that answers 
multiple back to back calls for the helpdesk queue does not get credit 
for it, when determining who least recently answered the incoming isp 
queue call.  At least if they are on a call from the helpdesk queue, 
they will not get a isp queue call.


The agentcallbacklogin() unforuantely hangs up once they login so you 
are stuck with the keeping them all as static agents on both queues or 
attempting to build a menu that will allow them to add/remove themselves 
dynamically from each queue.  They would still require at least 3 
logins(agent login, agent logout, and queue management menu).


Message me off list for further details.  We may be able to work 
together to come up with a workaround that would serve both are needs 
better.


--johann

[EMAIL PROTECTED] wrote:

We have 1 queue called helpdesk and are setting up a second one called isp. 
The helpdesk queue is for internal support calls and isp for our ISP customer

calls.  Both of these queues will be directed to the same agents (helpdesk
phone extensions).

We want to have the separate queues for tracking purposes but the queued calls
need to be ordered and answered as if there was only one queue.  For example,
if there are 3 calls in the helpdesk queue and 1 call in the isp queue, if a
new call comes in, no matter which queue, it should be 5th in the queue.

e.g.,
helpdesk 3 calls
  1   Zap/23-1
  2   Zap/12-1
  4   Zap/20-1

isp  2 calls
  3   Zap/22-1
  5   Zap/18-1

Is this type of setup possible and if so, what needs to be done in the config
files to accomplish this?

Thanks
Ron Bergin

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Re: [Asterisk-Users] so many FXS ports :)

[Seamus Abshere]

That's what I'm confused about:
* two 4 port FXS cards
* one 24 port FXS channel bank
both, neither, and if both -- why do you need the dual digium cards? 
shouldn't your channel bank just take MGCP or SIP or something?


What am I missing?

[EMAIL PROTECTED] said:

Shawn guessed correctly; Most likely a channel bank with 24FXS.  We have 2
cards each with 4 ports.

  1   Zap/23-1
  2   Zap/12-1
  4   Zap/20-1




Seamus said:

this is perhaps a silly question, but how do you have so many zaptel
FXS's? do you have six TDM400 cards with four FXS's each? or what am I
missing?


--
Seamus Abshere
Isthmus Group
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[Asterisk-Users] Always forward an extension?

[C. Hatton Humphrey]

Here's something I haven't been able to discover as of yet - I need to
set up a direct link from my Asterisk box to an external line...
basically I need to be able to pick up an internal extension and have
it call a local phone number.

This is call forwarding, I know - the question that I have is how do I
set it up so that the extension always forwards.  There will never be
a client logging in to it.

Thoughts, experiences, ideas?

Thanks!
Hatton
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Re: [Asterisk-Users] missing cdr records

[Paul Traue, Jr.]

Rosario,

Unfortunately this problem doesn't just affect you, I'm also affected 
and have been since 1.0.5.  If you set the debugging high enough and use 
mysql, you'll see the insert statements being generated by asterisk, but 
they never make it to the DB.


I'm glad to know I'm not the only one affected.  Any others experiencing 
this problem or have a fix?


Paul

Rosario Pingaro wrote:

I am experiencing a very wired problem.
 
Some of my cdr are lost.
 
I use logging cdr to csv, mysql and odbc. But some of them are lost. 
They miss in csv mysql and odbc, so i'm pretty sure it is related to 
asterisk functioning.
 
I am running asterisk 1.0.7; this is simple configuration file:
 
extensions.conf

[general]
static=yes
writeprotect=no
 
[macro-gw-voipjet]

exten = s,1,SetCallerID(${CALLERIDNAME})
exten = s,2,Dial,IAX2/[EMAIL PROTECTED]/${ARG1}
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Busy
exten = s-CHANUNAVAIL,1,Noop
exten = _s-.,1,Congestion
[macro-gw-nufone]
exten = s,1,SetCallerID(${CALLERIDNAME})
exten = s,2,Dial,IAX2/[EMAIL PROTECTED]/${ARG1}
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Busy
exten = s-CHANUNAVAIL,1,Noop
exten = _s-.,1,Congestion
[ser]
; combinazione 81 - per provider americani - destinaione usa e canada
exten = _81.,1,Macro(gw-nufone,${EXTEN:1})  ; NuFone
exten = _81.,2,Macro(gw-voipjet,${EXTEN:1}) ; VoipJet.com
exten = _81.,3,Congestion
 
; combinazione 8011 - per provider americani - destinaione 
rotteinternazionali

exten = _8011.,1,Macro(gw-voipjet,${EXTEN:1}) ; VoipJet.com
exten = _8011.,2,Macro(gw-nufone,${EXTEN:1})  ; NuFone
exten = _8011.,3,Congestion
 
 
the percentage of cdr lost is around 5% and they are pretty concentrate 
in the meaning that if I loose 5 cdrs they are lost 3 in around 2 
minutes interval and 2 in anothe short interval.
 
Any advice on how to debug ?
 
thanks

Rosario
 





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Re: [Asterisk-Users] Polycom display variable

[Kib Eki]

This works for me:

to display the following on the polycom phone:
From: Support-Group
x 
--- the caller id number


you can use the following code in extension.conf:
exten = 301, 1, Dial(SIP/456SIP/455SIP/457, 30)
exten = 301, 2, SetVar(foo=* Support-Group * ${CALLERIDNUM})
exten = 301, 3, SetCallerID(${foo})
exten = 301, 4, Dial(SIP/705)
exten = 301, 5, Hangup




Kib Eki wrote:


Hi,

does anyone know what Asterisk variable must be set to manipulate the 
line under From:-line with a polycom 500 ip phone?


Thanks + regards,
Kib

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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Armin Schindler]

On Thu, 23 Jun 2005, Klaus-Peter Junghanns wrote:
 Am Donnerstag, den 23.06.2005, 12:41 +0200 schrieb Armin Schindler:
 
   I strongly disagree. :-) You dont want to do echo cancelation in
   userspace. Especially not on a non-realtime operating system.
   To make echo cancelation work it has to be as close to the line
   interface as possible. Also the frames have to be as small
   as possible. This rules out capi pretty much.
  
  If you don't want echo-canceling in user-space, then neither Asterisk nor
  any chan_* plugin should do it.
  
  I don't know the zap channel code, but does the zap echo-cancel-code is 
  inside a kernel module?
 
 Yes, sir.
 
  If yes, then I have to disagree here. Something like 'playing' with 
  audio-data is nothing the kernel should be concerned with.
  This belongs in user-space and if you need realtime, then you should use a 
  realtime OS or use RT-scheduling. Just putting such a code into kernelspace 
  is a bad idea.
 
 What is so bad about playing with audio-data in kernel space?

Besides preemption or RT-patches, it is not easy (and noboady does it)
to be 'nice' and have a fair scheduling. With such work in kernel, you just
say I'm at the highest priority, I don't care about others. And that's 
just wrong in the kernel.
Normaly, the kernel just should provide access to the hardware 
and basic functions for interaction with the hardware.

 If you play with echo cancelation in user space you will need
 to de-jitter the audio first introducing more and more latency, so
 your echo cancelation becomes way more computationally expensive.

That depends on what scheduling priority this task runs. If you don't want 
to get interrupted by other tasks, then you need a higher priority. 
 
  So the correct way is either the hardware supports it or the 
  application knows what to do with the data received, like DTMF.
  
 
 Why should the application have to worry about things like echo
 cancelation?

In the case of Asterisk and echo-cancel, this application is the
position where is makes sense. Otherwise you need a copy of the echo-cancel 
code in each channel driver.

 Zaptel is not only used by Asterisk but also by other
 projects. With EC in kernel space (which gets switched on and off
 by userspace) there is no need to reinvent the EC-wheel for every
 project.

Okay, from that point of view it makes sense. On the other hand, something 
like echo-cancel and DTMF is not channel specific and therefore should not 
be part of that. Or would you add additional codecs into the channel driver?

Armin
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[Asterisk-Users] ChanSpy on Asterisk v1.0.7

[Tim Karl]

I am trying to find the app ChanSpy for Asterisk v1.0.7. I have tried 
looking on VOIP-info.org's ChanSpy page 
(http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ChanSpy)and 
also referred to the link regarding bug 3836 
(http://bugs.digium.com/bug_view_page.php?bug_id=0003836). I downloaded 
the attachments and tried to use the patch and compile the source. 
However, it seems that these files are for a different version of 
Asterisk. Searching Google provides no relevant material.


If anyone has any information as to where I can find ChanSpy for 
Asterisk v1.0.7 please reply. Thank you for your help.


--Timothy Karl
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Re: [Asterisk-Users] New Asterisk Implementation

[Don Brearley]

Leave what you have in place.  Install Asterisk and investigate the
various line interface options.  Enlist early adopters on your campus to
participate in the trial.  Connect Asterisk to your existing system with
PRI.  Gradually ramp up the Asterisk system and ramp down the existing
system.



Mike,

That's exactly what I plan on doing.  Initially I will roll it out in my office
only, then to a department that's already expressed interest in being
my guinea pig.  

Thanks for the advice.
-Don

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[Asterisk-Users] Monitoring Sirrix quad BRI channels

[David Wilson]

Hi all,

How are things going ?

Is there a way for me to individually identify 
each BRI channel on the Sirrix quad BRI board.

The reason I ask is because our client uses the 
"Asterisk Flash Operator Panel" to monitor its external lines and transfer calls 
from the lines to the various SIP phones.

The "Flash Operator Panel" requires that we set a 
static value for each line or channel. With analogue cards its easy as the lines 
are Zap/1, Zap/2, Zap/3 etc. With the Sirrix board the value seems to change: 
0814f1f8, 08129f38, 0837ad40.

Is there anyway I can get this right so that each 
channel (8 of them) can be monitored ?

Thanks in advance.
Kindest regardsDavid Wilson___D 
c D a t aTel +27 33 342 7003Fax +27 33 345 4155Cell +27 82 
4147413http://www.dcdata.co.za[EMAIL PROTECTED]Powered by Linux, 
driven by passion ! ___

"Computers are not intelligent. They only think they 
are."
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Re: [Asterisk-Users] New Asterisk Implementation

[Don Brearley]

 I just want to be sure that it's possible to do this, and that im not wasting 
 my time.

No time wasting. This is a great fit for you. Stay away from some of
the Dell servers until you know that they work well. (interupt and
ACPI issues)

--

Thanks for the advice on staying away from Dell servers..  

I have three Sun  Ultra Enterprise 450's (Quad CPU, gigs of RAM) that I want to
use for this project.

I would like one to be a primary asterisk box, and have the other two
as backups.

Have you, or anyone else heard of similar problems with this kind of
gear?   

Thanks again!
- Don

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RE: [Asterisk-Users] CallerID name lookup AGI script

[Oswaldo Arratia]

Hi List, 
I've managed to install this great sript and it's working fine.

I am using this in the US, just want to know if this is possible and if so,
how:

1- Remove the '!' before the name when the calling number is a Cell phone
2- Remove the '1' before the number. I'd like the number to appear as
xxx-xxx-  instead of 1-xxx-xxx-.

Thanks very much for your answers.

Oswaldo A.

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Friday, May 20, 2005 9:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] CallerID name lookup AGI script

Good find -- this could WELL be the case.  I'll spend a little time this
weekend gettting this to work on php4 as well.  There's no good reason to
restrict to php5, other than that the code actually looks better.

 -Original Message-
 From: Chris Coulthurst [mailto:[EMAIL PROTECTED]
 Sent: Friday, May 20, 2005 2:09 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] CallerID name lookup AGI script
 
 
   I don't know if you ran in to the same thing I did, but did you 
 upgrade your PHP to 5.0.4 to make this work?  I found that the 
 cid_rewrite.php script calls the php from /usr/bin, whereas the PHP 
 5.0.4 install defaults to /usr/local/bin if you don't specify --prefix 
 on the build. So you might still be running the old php4.  I just 
 renamed /usr/bin/php to /usr/bin/php.old, symlinked the 
 /usr/local/bin/php to /usr/bin, and did the same for pear, and it 
 worked.
 
 
 Chris Coulthurst
 [EMAIL PROTECTED]
  
 
 |-Original Message-
 |From: [EMAIL PROTECTED]
 [mailto:asterisk-users-
 |[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia
 |Sent: Friday, May 20, 2005 11:34 AM
 |To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 |Subject: RE: [Asterisk-Users] CallerID name lookup AGI script
 |
 |Hi there,
 |
 |I am trying to the the cid_rewrite.php script but if I run
 it from the
 |directly I get this error:
 |
 |./cid_rewrite.php
 |
 |br /
 |bParse error/b:  parse error, expecting `T_OLD_FUNCTION' or 
 |`T_FUNCTION' or `T_VAR' or `'}'' in b/var/lib/asterisk 
 |/agi-bin/astlib_jm.php/b on line b73/bbr / br / bFatal 
 |error/b:  Cannot instantiate non-existent class:  agi in 
 |b/var/lib/asterisk/agi-bin/cid_rewrite.php/b on line b60/bbr 
 |/
 |
 |
 |
 |Here is the PHP version I am using:
 |PHP 5.0.4 (cgi) (built: May 20 2005 14:08:40) Copyright (c) 1997-2004 
 |The PHP Group Zend Engine v2.0.4-dev, Copyright (c) 1998-2004 Zend 
 |Technologies
 |
 |Does anybody know what my problem is or if I am missing
 anythiong here?
 |
 |Thanks!!
 |
 |Oswaldo
 |
 |
 |
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Re: [Asterisk-Users] New Asterisk Implementation

[Don Brearley]

 [EMAIL PROTECTED] 06/22/05 12:55PM 
  I do understand that I would need to replace all of
  my existing telephones with VoIP-capable
  phones, and that I'll need to re-wire most of the
  campus telephone infrastructure (it's still
  all cat-3) -- these arent problems.   
why do you think that you need to do that ?
you could just install 2 *boxs with 2 4 port t1 cards in each
(sangoma or digium) and (4 * 1500 - $6K)
300+ analog ports  (7 * 1K ~ 7K)
13 x 24 port channel banks (adtran ta-750)
or 7  48 port channel banks(adit 600) 
feels less expensive than 300 voip hand sets
say 300 * 80 = 24K vs 13K

That's sweet!  Saving that much cash will only help me 
get the administration to approve this project.

Thanks again!
-Don

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Re: [Asterisk-Users] Asterisk Manager Interface Remote Buffer Overflow Vulnerability

[Brian West]

THANK YOU NANCY DREW!!!  Could be a bit more vague about this eh?

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 22, 2005, at 6:30 PM, trixter http://www.0xdecafbad.com wrote:


http://www.frsirt.com/english/advisories/2005/0851

A vulnerability was identified in Asterisk, which may be exploited by
authenticated attackers to execute arbitrary commands. This flaw is  
due

to a buffer overflow error in the manager interface that does not
properly handle specially crafted commands, which could be  
exploited by
an authenticated attacker to obtain root privileges. Note : the  
manager

interface is not enabled by default.


--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
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Re: [Asterisk-Users] ChanSpy on Asterisk v1.0.7

[Brian West]

Just use CVS-HEAD.. stable is a pile of crap. let the flames being

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 23, 2005, at 10:09 AM, Tim Karl wrote:

I am trying to find the app ChanSpy for Asterisk v1.0.7. I have  
tried looking on VOIP-info.org's ChanSpy page (http://www.voip- 
info.org/tiki-index.php?page=Asterisk+cmd+ChanSpy)and also referred  
to the link regarding bug 3836 (http://bugs.digium.com/ 
bug_view_page.php?bug_id=0003836). I downloaded the attachments and  
tried to use the patch and compile the source. However, it seems  
that these files are for a different version of Asterisk. Searching  
Google provides no relevant material.


If anyone has any information as to where I can find ChanSpy for  
Asterisk v1.0.7 please reply. Thank you for your help.


--Timothy Karl
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Re: [Asterisk-Users] MFC R2 - Can this problem be solved??????????

[Matias G.]

Take a look at the unicall.conf file, in the line with the 
protocolvariant=br,XX,YY be sure you're using the right amount of digits... 
XX should be the length of the ANI you're receiving and YY should be the 
length of the DNIS...


if it doesn't work please try a debug of the unicall (in unicall.conf - 
loglevel=1023)


hope this helps, please let me know if this solved the issue

bye,
M.
- Original Message - 
From: j_amorim [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Sent: Tuesday, June 21, 2005 5:02 PM
Subject: [Asterisk-Users] MFC R2 - Can this problem be solved??



Ok Steve,

Wich will be the version I will need to install to solve this 
problem??



Best Regards,

OBS: I am really in Brazil and I am using a R2 E1 from Embratel( Telco
company here in Brazil).








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[Asterisk-Users] Ant: Re: [Asterisk-biz] sipredirect question

[Axel Schemberg]

Hi Emanuele,

you are right. I installed the CVS HEAD now (tried out asterisk -V) but sipredirect is unknown.

Do you have any hint for me, where I can have a look in which version it will be included?

Kind regards,

AxelEmanuele Pucciarelli [EMAIL PROTECTED] schrieb:
Axel Schemberg wrote: I use Asterisk on Debian via: ap-get install asterisk, which is Version 1.07.The page you linked says: "new in Asterisk 1.2.x". I guess that thispretty much explains why it does not work in your case :)BTW, this looks like a -users question to me, so I've moved it there.-- Emanuele
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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Klaus-Peter Junghanns]

  
   If yes, then I have to disagree here. Something like 'playing' with 
   audio-data is nothing the kernel should be concerned with.
   This belongs in user-space and if you need realtime, then you should use 
   a 
   realtime OS or use RT-scheduling. Just putting such a code into 
   kernelspace 
   is a bad idea.
  
  What is so bad about playing with audio-data in kernel space?
 
 Besides preemption or RT-patches, it is not easy (and noboady does it)
 to be 'nice' and have a fair scheduling. With such work in kernel, you just
 say I'm at the highest priority, I don't care about others. And that's 
 just wrong in the kernel.

That is actually what you want to do if your system is a PBX. You want
to give as much as priority to your audio quality as you can. Even if
this means that userspace applications get unfair scheduling results.
If you take care of the critical audio handling (like EC) inside the
kernel then your (maybe very unexperienced) users cannot easily
disturb this process by causing a high load in user space, e.g. by
running webservers, fileservers, mailservers or X on their PBX!
It's far better to have good audio quality (with a working EC) and
a slowed down webserver than a garbled audio and fast webserver.

Just my 2 eurocents.

 Normaly, the kernel just should provide access to the hardware 
 and basic functions for interaction with the hardware.
 
  If you play with echo cancelation in user space you will need
  to de-jitter the audio first introducing more and more latency, so
  your echo cancelation becomes way more computationally expensive.
 
 That depends on what scheduling priority this task runs. If you don't want 
 to get interrupted by other tasks, then you need a higher priority. 

This is true in a perfect world. :) However there are lots of nasty
things in your linux box (like harddisk controllers hogging your cpu, 
etc...) that make your system a non-realtime system.

  
   So the correct way is either the hardware supports it or the 
   application knows what to do with the data received, like DTMF.
   
  
  Why should the application have to worry about things like echo
  cancelation?
 
 In the case of Asterisk and echo-cancel, this application is the
 position where is makes sense. Otherwise you need a copy of the echo-cancel 
 code in each channel driver.
 
  Zaptel is not only used by Asterisk but also by other
  projects. With EC in kernel space (which gets switched on and off
  by userspace) there is no need to reinvent the EC-wheel for every
  project.
 
 Okay, from that point of view it makes sense. On the other hand, something 
 like echo-cancel and DTMF is not channel specific and therefore should not 
 be part of that. Or would you add additional codecs into the channel driver?
 

I would even put more things into kernel space just to reduce latency.
There are people that would even enjoy RTP in kernel space.

Running things in userspace makes sense from a software architectural
point of view. But in real life this can be very dangerous if you dont
have control over the complete userspace (e.g. users on crack running
make bzImage -j).

 Armin

Klaus

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Re: [Asterisk-Users] privacy manager

[Brian West]

Why wait? And why use agi? and why in the hell use parking?

Call comes in without callerid:

; call gets answered
exten = s/,1,Answer
exten = s/,2,Set(SCREENFILE=/tmp/screen-${CALLERIDNUM})
; ask the callers name and records it
exten = s/,3,Playback(screen-record)
exten = s/,4,Record(${SCREENFILE}.gsm|60|20)
; dials you playing music to the caller (no need to park it)
exten = s/,5,Dial(SIP/me|120|mM(screen^${SCREENFILE}))
exten = s/,6,Voicemail(u200)

; You answer and this macro exec's caller still hears music
[macro-screen]
exten = s,1,Playback(silence/1)
exten = s,2,Playback(screen-from)
; plays recorded file
exten = s,3,Playback(${ARG1})
;asks you to accept it by pressing 1 anything else rejects the call
exten = s,4,Read(ACCEPT|screen-accept|1||3)
exten = s,5,GotoIf($[${ACCEPT} = 1 ] ?7:6)
; if you pressed 1 this will make it bridge the call else it sends it  
to voicemail

exten = s,6,SetVar(MACRO_RESULT=CONTINUE)
exten = s,7,System(/bin/rm -f ${ARG1}*)


Don't forget to visit http://www.pbxfreeware.org and http:// 
www.cluecon.com


/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 23, 2005, at 10:11 AM, John Hill wrote:






1- Call comes in without callerid
2- AGI script answers line
3- AGI script asks to record name
4- Park the call and get the parked extension number
5- Ring all the phones in the house (exec Dial)
6- If phone is picked up, play recorded name
7- Wait for DTMF to accept or decline call
8- If accepted, bridge parked call and current call.





Mike,
I am wanting this same application. Have you found a soulution?

--john

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Re: [Asterisk-Users] Connecting extern telephones,

[Carlos Chavez]

On Thu, 2005-06-23 at 00:46 +0200, satchid wrote:
 Dear List members,
 I have an asterisk box whereon 45 GXP-2000 telephones from Grandstream are
 connected at my work. This works fine. 
 Now I want to take 5 GXP-2000s to different homes on internet and want them
 to be part of the same internal telephone system. One external GXP-2000 is
 to be the night receptionist that should be able to transfer calls to any
 other of our extentions internally or externally. 
 Then also I have 20 WIFI Handsets (F1000 Utstarcom) that work well on the
 local Wifi stations, but they have to work on free AP al over the world as
 well as internally as part of the local telephone system. 
 
 What are my options to get these working for me? How can I get them
 communicating to each other.
 
 I hope that I gave the needed information, Please ask for more if needed.
 
I really think you should read the documentation for Asterisk.  The
only requirement to connect a phone from outside your network is that
the IP address of the * server is accessible from the Internet.  You can
either have a public IP for your server or use a DMZ if you are behind a
NAT router.  You only have to configure the phones with the correct IP
address and they operate exactly the same as the internal phones.

The only thing you should consider is using voice compression to save
bandwidth.  Maybe buy a few G.729 licenses or at least use G726-32.

-- 
--
Telecomunicaciones Abiertas de Mexico
Carlos Chavez
Director de Tecnologia
+52-55-91169161 Ext. 2001

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RE: [Asterisk-Users] ChanSpy on Asterisk v1.0.7

[Lee Archer]

What's the best way to get 1.0.8?  I've downloaded the latest from CVS but when 
I compile it it says 1.0.6!!  Is that right?

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: 23 June 2005 16:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ChanSpy on Asterisk v1.0.7

Just use CVS-HEAD.. stable is a pile of crap. let the flames being

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 23, 2005, at 10:09 AM, Tim Karl wrote:

 I am trying to find the app ChanSpy for Asterisk v1.0.7. I have tried 
 looking on VOIP-info.org's ChanSpy page (http://www.voip- 
 info.org/tiki-index.php?page=Asterisk+cmd+ChanSpy)and also referred to 
 the link regarding bug 3836 (http://bugs.digium.com/ 
 bug_view_page.php?bug_id=0003836). I downloaded the attachments and 
 tried to use the patch and compile the source. However, it seems that 
 these files are for a different version of Asterisk. Searching Google 
 provides no relevant material.

 If anyone has any information as to where I can find ChanSpy for 
 Asterisk v1.0.7 please reply. Thank you for your help.

 --Timothy Karl
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Re: [Asterisk-Users] Asterisk Manager Interface Remote Buffer Overflow Vulnerability

[Zoa]

Haha, fun.


Why use the bufferoverflow if you already have the permissions to
execute any linux command using the manager interface :p


Brian West wrote:


THANK YOU NANCY DREW!!!  Could be a bit more vague about this eh?

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 22, 2005, at 6:30 PM, trixter http://www.0xdecafbad.com wrote:


http://www.frsirt.com/english/advisories/2005/0851

A vulnerability was identified in Asterisk, which may be exploited by
authenticated attackers to execute arbitrary commands. This flaw is  due
to a buffer overflow error in the manager interface that does not
properly handle specially crafted commands, which could be  exploited by
an authenticated attacker to obtain root privileges. Note : the  manager
interface is not enabled by default.


--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
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Re: [Asterisk-Users] Always forward an extension?

[Brian West]

You're trying way too hard on this one.

exten = 555,1,Dial(Zap/g1/18005551212)

its no different than anything else in the PBX just set it up.. no  
need to forward it.


replace 555 with the internal extension you wished to

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 23, 2005, at 10:07 AM, C. Hatton Humphrey wrote:


Here's something I haven't been able to discover as of yet - I need to
set up a direct link from my Asterisk box to an external line...
basically I need to be able to pick up an internal extension and have
it call a local phone number.

This is call forwarding, I know - the question that I have is how do I
set it up so that the extension always forwards.  There will never be
a client logging in to it.

Thoughts, experiences, ideas?

Thanks!
Hatton
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Re: [Asterisk-Users] flash panel only works with IP address

[Carlos Chavez]

On Thu, 2005-06-23 at 08:54 +0200, [EMAIL PROTECTED] wrote:
 Hi,
 
  
 
 It seems that my flash panel only works when I specify my ip address
 and not the host name.
 
 I've tried quite a few things (change host file, dns resolve,
 proxying….) but couldn’t get it to work.
 
 Anyone knows how to solve this?
 

There is a specific list for FOP you should directo your questions to.

What did you put as the web address for your server in the
op_server.cfg file?  It should be something like:

web_hostname=server.name.com

If you put your IP address there that is why only your IP address
works.  Only the name or address listed there will work.

-- 
--
Telecomunicaciones Abiertas de Mexico
Carlos Chavez
Director de Tecnologia
+52-55-91169161 Ext. 2001

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RE: [Asterisk-Users] Cisco 7960 firmware upgrade promblems

[Tarpo, Louie]

That is not a typo.  One is the loader, the other is the firmware.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Geoff
Manning
Sent: Thursday, June 23, 2005 6:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Cisco 7960 firmware upgrade promblems


 I have a second-hand 7960 which I am attempting to upgrade to 
 use a SIP
 image.
 
 The phone currently has a firmware release which doesn't seem 
 to be listed
 in Cisco docs - P003AM30. On reboot, it finds the tftp server 

Here's how I performed the upgrade:

Downgrade from the stock P003AM30 to POS30203

Upgrade to version 5.1 (first signed binary firmware)

Upgrade to version 7.1 * (most recent version? maybe 7.4?)

* When upgrading to 7.1 there is a typo in the OS79XX file, it will say
P00x change it to P0Sxgreat typo by Cisco.


Check the comments on this wiki page: 

http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx?page=Asterisk%20phon
e%20cisco%2079xxcomments_threshold=0comments_offset=0comments_sort_mode=c
ommentDate_desccomments_maxComments=10comments_parentId=353#threadId358

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Re: [Asterisk-Users] Server Load/Capacity

[Waldo Rubinstein]

I was reading up on DUNDI and although it sounds like it would solve  
some of my problems, I don't know if it will do everything I want to  
do. For example, among the things I wanted to do is something like:  
the call director will have access to a database that will tell it  
which server a particular agent is logged in. Then, someone like a  
supervisor, could request the call director that he/she wishes to  
monitor (zapbarge or agentbarge) the agent's conversation and the  
call director will intelligently know where to direct the  
supervisor to monitor the call.


Assuming this setup works in general, I don't even know if I'm going  
to be able to do things like that. If you know, let me know please.


Thanks,
Waldo

On Jun 23, 2005, at 7:47 AM, tim panton wrote:



On 23 Jun 2005, at 10:48, Waldo Rubinstein wrote:


I'm trying to figure out how much call load I can put on a Dual  
Xeon 2.4 Ghz Asterisk server acting strictly as an IAX2 call  
director, as show in the diagram below.


The idea is that I have N number of gateway asterisk servers  
connected to the PSTN using T1 Digium boards. Then, I have M  
number of servers where my agents and/or telephone extensions  
(whether they are IAX or SIP hard/soft phones). What I'm trying to  
accomplish is put a server in between these two groups of machines  
which will simply be able to intelligently route calls in either  
direction. This call director server will only use IAX2 (ulaw)  
to minimize any transcoding and alleviate load.


Under this scenario, does anyone have any idea how many calls this  
call director server may be able to handle/direct?





At Astricon the man from Signate showed some benchmark results  
which indicated
a 'stock' PC server could do 122 ulaw SIP passthrough calls at  
acceptable

call quality.
Their own-brand servers can do  2k (If I remember right).

However I think you should look into Dundi - used correctly with a
clear dialplan you may be able to get rid of the director and
have a cloud of dundi peers instead.

Tim.
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Re: [Asterisk-Users] the table tariffrate is empty in Areskicc !

[Areski K]

Hi Zhu!

Please check if you assign ratecard(s) to the tariffgroup!
Go on list Tariffgroup, click edit on the one u re using,
then check if one of the ratecard, at least, has been added.

Schusss,
Areskaille la canaille

On 6/21/05, zhu [EMAIL PROTECTED] wrote:
 hello, guys:
 my areskicc can connect to asterisk. but problem is that my input number
 is invalid. later I checked the source code and found that the table
 tariffrate in asterisk is empty. actually I added all necessary data
 from areskicc. I do not why this table still be empty. who call tell me
 how to insert data from areskicc in right format.
 any help would be appreciated !
 zhu
 
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[Asterisk-Users] mini itx

[jltaylor]

I've seen the embedded posts.
Is anyone running Asterisk on the MINI ITX?

James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx  75503
903-793-1956

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Re: [Asterisk-Users] *77 does not work ..

[Jorge Carrasquillo]

I had the same issue with a Sipura 2002 and the Firefly
softphone. The Sipura had another code applied to *77. I
was able to change default setting for *77 via the web interface of the
sipura to *777 and after that I was able to get into the digital
receptionist. As for the Firefly softphone I never could
get it to dial the digital receptionist.

Hope that helps,

JorgeOn 6/22/05, Brian Watters [EMAIL PROTECTED] wrote:
I can not get *77 to work on our Asterisk server .. @ home 1.1 final ...Other * codes seem to work without issue .. Just can't use the *77 code ..Anyone have any ideas what to look for ??BRW___
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Re: [Asterisk-Users] SER and Asterisk question

[Iqbal]

Then you sip.conf is not defined properyly, or your extensions.conf does 
not have any end, the only way the call should really go back is if 
asterisk is telling it to, see what is called in what order in your 
dialplan, remeber the dialplan is not called in the order it is written 
in the file


Iqbal

Mohamed A. Gombolaty wrote:


Dear Yair,

Actually what happens is that from SER debug I can see the call is looping
between Asterisk and SER. but adding a number makes no loops.

Thx
MAG



Yair Hakak wrote:

 


yes, there is.
run everything through asterisk, no matter how long the extensions
are. for example, 666 calls 999
goes to asterisk, sees a dial sip:[EMAIL PROTECTED], goes back to SER.

bounces back to ser. If everything is working well asterisk will set
up the call and get out of the way.

I don't see why you need to prepend digits in order to make this work,
if i'm missing something let me know.

-yair
   





 


On 6/16/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote:
   


Dear All,

I am trying to make the phones always talk to each other (peer to peer)
using SER as a sip proxy, and incase the call is not answered we will
use the voicemail of asterisk and other feautures, I have done that
already, but in order to do so I found that I have to make the users
dial different exten numbers, here is an example:

user with exten 666 wants to call 999 .
666 dials 1999 and   which has a uri rule that says forward 4 digit
starting with 1  to the asterisk sip port
the asterisk extensions.conf has an entry for 1999  and dials
[EMAIL PROTECTED], if not answered voicemail runs and so on.

ain't there a way to make 666 directly call 999 without using 1999.


--
Thx
MAG



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--
Thx
MAG



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.

 


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Re: [Asterisk-Users] OT: MAX TNT and PRI calling name (CNAM) facility message

[Matt Fredrickson]

On Thu, Jun 23, 2005 at 12:20:34AM -0600, Kevin Blackham wrote:
 Does anyone have a MAX/APX with working ingress PRI calling name?
 
 I recently acquired a MAX TNT on the cheap and it's integrating fine
 except for one thing.  In the 11.0.0 release notes, it is stated that
 ISDN calling name will, if present and permitted by presentation
 flags, be added to the From: and Remote-Party-ID: headers of the
 INVITE.  I'm not able to make this happen.  Pcap captures show it is
 indeed in neither header, and I suspect the MAX is sending the INVITE
 before it receives this data.  Debug traces show it does receive the
 message, but due to limitations of the CLI, I cannot correlate whether
 it's received before or after the INVITE is dispatched.  It works
 great direct to Asterisk (of course) via TE410P on the same NI-2
 spans.
 
 My FACILITY message that contains the CNAM wanders in from 100 to
 400ms after the initial SETUP.  I can't seem to find any way to get
 the MAX to stall for a half-second before invoking the INVITE (if
 that's even the issue).  Is my provider too slow?  Is there another
 valid way for CNAM to be provided during the SETUP message, assuming
 my provider can stall the call setup until the SS7 query is returned?
 (google for Q.931 docs not helping me much there either)

That's one of the (many) ways that caller name is provided.  In fact, it's
pretty much the most common way that I've seen for ISDN PRI.  I don't
know if you're provider supports it, but sometimes you can get it in the
SETUP message.  I'm not sure what level of control they have though.

Matthew Fredrickson
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[Asterisk-Users] Re: Cisco 7960 firmware upgrade promblems

[Patrick Lidstone (Personal E-mail)]

  I have a second-hand 7960 which I am attempting to upgrade to 
  use a SIP
  image.
  
  The phone currently has a firmware release which doesn't seem 
  to be listed
  in Cisco docs - P003AM30. On reboot, it finds the tftp server 
 
 Here's how I performed the upgrade:
 
 Downgrade from the stock P003AM30 to POS30203
 
 Upgrade to version 5.1 (first signed binary firmware)
 
 Upgrade to version 7.1 * (most recent version? maybe 7.4?)
 
 * When upgrading to 7.1 there is a typo in the OS79XX file, 
 it will say
 P00x change it to P0Sxgreat typo by Cisco.
 
 
 Check the comments on this wiki page: 
 
 http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx?page=A
 sterisk%20phon
 e%20cisco%2079xxcomments_threshold=0comments_offset=0commen
 ts_sort_mode=c
 ommentDate_desccomments_maxComments=10comments_parentId=353#
 threadId358

Thanks, I've almost cracked it now! 2 out of 3 phones are OK, but the third
phone sticks on the Universal Application Loader. I've put a packet sniffer
on the network, and I can see it requesting a DHCP address which isn't on my
network (some legacy config presumably?), and when that times out, it
requests and is issued with an address that is valid for my network, but
then never attempts to connect to any TFTP server. I then set up a dummy
network (DHCP, TFTP server) which matches the network parameters of the
legacy config, and I still don't see any TFTP requests. Any suggestions on
what to do next? I'm out of ideas...

Patrick

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Re: [Asterisk-Users] mini itx

[Iain Young]

On Thu, Jun 23, 2005 at 11:39:21AM -0500, jltaylor wrote:

 I've seen the embedded posts.
 Is anyone running Asterisk on the MINI ITX?

Yes, no problems, I have an X100P in the PCI slot, but its only
a single POTS line. I used the MII board, but only because thats
what I had avaliable.


Iain
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[Asterisk-Users] ASTCC not making calls

[Juan Luis Moyano]

Hi, im trying to setup ASTCC but I'm getting it difficult. I've
correctly set up the mysql database astcc and added a brand, trunk,
route and a card as follows:

brands
+--+--+--+--+--++--+--+
| name | language | inc  | publishednum | did  | markup | days | fee  |
+--+--+--+--+--++--+--+
| FWD  | es   | 6| 4| 4|  0 | 30   |0 |
+--+--+--+--+--++--+--+

trunks
+--+--+-+
| name | tech | path|
+--+--+-+
| FWD  | IAX2 | 657XXX:[EMAIL PROTECTED] |
+--+--+-+

routes
+-+---++-+-+--+
| pattern | comment   | trunks | connectcost | includedseconds | cost |
+-+---++-+-+--+
| ^4. | FWD   | FWD|   0 |   0 |  150 |
+-+---++-+-+--+

-Added a card with $25 credit, using 'FWD' brand.

extensions.conf
---
[outbound-fwd]
;
exten = _4.,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:1})
exten = _4.,2,Hangup()

iax.conf

register = 657XXX:[EMAIL PROTECTED]


The problem is that when, for example, I dial '4612' i get:

-- Executing DeadAGI(IAX2/[EMAIL PROTECTED]/3, astcc.agi|21|612) in new 
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
-- Playing 'digits/1' (language 'en')
-- AGI Script astcc.agi completed, returning 0
-- Executing Hangup(IAX2/[EMAIL PROTECTED]/3, ) in new stack

and i hear allison saying I'm sorry that is not a recognized phone
number, goodbye.

Anyone knows what could be happening right here?

Many thanks in advance.

-- 
Juan Luis Moyano
[EMAIL PROTECTED]

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Re: [Asterisk-Users] mini itx

[Matt Gibson]

jltaylor wrote:


I've seen the embedded posts.
Is anyone running Asterisk on the MINI ITX?
 

Not directly related, but I got OpenBSD to boot on a CF card , on my 
Soekris this weekend.


Soekris is also selling units with the sangoma card as a daughterboard, 
might be a cheaper/quiter alternative

to Mini ITX if you don't have to transcode.

Matt


--
Matt Gibson
VOIP Director
Voxip.ca A Division of NJ Tech Solutions
Mobile: 1.613.868.9318
Tel: 1.314.480.4550 ex 6400
Toll Free: 1.888.999.4678 ex 6400
Email: [EMAIL PROTECTED]
Fax: 1.613.761.1828

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