[asterisk-users] Calls audio stops with latest Gigaset C450IP firmware + voicemail

[gincantalupo]

Hi,
I'm using Asterisk 1.2.18 on a Debian Etch box. I've noticed a very 
strange fact which causes a bad prob. When I get an inbound call, I make 
4 phones ring at the same time, one is a Snom while others are Gigaset 
C450IP with _latest firmware_.
When I get a call and answer with the Gigaset, a second call going to 
voicemail makes the first call received on the gigaset C450IP stop audio 
(it seems the call does not drop).
This does not happen if I answer the first call with the snom or with 
Siemens using older firmware or when gigaset are called in cascade 
sequence (Dial(SIP/1) then another Dial(SIP/2), etc...)
Is there anybody who can avoid this strange behaviour (I need to make 
all phones ring together)?

TIA

Giorgio

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Re: [asterisk-users] Ring/Off-hook in strange state 6

[Dimitri Volski]

Hi,

I had a similar problem when I had signalling set at FXS_KS for my 4 FXS 
port TDM400P card. I've read long time ago that the signalling in 
Australia (where I am) is FXS_LS, so that solved that for me. Try 
different signalling methods, hopefully that will solve your problem.

Apparently, the Digium tech support has a 2 week queue of requests.

Cheers,
Dimitri


Alex Mcdowell wrote:
 Can anybody at least point me in a direction??

 On 6/25/07, Alex Mcdowell [EMAIL PROTECTED] wrote:
   
 I don't think my cards are bad, but maybe there is a problem with the
 one. It has been two weeks since I put my ticket in with Digium...and
 still no word. I am starting to get frustrated.

 On 6/22/07, Daniel Hazelbaker [EMAIL PROTECTED] wrote:
 
 Alex,


   I had this problem with a new TDM2400 card that we purchased.  
 Specifically I would get that message and then it would pick up the ringing 
 line AND the line next to it.  Basically, lines 1  2 had been cross-linked 
 somehow.  After a few weeks of trouble-shooting with Digium tech support 
 they cross-shipped me a new card and the problem (and that message) went 
 away.


 Daniel Hazelbaker
 High Desert Church



 On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote:



 HI I have two servers both of which get this message on one of the lines.

 Ring/Off-hook in strange state 6. The one server seems to be ok with it, but

 the other one when an extension picks up there is no one there and the

 incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like

 someone had suggested, but it didn't do anything. I also upgraded zaptel to

 the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to

 no, as well as busydetect=no. This is a major problem since this box only

 has 1 other line, but at least it works. I can't seem to find much info on

 this issue. I can't believe others haven't run into it.  I started a ticket

 with digium, but I guess they are pretty backed up. Here is what I am

 getting in the CLI:  Thanks for any help -Alex

 -- Starting simple switch on 'Zap/4-1'

 -- Executing Wait(Zap/4-1, 1) in new stack

 -- Executing Answer(Zap/4-1, ) in new stack

 -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack

 -- Called 4125

 Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event:

 Ring/Off-hook in strange state 6 on channel 4

 -- SIP/4125-09559118 is ringing

 Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event:

 Ring/Off-hook in strange state 6 on channel 4

 -- SIP/4125-09559118 answered Zap/4-1

   == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1'

 -- Hungup 'Zap/4-1'

 -- Starting simple switch on 'Zap/4-1'

 -- Executing Wait(Zap/4-1, 1) in new stack

 -- Executing Answer(Zap/4-1, ) in new stack

 -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack

 -- Called 4125

 Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event:

 Ring/Off-hook in strange state 6 on channel 4

 -- SIP/4125-09559118 is ringing

 Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event:

 Ring/Off-hook in strange state 6 on channel 4

 -- SIP/4125-09559118 answered Zap/4-1

   == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1'

 -- Hungup 'Zap/4-1'

 -- Starting simple switch on 'Zap/4-1'

 -- Executing Wait(Zap/4-1, 1) in new stack

 -- Executing Answer(Zap/4-1, ) in new stack

 -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack

 -- Called 4125

 -- SIP/4125-09559118 is ringing
   



-- 
This message has been scanned for viruses and
dangerous content by Mail Call antivirus software, and is
believed to be clean.


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[asterisk-users] registering Asterisk on SIP/Nortel MCS thing

[Kate Kretz]

hello there...

our telecom sold us VoIP-numbering, managed by Nortel MCS
I successfully registered Ekiga to it (
http://sol.chel.skbkontur.ru/ekiga.png)

What exactly do I have to write in sip.conf to make Asterisk register on
this SIP ?

Cheers,
Kate
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Re: [asterisk-users] ooh323 hang up after the call is answered

[Kate Kretz]

ooh323c requires You to put the following lines:

disallow=all
allow=ulaw
allow=g723.1
..

otherwise it doesn't work

On 2/24/07, Guillermo Salas M. [EMAIL PROTECTED] wrote:


Solved... installed chan_oh323 :)


http://www.ecualug.org/?q=2006/02/28/comos/asterisk_1_2_4_agregando_soporte_para_el_protocolo_h_323

I don't know why ooh323 does not work.

Regards,


On Fri, 2007-02-23 at 11:21 -0500, Guillermo Salas M. wrote:
 I fogot, the H.323 device is one Antek networks INC with two fxo ports.

 Regards,

 On Fri, 2007-02-23 at 11:07 -0500, Guillermo Salas M. wrote:
  Hi,
 
  I'm trying to make ooh323 works with one asterisk box running 1.2.15
  version.
 
  I can ring from a h.323 to SIP and SIP to H.323, but when the call is
  finished when the phone is answered.
 
  This is the log when I call from the H.323 device to a SIP device:
 
  Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing
  Dial(OOH323/Telconet Mantaer-c5f8, SIP/666|30|TtrwWC) in new stack
  Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on RTP to 524288
  Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on VRTP to 524288
  Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Outgoing Call for 666
  Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Called 666
  Feb 23 10:57:32 DEBUG[6096] channel.c: Driver for channel
  'OOH323/Telconet Mantaer-c5f8' does not support indication 3,
emulating
  it
  Feb 23 10:57:32 DEBUG[6096] channel.c: Prodding channel
'OOH323/Telconet
  Mantaer-c5f8'
  Feb 23 10:57:32 DEBUG[6096] channel.c: Scheduling timer at 160 sample
  intervals
  Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping
  retransmission (but retaining packet) on
  '[EMAIL PROTECTED]' Request 102: Found
  Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping
  retransmission (but retaining packet) on
  '[EMAIL PROTECTED]' Request 102: Found
  Feb 23 10:57:32 DEBUG[6068] channel.c: Avoiding initial deadlock for
  'SIP/666-098cde60'
  Feb 23 10:57:32 VERBOSE[6096] logger.c: -- SIP/666-098cde60 is
  ringing
  Feb 23 10:57:36 DEBUG[6079] chan_sip.c: Auto destroying call
  '[EMAIL PROTECTED]'
  Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Acked pending invite 102
  Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Stopping retransmission on
  '[EMAIL PROTECTED]' of Request 102: Match
  Found
  Feb 23 10:57:40 DEBUG[6079] chan_sip.c: build_route: Contact hop:
  Guillermo Salas M sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp
  Feb 23 10:57:40 VERBOSE[6096] logger.c: -- SIP/666-098cde60
answered
  OOH323/Telconet Mantaer-c5f8
  Feb 23 10:57:40 WARNING[6096] src/chan_h323.c: Don't know how to
  indicate condition -1 on ooh323c_1
  Feb 23 10:57:40 DEBUG[6096] channel.c: Scheduling timer at 0 sample
  intervals
  Feb 23 10:57:40 DEBUG[6096] channel.c: Didn't get a frame from
channel:
  OOH323/Telconet Mantaer-c5f8
  Feb 23 10:57:40 DEBUG[6096] channel.c: Bridge stops bridging channels
  OOH323/Telconet Mantaer-c5f8 and SIP/666-098cde60
  Feb 23 10:57:40 DEBUG[6096] chan_sip.c: update_call_counter(666) -
  decrement call limit counter
  Feb 23 10:57:40 DEBUG[6096] app_dial.c: Exiting with
DIALSTATUS=ANSWER.
  Feb 23 10:57:40 VERBOSE[6096] logger.c:   == Spawn extension
  (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8'
in
  macro 'dial'
  Feb 23 10:57:40 VERBOSE[6096] logger.c:   == Spawn extension
  (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8'
in
  macro 'exten-vm'
  Feb 23 10:57:40 VERBOSE[6096] logger.c:   == Spawn extension
  (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8'
 
 
  And the h323 log:
 
  10:57:32:717  Created a new call (incoming, ooh323c_1)
  10:57:32:753  Received SETUP message (incoming, ooh323c_1)
  10:57:32:753  Tunneling disabled by remote endpoint. (incoming,
  ooh323c_1)
  10:57:32:753  Enabled RFC2833 DTMF capability for (incoming,
ooh323c_1)
  10:57:32:754  Sent Message - CallProceeding (incoming, ooh323c_1)
  10:57:32:754  Sent Message - Alerting (incoming, ooh323c_1)
  10:57:40:475  Cmd connection accepted
  10:57:40:476  Processing Answer Call command for ooh323c_1
  10:57:40:476  Creating H245 listener
  10:57:40:476  H245 listener creation - successful(port 12031)
(incoming,
  ooh323c_1)
  10:57:40:476  H.245 Listerner socket being monitored (incoming,
  ooh323c_1)
  10:57:40:476  Sent Message - Connect (incoming, ooh323c_1)
  10:57:40:476  H.245 Listerner socket being monitored (incoming,
  ooh323c_1)
  10:57:40:501  H.245 connection established (incoming, ooh323c_1)
  10:57:40:501  Sent Message - TerminalCapabilitySet (incoming,
ooh323c_1)
  10:57:40:502  Sent Message - MasterSlaveDetermination (incoming,
  ooh323c_1)
  10:57:40:538  Sent Message - TerminalCapabilitySetAck (incoming,
  ooh323c_1)
  10:57:40:542  Master Slave Determination received (incoming,
ooh323c_1)
  10:57:40:542  MasterSlaveDetermination done - Slave(incoming,
ooh323c_1)
  10:57:40:542  Sent Message - MasterSlaveDeterminationAck (incoming,
  ooh323c_1)
  

[asterisk-users] Work

[Paul Hales]

Looking for a full time Asterisk tech to work in Melbourne, Australia.

Full time, immediate start.

Anyone?

PaulH


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[asterisk-users] fail to load modules

[clive.chan\(Alpha Trilogies Networks\)]

Hi all, 

I am a bit out with the Asterisk 1.4.4, after I complied and installed the
Asterisk and I got such error messages 

[Jun 28 16:56:19] WARNING[28625] res_smdi.c: No SMDI interfaces are
available to listen on, not starting SDMI listener.

[Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module
'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined
symbol: ast_rtp_bridge

[Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module
'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so:
undefined symbol: option_verbose

 

I got nothing error during installation of asterisk-addons-1.4.2 after I had
change the Make file on the chan_ooh323.so.1.0.1. 

 

Tried;

I tried to define noload to the chan_00h323.so and res_config_mysql.so, my
asterisk start but give me others problems as bellowing...

[Jun 28 17:03:12] WARNING[28637] loader.c: Module 'cdr_addon_mysql.so' did
not register itself during load

[Jun 28 17:03:12] WARNING[28637] loader.c: Module 'cdr_addon_mysql.so' could
not be loaded.

[Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_addon_sql_mysql.so'
did not register itself during load

[Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_addon_sql_mysql.so'
could not be loaded.

[Jun 28 17:03:12] WARNING[28637] loader.c: Module 'format_mp3.so' did not
register itself during load

[Jun 28 17:03:12] WARNING[28637] loader.c: Module 'format_mp3.so' could not
be loaded.

[Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_saycountpl.so' did
not register itself during load

[Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_saycountpl.so' could
not be loaded.

 

 

 

 

Can some one shares experience ??

 

 

 

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[asterisk-users] error while compiling asterisk-1.2.19

[Goke Aruna]

hi,

I am installing the asterisk-1.2.19 with zaptel-1.2.8 on FC5.

I got install installed ok.. after i had disable the xpp_usb module.

However, when i try to compile asterisk and having this error

I will be glad for your kind response.

Goksie

chan_zap.c: In function âpri_dchannelâ:
chan_zap.c:9203: error: âpri_event_setup_ackâ has no member named âcallâ
make[1]: *** [chan_zap.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-1.2.19/channels'
make: *** [subdirs] Error 1

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Re: [asterisk-users] voicemail.conf serveremail

[Dave Bour]

vm_general.conf is where I've set mine (freepbx installation)
D.



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Pfeifer
Sent: Thursday, June 28, 2007 12:03 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] voicemail.conf serveremail


Hello,

I was wondering if there is a way to change the From address (not just
the Return-Path) for voicemail notification emails in Asterisk.

It looks like the serveremail directive in voicemail.conf just changes
the Return-Path.

I'm looking for something analogous to the -r option in mailx, for
example.  I need this since the mail server I'm using requires the
sender to be on the system.

Any advice would be appreciated. 

Thanks

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[asterisk-users] CDR and call transfer

[Grigoriy Puzankin]

Hello,

I'm using digium E1 cards and serving SIP users at Asterisk. After the 
following call (see below) CDR shows two records. First looks as 
outbound call, but the second - as inbound call. Is it a bug or intended 
behavior?

Call flow:

SIP (ext: 100) - ZAP (national number)
SIP (ext: 100) transfers to SIP (ext: 200)
SIP (ext: 200) - ZAP (national number).

In CDR it looks like

SIP (ext: 100) - ZAP (national number)
ZAP (national number) - SIP (ext: 200)

How to identify the second CDR as outbound call?

Best regards,
--
Grigoriy Puzankin

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Re: [asterisk-users] Missing 'init keys' command

[Jonathan Unai Marquez]

Thanks for your answer Jared, but I also tried that with no luck:

Connected to Asterisk 1.4.5 currently running on moe (pid = 22879)
-- Remote UNIX connection
Verbosity is at least 6
moe*CLI keys show
No such command 'keys show' (type 'help' for help)

Any clue of what can be wrong with my intallation?

J


Jonathan Unai Marquez wrote:
 Hi,

 I have two new Asterisk installations (1.4.4 and 1.4.5) and I have 
 created rsa keys and they can now see each other as online peers:

 moe*CLI iax2 show peers
 Name/UsernameHost Mask Port  
 Status
 bart 192.168.2.201   (S)  255.255.255.255  4569  
 OK (48 ms)
 1 iax2 peers [1 online, 0 offline, 0 unmonitored]


 but on the 1.4.5 instllation I cannot execute 'show keys' neither 
 'init keys'

 moe*CLI init keys
 No such command 'init keys' (type 'help' for help)

 This may be the reason that  I cannot place calls from one Asterisk to 
 the other.

 chan_iax2.c:7285 socket_process: I don't know how to authenticate moe 
 to 192.168.2.201

 thanks in advance,
 Jonathan.




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[asterisk-users] Fax passthrough howto codec upspeed

[Ivoc]

Hello everybody,
   
  Just was wondering if somebody can help for G711 fax passthrough w/ asterisk.
  The issue I have is regarding codec upspeed when the call is already 
connected using G729 for example. The setup is 
fax---ATA---asterisk---Cisco---fax
   
  When codec upspeed should happen, ATA or Cisco will send a G711 reINVITE 
causing the codec to be switched over, but asterisk does NOT transmit this 
reINVITE into the other voice leg so both voice legs regarding asterisk will be 
switched to G711. This whay the call is transcoded which is not desirable.
   
  Any ideas how to fix this?
   
   

   
-
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Re: [asterisk-users] Voicestick / i2telecom.com

[Huw Richards]

Sip registration started working again at 21.46 EDT 6/27/07 so it must
have been a problem with Voicestick - no information on their website
however.



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Huw
Richards
Sent: Wednesday, June 27, 2007 20:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicestick / i2telecom.com


I have one of their free pre-pay accounts i.e. no monthly charge. Still
have the origianl $5 signup credit as I've never made an outbound call
via voicestick. I only use the account for the inbound number. 
 
Maybe the inability to setup voicemail on the voicestick server is an
indication that there is something wrong with my account.



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Bour
Sent: Wednesday, June 27, 2007 19:46
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Voicestick / i2telecom.com



The most obvious question first. Your account is paid up to date?

Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]

For those who just want it to work...
Giving you complete IT peace of mind.

(Sent via Blackberry - hence message may be shorter than my usual
verbose responses)
PIN 4cc364db (as of March 24, 2007) 

- Original Message -
From: [EMAIL PROTECTED]
[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Sent: Wed Jun 27 19:21:40 2007
Subject: [asterisk-users] Voicestick / i2telecom.com

Hello,

I have been using Voicestick inbound (no outbound) successfully for the
last few months.

Noticed in my logfile that sip registration failed on 6/27/07 at 3am EDT
and no successful registration since. Calls to my number eventually
timeout as I don't have voicemail setup - as the first step in trouble
shooting I tried to enable voicemail on the voicestick website but this
fails also Transaction Failed. Please try again later.

Nothing changed in my config. Asterisk 1.2.18.

Can anyone confirm that there's an outage with Voicestick inbound?

Huw

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[asterisk-users] Asterisk 1.4.5 Inserting Random Digits in Dialed Number!

[Russell Brown]

Eeeeck!   Asterisk is inserting random digits in dialed numbers.

So far I've seen it insert a 2 after the STD (area) code and insert an
extra 6 or 7 in the STD code.  It's pretty repeatable although the
inserted number changes.

My Config is: Asterisk 1.4.5, Zaptel 1.4.3, Digium TE205P (rev 02). 

There's an ISDN PBX on the second span and a BRI euroisdn on the first.

Calls from the ISDN PBX (Network Alchemy Argent Office FWIW) get put
into  the 'alchemy' context which contains:

[alchemy]
;
include = dial_pstn
;
;   Dial Out to PSTN
;
[dial_pstn]
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Congestion
;

Pretty simple and didn't cause problems before I went to 1.4.4/5

Here's the console log from a failing call (non-significant numbers
obscured to protect the guilty!).

 *CLI -- Accepting overlap call from '1780471800' to '0173' on channel 
0/15, span 2
   -- Starting simple switch on 'Zap/46-1'
 [Jun 28 12:00:32] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '3' received 
on Zap/46-1, duration 0 ms
 [Jun 28 12:00:32] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '2' received 
on Zap/46-1, duration 0 ms
 [Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '3' received 
on Zap/46-1, duration 0 ms
 [Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '5' received 
on Zap/46-1, duration 0 ms
 [Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end 'x' received 
on Zap/46-1, duration 0 ms
 [Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end 'x' received 
on Zap/46-1, duration 0 ms
 [Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end 'x' received 
on Zap/46-1, duration 0 ms
   -- Executing [EMAIL PROTECTED]:1] Dial(Zap/46-1, 
Zap/g1/017332235xxx) in new stack
   -- Requested transfer capability: 0x00 - SPEECH
   -- Called g1/017332235xxx
   -- Zap/1-1 is proceeding passing it to Zap/46-1

As you can see, the ISDN PBX passed 01733 235xxx but Asterisk decided to
dial 01733 2235xxx (an extra '2' after the STD code).


Arrgh!!!  This is driving both us and the people who're getting
called incorrectly potty :-(

Can anyone help?  Thanks in advance.

-- 
 Regards,
 Russell
 
| Russell Brown  | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com  |
| Peterborough, England  | WWW Play: http://www.ruffle.me.uk |
 

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[asterisk-users] registering Asterisk on SIP/Nortel MCS server

[Kate Kretz]

hello there...

our telecom sold us VoIP-numbering, managed by Nortel MCS
I successfully registered Ekiga to it (
http://sol.chel.skbkontur.ru/ekiga.png)

What exactly do I have to write in sip.conf to make Asterisk register on
this SIP ?

Cheers,
Kate
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Re: [asterisk-users] error while compiling asterisk-1.2.19

[Tzafrir Cohen]

On Thu, Jun 28, 2007 at 11:30:04AM +0100, Goke Aruna wrote:
 hi,
 
 I am installing the asterisk-1.2.19 with zaptel-1.2.8 on FC5.

Zaptel 1.2.18, right?

 
 I got install installed ok.. after i had disable the xpp_usb module.

Are you using the default kernel of FC5 (2.6.15) or the one in the
updates?

 
 However, when i try to compile asterisk and having this error
 
 I will be glad for your kind response.
 
 Goksie
 
 chan_zap.c: In function âpri_dchannelâ:
 chan_zap.c:9203: error: âpri_event_setup_ackâ has no member named âcallâ
 make[1]: *** [chan_zap.o] Error 1
 make[1]: Leaving directory `/usr/src/asterisk-1.2.19/channels'
 make: *** [subdirs] Error 1

What version of libpri do you have?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] error while compiling asterisk-1.2.19

[Goke Aruna]

Tzafrir Cohen wrote:
 On Thu, Jun 28, 2007 at 11:30:04AM +0100, Goke Aruna wrote:
 hi,

 I am installing the asterisk-1.2.19 with zaptel-1.2.8 on FC5.
 
 Zaptel 1.2.18, right?

YES .. MY zaptel is 1.2.18
 
 I got install installed ok.. after i had disable the xpp_usb module.
 
 Are you using the default kernel of FC5 (2.6.15) or the one in the
 updates?

default kernel-smp and its devel

 
 However, when i try to compile asterisk and having this error

 I will be glad for your kind response.

 Goksie

 chan_zap.c: In function âpri_dchannelâ:
 chan_zap.c:9203: error: âpri_event_setup_ackâ has no member named âcallâ
 make[1]: *** [chan_zap.o] Error 1
 make[1]: Leaving directory `/usr/src/asterisk-1.2.19/channels'
 make: *** [subdirs] Error 1
 
 What version of libpri do you have?
 

libpri-1.2.1


Goksie

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Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware

[Olivier]

2007/6/27, Greg Oliver [EMAIL PROTECTED]:


On Tue, 2007-06-26 at 21:45 +0200, Olivier wrote:
 Hi,

 Has anyone met any success, installing localized (ie non-english)
 menus within SIP firmware enabled Cisco 7941 ?

 Those phones seem to be trying to download localized menus from Cisco
 Call Manager but as they are managed by an Asterisk server, I'm
 looking for a workaround.
 Any advice ?

 Regards

Actually Cisco only sendx xml for certain things.  It uses a modified
SIP stack and it's native SCCP stack to provision button templates,
softkeys, etc..

I did hours of packet captures to try and get the info, but it is
embedded into the call control stack of their phones.

If you read the chan_sccp code a bit, it has a few different button
layout options, that are encoded in the SCCP driver and not xml files.

I wish they would go to all config files, but I doubt they will...

-Greg

So, if you ever use a Cisco SIP Phone with an Asterisk server, it's not

possible to localize menus, soft keys, and so on ?
Cheers
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Re: [asterisk-users] error while compiling asterisk-1.2.19

[Tzafrir Cohen]

On Thu, Jun 28, 2007 at 01:28:28PM +0100, Goke Aruna wrote:
 Tzafrir Cohen wrote:
  On Thu, Jun 28, 2007 at 11:30:04AM +0100, Goke Aruna wrote:
  hi,
 
  I am installing the asterisk-1.2.19 with zaptel-1.2.8 on FC5.
  
  Zaptel 1.2.18, right?
 
 YES .. MY zaptel is 1.2.18
  
  I got install installed ok.. after i had disable the xpp_usb module.
  
  Are you using the default kernel of FC5 (2.6.15) or the one in the
  updates?
 
 default kernel-smp and its devel

Generally it is recommended to get updates. While I cannot point you to
a specific bug relevant to Asterisk, there are probably some security
updates and such.

 
  
  However, when i try to compile asterisk and having this error
 
  I will be glad for your kind response.
 
  Goksie
 
  chan_zap.c: In function âpri_dchannelâ:
  chan_zap.c:9203: error: âpri_event_setup_ackâ has no member named âcallâ
  make[1]: *** [chan_zap.o] Error 1
  make[1]: Leaving directory `/usr/src/asterisk-1.2.19/channels'
  make: *** [subdirs] Error 1
  
  What version of libpri do you have?
  
 
 libpri-1.2.1

Get libpri 1.2.4 .

Generally build Asterisk vs. the version of libpri that was the latest
at the time it was released. Note that you only actually need libpri if
you have digital zaptel hardware, e.g: a T1/E1 card.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Error While Calling AGI

[Nitesh Divecha]

Hello All,
Please anyone can help me with this error...

Found some strange problem while Asterisk trying to call the AGI file.
If I pick up the call on the first attempt, it will execute my AGI file 
properly.
But if I don't pick up the call and let Asterisk call me again, it adds 
StartRetry: 3700 1 (1182971439) next to my AGI file name, which will 
cause the AGI to fail to execute.

-- Attempting call on SIP/5181 for application AGI(recordvoice.php) 
(Retry 1)
   -- Attempting call on SIP/5181 for application 
AGI(recordvoice.phpStartRetry: 3700 1 (1182971439)) (Retry 2)
   -- Attempting call on SIP/5181 for application 
AGI(recordvoice.phpStartRetry: 3700 1 (1182971439)) (Retry 3)
   Channel SIP/08f39360 was answered.
   Launching AGI(recordvoice.phpStartRetry: 3700 1 (1182971439)) on 
SIP/08f39360
   -- Launched AGI Script 
/var/lib/asterisk/agi-bin/recordvoice.phpStartRetry: 3700 1 (1182971439)
   -- AGI Script recordvoice.phpStartRetry: 3700 1 (1182971439) 
completed, returning 0

Can anyone help? By the way I am executing using *.call file.

File make.call: -
Channel: SIP/5181
MaxRetries: 3
RetryTime: 30
WaitTime: 15
Application: AGI
Data: recordvoice.php

Cheers,
Nitesh




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[asterisk-users] FXS channel bank

[pixiesfr]

hello,

We looking for a channel bank to connect 120 analogs phones, in SIP to 
an Asterisk ..

Did someone have some product in mind.

Thanks for you help..

bye bye

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Re: [asterisk-users] FXS channel bank

[Jerry Jones]

On Jun 28, 2007, at 8:00 AM, pixiesfr wrote:

 hello,

 We looking for a channel bank to connect 120 analogs phones, in SIP to
 an Asterisk ..

 Did someone have some product in mind.

A channel bank must connect via a T1 by definition, which would then  
give you 24 phone lines per T1. This would require 5 T1 connected to  
your asterisk server. OK 4 if E1 as it probably is in your case.

However with your requirement for SIP you are looking for a gateway  
to connect your phones. Most are 24 port, though some are 48 port.  
Names to look at would be Carrier Access, Audiocodes, Vega etc.

I do like the Vega unit except for their support - or lack thereof -  
here in the US. They do have both 24 and 48 port units.

Your other option would be to do GR303 which would allow you to hang  
many lines off a few T1/E1 circuits, except it is definately not SIP.  
If phones are not at location of your asterisk server and you really  
want to do sip, it may be simpler for this many phones to install an  
additional asterisk server at the remote location and install a quad  
port T1/E1 card and hang channel banks off it.

Good Luck

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Re: [asterisk-users] Using MSAccess to dial on a Zap line

[Martin Smith]

We're really happy with SIP Tapi:

http://sourceforge.net/projects/siptapi/
http://www.enum.at/index.php?id=479

We've been trying to document our setup at:
http://projects.bebr.ufl.edu/wiki/AsteriskTAPI

We like it as it requires no manager interface (it uses SIP Refer) to be
turned on. It can be a little awkward with voicemail if you use it to
call yourself, but you can pick one extension that will call everyone.

 Is there a better way to do this? The complaint we are 
 getting now is the call 
 rep doesn't want their phone to ring when making a call. Can 
 the manager 
 interface give a phone number to dial on an off hook Zap line?

Unfortunately, we haven't found a way around this either, except to have
autoanswer on the phone. Some phones could probably be configured to
autoanswer for some callerids or lines, but if you find anything else
interesting about this or non-ringing auto-dialing, I'd be curious to
hear as well!

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221

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Re: [asterisk-users] CDR and call transfer

[Grigoriy Puzankin]

I did a lot of googling until I found this thread:

http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html

--
Grigoriy Puzankin

Grigoriy Puzankin wrote:
 Hello,

 I'm using digium E1 cards and serving SIP users at Asterisk. After the 
 following call (see below) CDR shows two records. First looks as 
 outbound call, but the second - as inbound call. Is it a bug or intended 
 behavior?

 Call flow:

 SIP (ext: 100) - ZAP (national number)
 SIP (ext: 100) transfers to SIP (ext: 200)
 SIP (ext: 200) - ZAP (national number).

 In CDR it looks like

 SIP (ext: 100) - ZAP (national number)
 ZAP (national number) - SIP (ext: 200)

 How to identify the second CDR as outbound call?

 Best regards,
 --
 Grigoriy Puzankin

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[asterisk-users] Query

[sanchal . singh]

Hi,
 I am trying to establish call through sip phone between two PC  connected 
to linux box on which asterisk server is running
  
   1st PC is having IP Adress : 192.168.1.149
   2nd PC is having IP Adress : 192.168.1.53

   Now, I am tying to dial from 1st PC to 2nd PC

   I am trying to dial from 1st PC to 2nd PC through asterisk server
   The problem is 1st PC is calling directly to 2nd PC not through asterisk 
server

  I am doing the following additions in configuration files

 1) sip.conf

[general]
context=sip
bindport=5060   
bindaddr=0.0.0.0   

 [phone1]
 type=friend
 host=192.168.1.149
 port=5060
 nat=yes
 dtmfmode=rfc2833
 context=sip

 [phone2]
 type=friend
 host=192.168.1.53
 port=5060
 nat=yes
 dtmfmode=rfc2833
 context=sip

2) extensions.conf
exten = 11,1,Dial(SIP/phone2,20,tr)

Now, I am calling from sip phone1 by name [EMAIL PROTECTED]
 It is not being called through asterisk server running on linux m/c. It is 
calling directly. As, I am running sip debub but no packet dumping is taking 
place. Can anybody will tell me the error I am doing.
Thanx and regards
sanchal
  


   

  










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[asterisk-users] Caller ID Spoofing to be banned in the USA

[Dean Collins]

Anyone running caller id spoofing applications in the USA running
asterisk?

 

Then it's time to move them to Canada or similar.

http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-t
o-be-outlawed.html 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

 

 

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[asterisk-users] E1 not coming up

[Alexander Zielke]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hello List,

since some days i run into the problem that one span on a TE407P is not
comming up correctly. With intense debug on that span i get:

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended) ]
 0 bytes of data
- -- Got SABME from network peer.
Sending Unnumbered Acknowledgement

 [ 02 01 73 ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
 0 bytes of data
- -- Restarting T203 counter
- -- Restarting T203 counter
  == Primary D-Channel on span 4 up

this looks good so far, but actually, these 2 messages repeat over and
over again.

At first i thought that there might be a problem with the carrier, so i
just swapped 2 of the connected PRIs, but the problem stayed persistent
on the same span.

Anyone ever run into a Problem like this aswell? I haven't found any
solution so far so i ask here.

asterisk/zaptel/libpri are all lates stable out of the 1.2 branch


and these are the related configs:

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

span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

span=3,0,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78

span=4,0,0,ccs,hdb3,crc4
bchan=94-108,110-124
dchan=109

span=5,0,0,ccs,hdb3,crc4
bchan=125-139,141-155
dchan=140

span=6,0,0,ccs,hdb3,crc4
bchan=156-170,172-186
dchan=171

span=7,0,0,ccs,hdb3,crc4
bchan=187-201,203-217
dchan=202

span=8,0,0,ccs,hdb3,crc4
bchan=218-232,234-248
dchan=233

loadzone=de
defaultzone=de

- -

/etc/asterisk/zapata.conf:
[channels]
busydetect=no
callprogress=no
switchtype=euroisdn
immediate=no
overlapdial=yes
echocancel=no
signalling=pri_cpe
relaxdtmf=yes
resetinterval=60
txgain=2.0

context=incoming
group = 1
channel=1-15,17-31
channel=32-46,48-62
channel=63-77,79-93
channel=94-108,110-124
channel=125-139,141-155
channel=156-170,172-186
channel=187-201,203-217
channel=218-232,234-248

- --
with regards
Alexander Zielke
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFGg8tc2mzHt+9iqYMRAhA/AJoD5X/uu5mC1psXH1KELAzZYg7ZTQCfT75h
/k/OhI1/zY1G4ExYmbVMg0w=
=k8fU
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Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware

[Greg Oliver]

On Thu, 2007-06-28 at 14:52 +0200, Olivier wrote:
 
 2007/6/27, Greg Oliver [EMAIL PROTECTED]:
 On Tue, 2007-06-26 at 21:45 +0200, Olivier wrote:
  Hi,
 
  Has anyone met any success, installing localized (ie
 non-english)
  menus within SIP firmware enabled Cisco 7941 ?
 
  Those phones seem to be trying to download localized menus
 from Cisco 
  Call Manager but as they are managed by an Asterisk server,
 I'm
  looking for a workaround.
  Any advice ?
 
  Regards
 
 Actually Cisco only sendx xml for certain things.  It uses a
 modified 
 SIP stack and it's native SCCP stack to provision button
 templates,
 softkeys, etc..
 
 I did hours of packet captures to try and get the info, but it
 is
 embedded into the call control stack of their phones. 
 
 If you read the chan_sccp code a bit, it has a few different
 button
 layout options, that are encoded in the SCCP driver and not
 xml files.
 
 I wish they would go to all config files, but I doubt they
 will... 
 
 -Greg
 
 So, if you ever use a Cisco SIP Phone with an Asterisk server, it's
 not possible to localize menus, soft keys, and so on ?
 Cheers
 ___

Not unless someone wants to add support for it in the SIP channel, which
I doubt.  I would be more than willing to provide the SIP messages that
a CallManager sends to accomplish it though.

-Greg


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Re: [asterisk-users] Query

[David Gomillion]

On 6/28/07, [EMAIL PROTECTED] 
[EMAIL PROTECTED] wrote:


Hi,
 I am trying to establish call through sip phone between two
PC  connected to linux box on which asterisk server is running

   1st PC is having IP Adress : 192.168.1.149
   2nd PC is having IP Adress : 192.168.1.53

   Now, I am tying to dial from 1st PC to 2nd PC

   I am trying to dial from 1st PC to 2nd PC through asterisk server
...

2) extensions.conf
exten = 11,1,Dial(SIP/phone2,20,tr)

Now, I am calling from sip phone1 by name [EMAIL PROTECTED]



Why? Shouldn't you just pick up Phone1 and dial 11? If you dial it by the
IP address, why would it go through Asterisk?

It is not being called through asterisk server running on linux m/c. It

is calling directly. As, I am running sip debub but no packet dumping is
taking place. Can anybody will tell me the error I am doing.



I am going to assume that the typo is in the above paragraph, and you really
mean sip debug. If not, that's another problem.

Thanx and regards

sanchal



Hope that helps,
David
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Re: [asterisk-users] Asterisk 1.4.5 Inserting Random Digits in Dialed Number!

[Eric \ManxPower\ Wieling]

Remove any relaxdtmf= options from your zapata.conf.

Russell Brown wrote:
 Eeeeck!   Asterisk is inserting random digits in dialed numbers.
 
 So far I've seen it insert a 2 after the STD (area) code and insert an
 extra 6 or 7 in the STD code.  It's pretty repeatable although the
 inserted number changes.
 
 My Config is: Asterisk 1.4.5, Zaptel 1.4.3, Digium TE205P (rev 02). 
 
 There's an ISDN PBX on the second span and a BRI euroisdn on the first.
 
 Calls from the ISDN PBX (Network Alchemy Argent Office FWIW) get put
 into  the 'alchemy' context which contains:
 
   [alchemy]
   ;
   include = dial_pstn
   ;
   ;   Dial Out to PSTN
   ;
   [dial_pstn]
   exten = _X.,1,Dial(Zap/g1/${EXTEN})
   exten = _X.,2,Congestion
   ;
 
 Pretty simple and didn't cause problems before I went to 1.4.4/5
 
 Here's the console log from a failing call (non-significant numbers
 obscured to protect the guilty!).
 
  *CLI -- Accepting overlap call from '1780471800' to '0173' on channel 
 0/15, span 2
-- Starting simple switch on 'Zap/46-1'
  [Jun 28 12:00:32] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '3' 
 received on Zap/46-1, duration 0 ms
  [Jun 28 12:00:32] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '2' 
 received on Zap/46-1, duration 0 ms
  [Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '3' 
 received on Zap/46-1, duration 0 ms
  [Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '5' 
 received on Zap/46-1, duration 0 ms
  [Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end 'x' 
 received on Zap/46-1, duration 0 ms
  [Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end 'x' 
 received on Zap/46-1, duration 0 ms
  [Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end 'x' 
 received on Zap/46-1, duration 0 ms
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/46-1, 
 Zap/g1/017332235xxx) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/017332235xxx
-- Zap/1-1 is proceeding passing it to Zap/46-1
 
 As you can see, the ISDN PBX passed 01733 235xxx but Asterisk decided to
 dial 01733 2235xxx (an extra '2' after the STD code).
 
 
 Arrgh!!!  This is driving both us and the people who're getting
 called incorrectly potty :-(
 
 Can anyone help?  Thanks in advance.
 


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Re: [asterisk-users] Updated Manual for Asterisk 1.4.x

[Eric \ManxPower\ Wieling]

GNUbie wrote:
 Hello all,
 
 Anybody can point me to the right URL where I can read an updated manual 
 for
 Asterisk 1.4.x?

Your best bet is to read UPGRADE.txt in the Asterisk source tree.  It 
should list most of the changes from 1.2.x to 1.4.x

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Re: [asterisk-users] Query

[Victor Toofic]

El Thu, Jun 28 de 2007 a las 20:18 +0530, [EMAIL PROTECTED] comentaba:
 Hi,
  I am trying to establish call through sip phone between two PC  
 connected to linux box on which asterisk server is running
   
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP Adress : 192.168.1.53
 
Now, I am tying to dial from 1st PC to 2nd PC
 
I am trying to dial from 1st PC to 2nd PC through asterisk server
The problem is 1st PC is calling directly to 2nd PC not through 
 asterisk server
 
   I am doing the following additions in configuration files
 
  1) sip.conf
 
 [general]
 context=sip
 bindport=5060   
 bindaddr=0.0.0.0   
 
  [phone1]
  type=friend
  host=192.168.1.149
  port=5060
  nat=yes
  dtmfmode=rfc2833
  context=sip
 
  [phone2]
  type=friend
  host=192.168.1.53
  port=5060
  nat=yes
  dtmfmode=rfc2833
  context=sip
 
 2) extensions.conf
 exten = 11,1,Dial(SIP/phone2,20,tr)
 
 Now, I am calling from sip phone1 by name [EMAIL PROTECTED]

I guess thats why the phones are talking directly: [EMAIL PROTECTED]

Either call extension '11' from phone1 or add a extension named 'phone2' to
extensions.conf and call that extension ('phone2') without the ip address.
Make sure your softphones are correctly configured: sip proxy address (*
address), username, etc.

Btw, both devices in sip.conf are declared as 'friend'.. thus you must specify
a secret (and optionally a username):

[phone2]
type=friend
username=phone2
secret=qwerty
host=192.168.1.53
port=5060
nat=yes
dtmfmode=rfc2833
context=sip

  It is not being called through asterisk server running on linux m/c. It 
 is calling directly. As, I am running sip debub but no packet dumping is 
 taking place. Can anybody will tell me the error I am doing.
 Thanx and regards
 sanchal
   

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[asterisk-users] RTCP NTP Clock skew

[John Millican]

Hello All,
I have Asterisk 1.4.5 running on a SuSE 10.3 x86_64  2.6.18.2-34
I upgraded from 1.4.2 to 1.4.5 on sunday the 24 of june and since have been 
getting:
Internal RTCP NTP clock skew detected: lsr=1402479300, now=1402675136, 
dlsr=196500 (2:998ms), diff=664
I see an entry in Mantis that Russell fixed code so that this will not show 
when it shouldn't. Would i be correct in assuming that if i pull a copy of 
1.4.5 from digium this weekend that this message will go away?  Also, just to 
show off my ignorance, what is this message telling me?  Is this simply a 
deference between unix time-t and NTP timestamps and therefore nothing of 
much concern?
JohnM


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Re: [asterisk-users] FXS channel bank

[Alex Balashov]

On Thu, 28 Jun 2007, Jerry Jones wrote:

 Your other option would be to do GR303 which would allow you to hang 
 many lines off a few T1/E1 circuits, except it is definately not SIP.

   Well, and it's probably worth pointing out that if you wanted to go the
GR.303 route, the devices on both ends -- and especially the service 
provider side -- is unlikely to be so inexpensive and simple as a mere
Asterisk server.

   You'd need some sort of DLC capable of doing GR.303, and, well, I don't 
know what supports GR.303 subscriber interfaces on the service side other 
than a bona fide Class 5 switch of some sort.  :-)

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

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Re: [asterisk-users] E1 not coming up

[Matthew Fredrickson]

What is the output of `cat /proc/zaptel/spannumberthatsbroken`

---
Matthew Fredrickson
Software Engineer
Digium, Inc.

On Jun 28, 2007, at 9:53 AM, Alexander Zielke wrote:

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Hello List,

 since some days i run into the problem that one span on a TE407P is  
 not
 comming up correctly. With intense debug on that span i get:

  [ 02 01 7f ]

  Unnumbered frame:
  SAPI: 00  C/R: 1 EA: 0
   TEI: 000EA: 1
M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced  
 mode
 extended) ]
  0 bytes of data
 - -- Got SABME from network peer.
 Sending Unnumbered Acknowledgement

 [ 02 01 73 ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
 0 bytes of data
 - -- Restarting T203 counter
 - -- Restarting T203 counter
   == Primary D-Channel on span 4 up

 this looks good so far, but actually, these 2 messages repeat over and
 over again.

 At first i thought that there might be a problem with the carrier,  
 so i
 just swapped 2 of the connected PRIs, but the problem stayed  
 persistent
 on the same span.

 Anyone ever run into a Problem like this aswell? I haven't found any
 solution so far so i ask here.

 asterisk/zaptel/libpri are all lates stable out of the 1.2 branch


 and these are the related configs:

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

 span=2,0,0,ccs,hdb3,crc4
 bchan=32-46,48-62
 dchan=47

 span=3,0,0,ccs,hdb3,crc4
 bchan=63-77,79-93
 dchan=78

 span=4,0,0,ccs,hdb3,crc4
 bchan=94-108,110-124
 dchan=109

 span=5,0,0,ccs,hdb3,crc4
 bchan=125-139,141-155
 dchan=140

 span=6,0,0,ccs,hdb3,crc4
 bchan=156-170,172-186
 dchan=171

 span=7,0,0,ccs,hdb3,crc4
 bchan=187-201,203-217
 dchan=202

 span=8,0,0,ccs,hdb3,crc4
 bchan=218-232,234-248
 dchan=233

 loadzone=de
 defaultzone=de

 - -

 /etc/asterisk/zapata.conf:
 [channels]
 busydetect=no
 callprogress=no
 switchtype=euroisdn
 immediate=no
 overlapdial=yes
 echocancel=no
 signalling=pri_cpe
 relaxdtmf=yes
 resetinterval=60
 txgain=2.0

 context=incoming
 group = 1
 channel=1-15,17-31
 channel=32-46,48-62
 channel=63-77,79-93
 channel=94-108,110-124
 channel=125-139,141-155
 channel=156-170,172-186
 channel=187-201,203-217
 channel=218-232,234-248

 - --
 with regards
 Alexander Zielke
 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.4.7 (GNU/Linux)
 Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

 iD8DBQFGg8tc2mzHt+9iqYMRAhA/AJoD5X/uu5mC1psXH1KELAzZYg7ZTQCfT75h
 /k/OhI1/zY1G4ExYmbVMg0w=
 =k8fU
 -END PGP SIGNATURE-

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Re: [asterisk-users] Updated Manual for Asterisk 1.4.x

[GNUbie]

Hello Eric,

On 6/28/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:



Your best bet is to read UPGRADE.txt in the Asterisk source tree.  It
should list most of the changes from 1.2.x to 1.4.x



What I just did was install the asterisk-doc package and I now have the
Asterisk Documentation for v1.4.5.  But the docs doesn't have a detailed
information about some commands and options used especially for the Dial().

Thanks anyway.
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Re: [asterisk-users] Using MSAccess to dial on a Zap line

[C F]

You could have the manager interface intiate the call to a local
channel that uses auto answer for your phone. That way it will be
answered automaticaly.



On 6/28/07, Martin Smith [EMAIL PROTECTED] wrote:
 We're really happy with SIP Tapi:

 http://sourceforge.net/projects/siptapi/
 http://www.enum.at/index.php?id=479

 We've been trying to document our setup at:
 http://projects.bebr.ufl.edu/wiki/AsteriskTAPI

 We like it as it requires no manager interface (it uses SIP Refer) to be
 turned on. It can be a little awkward with voicemail if you use it to
 call yourself, but you can pick one extension that will call everyone.

  Is there a better way to do this? The complaint we are
  getting now is the call
  rep doesn't want their phone to ring when making a call. Can
  the manager
  interface give a phone number to dial on an off hook Zap line?

 Unfortunately, we haven't found a way around this either, except to have
 autoanswer on the phone. Some phones could probably be configured to
 autoanswer for some callerids or lines, but if you find anything else
 interesting about this or non-ringing auto-dialing, I'd be curious to
 hear as well!

 Martin Smith, Systems Developer
 [EMAIL PROTECTED]
 Bureau of Economic and Business Research
 University of Florida
 (352) 392-0171 Ext. 221

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[asterisk-users] Linking Asterisk with another SIP PBX (or SIP Softswitch)

[bilal ghayyad]

Hi List;

If I need to do a trunk between Asterisk and another
SIP softswitch (so Asterisk will send a SIP calls to
that softswitch), then I have to configure this on the
sip.conf file or where exactly? And is it the same
when I configure iax trunk?

Should I determine the context in this case for this
SIP trunk? 

Regards
Bilal


   

Be a better Heartthrob. Get better relationship answers from someone who knows. 
Yahoo! Answers - Check it out. 
http://answers.yahoo.com/dir/?link=listsid=396545433

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Re: [asterisk-users] Call transfer feature

[Lee Jenkins]

satish patel wrote:
 Dear ALL
 
I want to transfer call from one phone 2 another 
 phone so this is asterisk feature or SIP Phone feature or endpoint 
 feature how can i transfer phone call from to another phone
 
 
 Rgd
 
 Satish patel
 

Check out this page:

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


-- 

Warm Regards,

Lee




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Re: [asterisk-users] Updated Manual for Asterisk 1.4.x

[Drew Gibson]

GNUbie wrote:

Hello Eric,

On 6/28/07, *Eric ManxPower Wieling* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:



Your best bet is to read UPGRADE.txt in the Asterisk source tree.  It
should list most of the changes from 1.2.x to 1.4.x


What I just did was install the asterisk-doc package and I now have 
the Asterisk Documentation for v1.4.5.  But the docs doesn't have a 
detailed information about some commands and options used especially 
for the Dial().



Detailed info on commands is available at the Asterisk console.

asterisk*CLI core show applications
   -= Registered Asterisk Applications =-
  AbsoluteTimeout: Set absolute maximum time of call
   AddQueueMember: Dynamically adds queue members
... Much Output ...

asterisk*CLI core show application dial
 -= Info about application 'Dial' =-

[Synopsis]
Place a call and connect to the current channel

[Description]
 Dial(Technology/resource[Tech2/resource2...][|timeout][|options][|URL]):
... Much Output ...

regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

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Re: [asterisk-users] Caller ID Spoofing to be banned in the USA

[J. Oquendo]

Dean Collins wrote:


Anyone running caller id spoofing applications in the USA running 
asterisk?


Then it’s time to move them to Canada or similar.

http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-to-be-outlawed.html 




Why it means nothing...

You're a carrier doing VoIP... Say a managed carrier. You
re-sell trunks. One of those trunks maintains their own PBX.
PBX admin decides to spoof out and is using a proxy say in
India. Hell make it Tor for that matter. What's to prosecute?
Prove it happened from where you say it did - remember the
burden is on the prosecution.

Now as the carrier (me) first thing I'm going to do is track
down which trunk it came from... Then go to that client...
So what happens if say the client was legitimately owned
and had various proxied addresses committing toll fraud.

Analogy... Gun dealer sells a .45 to an authorized gun
buyer. Gun owner leaves his gun at home. Someone breaks into
his home, cracks his gun safe, uses his gun for a crime,
re-enters and places the gun back in the safe. Now its
known it wasn't the gun owner because he was witnessed by
the court system and recorded say at jury duty... What do
you do, prosecute him? For what? Negligence?

It would be humorous to see how this plays out. To me its
more or less voting time let's sign pretend laws for
brownie points


--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
echo infiltrated.net|sed 's/^/sil@/g' 


Wise men talk because they have something to say;
fools, because they have to say something. -- Plato




smime.p7s
Description: S/MIME Cryptographic Signature
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[asterisk-users] Asterisk + hinting presence + macro

[Rob Schall]

I currently have about 50 polycom 501 phones on my asterisk setup. The 
dialplan is set to work with mysql (realtime), and all of the extensions 
for the phones route through the same macro (stdexten). This all works 
fine until I tried to set up notify status.

On voip-info, they say do something like...

,hint,SIP/
,1,Dial(SIP/)
blah blah blah

This functionality works fine. But what if you have a macro
s,hint,SIP/${ARG1}
s,1,Dial(SIP/${ARG1}

this adds a s hint which obviously doesn't work, instead of a hint for 
 as it should.

Any ideas?

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Re: [asterisk-users] RTCP NTP Clock skew

[Jared Smith]

On 6/28/07, John Millican [EMAIL PROTECTED] wrote:
 Would i be correct in assuming that if i pull a copy of
 1.4.5 from digium this weekend that this message will go away?

No... you'd have to pull the latest code from the 1.4 branch using
Subversion, or wait for 1.4.6 to be released.

-Jared

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Re: [asterisk-users] Asterisk + hinting presence + macro

[Jaswinder Singh]

It was due to changes in cdr in asterisk 1.4.5 previous version does not do
it .there is a fix on bugs.digium.com or you can wait till next release or
use asterisk 1.4.4

On 28/06/07, Rob Schall [EMAIL PROTECTED] wrote:


I currently have about 50 polycom 501 phones on my asterisk setup. The
dialplan is set to work with mysql (realtime), and all of the extensions
for the phones route through the same macro (stdexten). This all works
fine until I tried to set up notify status.

On voip-info, they say do something like...

,hint,SIP/
,1,Dial(SIP/)
blah blah blah

This functionality works fine. But what if you have a macro
s,hint,SIP/${ARG1}
s,1,Dial(SIP/${ARG1}

this adds a s hint which obviously doesn't work, instead of a hint for
 as it should.

Any ideas?

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[asterisk-users] Avoided deadlock for '0x864e70', 10 retries!

[ram]

Hi

iam using 1.2.X SVN

iam keep getting the below message

Jun 28 23:07:31 WARNING[2692]: channel.c:785 channel_find_locked: Avoided
deadlock for '0x864e70', 10 retries!

any help

ram
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Re: [asterisk-users] Asterisk + hinting presence + macro

[Rob Schall]

What was due to changes? I didn't read anything in the release notes 
about hinting in any newer versions (changes, etc). Do you have a link 
to this fix? and will this fix work with 1.2?


Rob

Jaswinder Singh wrote:
It was due to changes in cdr in asterisk 1.4.5 previous version does 
not do it .there is a fix on bugs.digium.com http://bugs.digium.com 
or you can wait till next release or use asterisk 1.4.4


On 28/06/07, *Rob Schall* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


I currently have about 50 polycom 501 phones on my asterisk setup. The
dialplan is set to work with mysql (realtime), and all of the
extensions
for the phones route through the same macro (stdexten). This all works
fine until I tried to set up notify status.

On voip-info, they say do something like...

,hint,SIP/
,1,Dial(SIP/)
blah blah blah

This functionality works fine. But what if you have a macro
s,hint,SIP/${ARG1}
s,1,Dial(SIP/${ARG1}

this adds a s hint which obviously doesn't work, instead of a
hint for
 as it should.

Any ideas?

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Re: [asterisk-users] Xorcom Bri 4 Port USB

[Tzafrir Cohen]

On Thu, Jun 28, 2007 at 10:59:34AM +1000, Nathan Dennis wrote:
 Thanks Tzafrir, that did the trick.
 But please note the that the bristuff patch from xorcom has broken 
 links in it. 

http://updates.xorcom.com/astribak/bristuff ? Updated and fixed, thanks 
for the note.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] network routing

[Ed Nuñez]

I have installed the Asterisk BE B.2.2 image file in a new server.  I need to 
make network routing changes.  However in their version of rPath (pound key) 
Digium has removed the netconfig command.  I am able to manually add the route 
with 

 

Route add default gw xxx.xxx.xxx.xxx however when I reboot I lose my routing.  
Does anyone know which conf file I need to edit in order to make this routing 
change permanent?

 

Thank you

 

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[asterisk-users] Avaya IP Office DTMF Issue

[Jon Farmer]

Hi

I have a client using a Avaya IP Office PBX that is taking a SIP trunk
from me terminating on a * box. It all works perfectly apart from DTMF.
Although you can hear the tones they don't seem to get recognised. I
have tried DTMF mode auto, inband, out of band and rfc2833 but no luck.
Any ideas?

Regards

Jon


-- 
Jon Farmer
Telford, Shropshire, UK

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[asterisk-users] Asterisk and IPv6

[Bent Bagger]

In October of last year Marc Blanchet of the Canadian company Viagénie
made a presentation on how he and others had build IPv6 support into
Asterisk and furthermore demonstrated that it worked. Marc Blanchet
went into some details on how it was done and the amount of work that
had gone into it.

A question is this connection:

When will these additions make their way into the Asterisk mainstream

I would like to use IPv6 in order to get around all issues connected
with NAT/Masquerading.

Kind regards,

Bent
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Re: [asterisk-users] network routing

[~Russell]

try to edit /etc/sysconfig/network-scripts/ifcfg-eth0  if u have eth0

if not try ifcfg-eth1 for eth1



On 6/29/07, Ed Nuñez [EMAIL PROTECTED] wrote:


 I have installed the Asterisk BE B.2.2 image file in a new server.  I
need to make network routing changes.  However in their version of rPath
(pound key) Digium has removed the netconfig command.  I am able to manually
add the route with



Route add default gw xxx.xxx.xxx.xxx however when I reboot I lose my
routing.  Does anyone know which conf file I need to edit in order to make
this routing change permanent?



Thank you



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Re: [asterisk-users] Updated Manual for Asterisk 1.4.x

[Eric \ManxPower\ Wieling]

GNUbie wrote:
 Hello Eric,
 
 On 6/28/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:


 Your best bet is to read UPGRADE.txt in the Asterisk source tree.  It
 should list most of the changes from 1.2.x to 1.4.x

 
 What I just did was install the asterisk-doc package and I now have the
 Asterisk Documentation for v1.4.5.  But the docs doesn't have a detailed
 information about some commands and options used especially for the Dial().
 
 Thanks anyway.

The best source for documentation of dialplan applications is  show 
application X in the Asterisk CLI.

Example:

show application dial

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Re: [asterisk-users] Asterisk + hinting presence + macro

[Eric \ManxPower\ Wieling]

Rob Schall wrote:
 I currently have about 50 polycom 501 phones on my asterisk setup. The 
 dialplan is set to work with mysql (realtime), and all of the extensions 
 for the phones route through the same macro (stdexten). This all works 
 fine until I tried to set up notify status.
 
 On voip-info, they say do something like...
 
 ,hint,SIP/
 ,1,Dial(SIP/)
 blah blah blah
 
 This functionality works fine. But what if you have a macro
 s,hint,SIP/${ARG1}
 s,1,Dial(SIP/${ARG1}
 
 this adds a s hint which obviously doesn't work, instead of a hint for 
  as it should.

Yes.  Put in the correct hint.  There is no reason that 
,hint,SIP/ would not work in a macro.

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[asterisk-users] Shared Extension Appearance

[Mike Ryan]

If SLA supports IP trunks, can shared extension appearance be achieved using
a local SIP trunk in place of an extension?

Basically, I'm trying to allow some stations (Polycom IP 650) to have a
shared extension amongst all of them.  Ideally, I'd like for the LED to show
if that extension is in use, and I'd like for the extension to ring all
stations on that extension when a call comes in.

I have done the latter, which I was able to achieve without using SLA.  In
short, if the extenison is 213, I created SIP extentions for each station
(station1213, station2213, station3213, etc.).  On the stations, I just
change the label to 213, so it looks like it's 213.  In the dial plan, I
modified the stdexten macro to ring all of the stations when a call comes
into 213, and I changed the callerid to show 213 when dialing out.

I'm only missing one thing: I want the line to show busy, or available, and
I'm sure there's probably a work around for this too.

My question is: Can SLA give me the same results?  And if so, does it make
more sense to use SLA to achieve this?  Lastly, if I use SLA, will I also
have the ability to barge and will I be able to park using the hold button?

Thank you,

Mike Ryan
Installation Support Engineer
Percipia, Inc.
858 Morrison Rd.
Gahanna, OH 43230
+1 614-856-1123 (office)
+1 614-579-6055 (cell)
+1 614-751-2018 (fax)
mykryen (skype)
[EMAIL PROTECTED] (yahoo)
[EMAIL PROTECTED] (msn)
[EMAIL PROTECTED] (gtalk)
http://www.percipia.com/


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[asterisk-users] Robo Dialer

[NedTel NedTel]

Hi,

I would like to set up in the Asterisk system (downloaded from Nerdvittles)
a robo-dialer for an outbound call center. Idea is that the dialer should do
predictive dialing and once the call is answered pass it through to the next
free agent. CTI would be a nice to have. ;-)

Anyone who can help me with this?

Thanks,

Vikash
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[asterisk-users] Anyone who can do live video feed to co-host asterisk show next week?

[randulo]

Hi all,

I'm looking for an asterisk user (can be a n00b who knows enough about
asterisk to ask intelligent questions or a brilliant specialist) to
talk about what they do with asterisk. I would like to have a co-host
next week, someone who uses video via the web (it's a Flash
application that can send to the server from a webcam or a connected
firewire or USB camera). Ideally this is someone in the USA or Asia.

The idea is that we would cut to you for questions or comments during
the live broadcast. If you can picture what I'm talking about and
have any interest in this, please write me off-list. It should be fun,
next Friday July 6th at 12:30PM EDT. We'll need to set up an account
and test at least once between now and then.

If you're not interested in live but you have produced any
asteriskrelated video content you'd like to share, please contact me
off list and send URL if you can.

tia,

/r

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[asterisk-users] CDR Log analizer software

[Mark Coccimiglio]

  Hello all,
I'm looking for software for my asterisk logs that will compile the 
information into nice web-based charts and graphs.  Something that works 
similar to webalizer for apache.  I want to be able to spot trends of 
usage, call volume levels, disconnect/failure levels, and basically see 
exactly where my system has been at over the past day/week/month, etc. 
.  I would prefer for the software to work with 1.2 and 1.4 but 1.4 is 
the more important version for me these days.

I have found FOP to be a great tool for a current snapshot of where I'm 
at but I have no indication of where i've been unless I'm watching it (a 
little to busy these days to just watch my pbx).

Your input is greatly appreciated.

Mark C


http://www.psh-inc.com
http://www.alohafreewifi.com
http://www.coccimiglio.net

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Re: [asterisk-users] Asterisk + hinting presence + macro

[Rob Schall]

Eric ManxPower Wieling wrote:

Rob Schall wrote:
  
I currently have about 50 polycom 501 phones on my asterisk setup. The 
dialplan is set to work with mysql (realtime), and all of the extensions 
for the phones route through the same macro (stdexten). This all works 
fine until I tried to set up notify status.


On voip-info, they say do something like...

,hint,SIP/
,1,Dial(SIP/)
blah blah blah

This functionality works fine. But what if you have a macro
s,hint,SIP/${ARG1}
s,1,Dial(SIP/${ARG1}

this adds a s hint which obviously doesn't work, instead of a hint for 
 as it should.



Yes.  Put in the correct hint.  There is no reason that 
,hint,SIP/ would not work in a macro.


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So, if I understand you correctly, my macro would look something vaguely 
like...


[macro-stdexten]
${ARG1},hint,SIP/${ARG1}
s,1,Dial(${ARG1})?

This will work? My understand was that by going into a macro, you were 
going to be using the s extension. I'm not sure how that hint would 
get called if its not inside the s extension.


Rob
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Re: [asterisk-users] network routing

[Ed Nuñez]

This allows me to edit the IP Address of the NIC card, but not edit my IP
routing.

 

Thanks 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ~Russell
Sent: Thursday, June 28, 2007 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] network routing

 

try to edit /etc/sysconfig/network-scripts/ifcfg-eth0  if u have eth0   

if not try ifcfg-eth1 for eth1




On 6/29/07, Ed Nuñez  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
wrote:

I have installed the Asterisk BE B.2.2 image file in a new server.  I need
to make network routing changes.  However in their version of rPath (pound
key) Digium has removed the netconfig command.  I am able to manually add
the route with 

 

Route add default gw xxx.xxx.xxx.xxx however when I reboot I lose my
routing.  Does anyone know which conf file I need to edit in order to make
this routing change permanent?

 

Thank you

 


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[asterisk-users] setup multiple phones for 1 extension

[Ryan Stille]

I'll start by saying I'm a trixbox user, and a new one at that, so 
hopefully you can respond to me on those terms.

I have a user who works from home 1 day a week.  On that day I'd like 
for him to be able to connect with a softphone and be reachable by just 
dialing his extension as we normally would.  I could set him up a new 
extension, then he could forward his phone there on those days.  But I'd 
like an easier, more transparent way to do it.

So I'd like that whenever his softphone is connected (registered is 
the term I believe), that dialing his extension would ring his desk 
phone and his softphone (at the same time).  If the softphone isn't 
connected, then of course just the desk phone would ring.  To do this, 
do I need to setup his extension to actually be a ring group, and give 
him two separate extensions, then put those in the ring group?  Is that 
the best way to do it?  Or could he just log on twice with the same 
extension/secret?

Thanks,
-Ryan



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Re: [asterisk-users] setup multiple phones for 1 extension

[Gustavo Hernandez]

use folow-me
- Original Message - 
From: Ryan Stille [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, June 28, 2007 4:16 PM
Subject: [asterisk-users] setup multiple phones for 1 extension


 I'll start by saying I'm a trixbox user, and a new one at that, so 
 hopefully you can respond to me on those terms.
 
 I have a user who works from home 1 day a week.  On that day I'd like 
 for him to be able to connect with a softphone and be reachable by just 
 dialing his extension as we normally would.  I could set him up a new 
 extension, then he could forward his phone there on those days.  But I'd 
 like an easier, more transparent way to do it.
 
 So I'd like that whenever his softphone is connected (registered is 
 the term I believe), that dialing his extension would ring his desk 
 phone and his softphone (at the same time).  If the softphone isn't 
 connected, then of course just the desk phone would ring.  To do this, 
 do I need to setup his extension to actually be a ring group, and give 
 him two separate extensions, then put those in the ring group?  Is that 
 the best way to do it?  Or could he just log on twice with the same 
 extension/secret?
 
 Thanks,
 -Ryan
 
 
 
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Re: [asterisk-users] network routing

[~Russell]

How many GW you need to add ?  if it is one .. then add

GATEWAY=xxx.xxx.xxx.xxx into /etc/sysconfig/network

thanks
Russell


On 6/29/07, Ed Nuñez [EMAIL PROTECTED] wrote:


 I have installed the Asterisk BE B.2.2 image file in a new server.  I
need to make network routing changes.  However in their version of rPath
(pound key) Digium has removed the netconfig command.  I am able to manually
add the route with



Route add default gw xxx.xxx.xxx.xxx however when I reboot I lose my
routing.  Does anyone know which conf file I need to edit in order to make
this routing change permanent?



Thank you



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Re: [asterisk-users] setup multiple phones for 1 extension

[tracinet]

He can not have the same username/secret.  In trixbox - your ring group idea
is probably best...

On 6/28/07, Ryan Stille [EMAIL PROTECTED] wrote:


I'll start by saying I'm a trixbox user, and a new one at that, so
hopefully you can respond to me on those terms.

I have a user who works from home 1 day a week.  On that day I'd like
for him to be able to connect with a softphone and be reachable by just
dialing his extension as we normally would.  I could set him up a new
extension, then he could forward his phone there on those days.  But I'd
like an easier, more transparent way to do it.

So I'd like that whenever his softphone is connected (registered is
the term I believe), that dialing his extension would ring his desk
phone and his softphone (at the same time).  If the softphone isn't
connected, then of course just the desk phone would ring.  To do this,
do I need to setup his extension to actually be a ring group, and give
him two separate extensions, then put those in the ring group?  Is that
the best way to do it?  Or could he just log on twice with the same
extension/secret?

Thanks,
-Ryan



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Re: [asterisk-users] CDR Log analizer software

[Ivan Cetta]

Hi.
Maybe Asterisk Stat could help you.

http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54

Hope it helps you
Regards
Iván Cetta.

On 6/28/07, Mark Coccimiglio [EMAIL PROTECTED] wrote:
   Hello all,
 I'm looking for software for my asterisk logs that will compile the
 information into nice web-based charts and graphs.  Something that works
 similar to webalizer for apache.  I want to be able to spot trends of
 usage, call volume levels, disconnect/failure levels, and basically see
 exactly where my system has been at over the past day/week/month, etc.
 .  I would prefer for the software to work with 1.2 and 1.4 but 1.4 is
 the more important version for me these days.

 I have found FOP to be a great tool for a current snapshot of where I'm
 at but I have no indication of where i've been unless I'm watching it (a
 little to busy these days to just watch my pbx).

 Your input is greatly appreciated.

 Mark C


 http://www.psh-inc.com
 http://www.alohafreewifi.com
 http://www.coccimiglio.net

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Re: [asterisk-users] network routing

[Watkins, Bradley]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
 Sent: Thursday, June 28, 2007 3:13 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] network routing
 
 This allows me to edit the IP Address of the NIC card, but 
 not edit my IP routing.
 
  

In your instance, you're trying to add a default gateway.

Therefore, in your /etc/sysconfig/network-scripts/ifcfg-ethX file:

GATEWAY=XXX.XXX.XXX.XXX


If you need others, create /etc/sysconfig/network-scripts/route-ethX and use 
this format:


GATEWAY0=1.2.3.4
NETMASK0=255.255.255.0
ADDRESS0=1.2.3.0

GATEWAY1=1.2.3.5
NETMASK1=255.255.255.0
ADDRESS1=1.2.3.5

And so forth.

- Brad

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Re: [asterisk-users] North American voice BRI - Informal survey

[Benny Amorsen]

 SB == Stephen Bosch [EMAIL PROTECTED] writes:

SB Hi, folks: I remain intrigued by the gap in BRI implementation
SB between North America and Europe, and I wanted to get feedback
SB from the list members on the matter. I'm seriously considering
SB making the leap in our office.

BRI is being phased out in some parts of Europe. Try ordering a new
BRI line from Telia in Sweden...


/Benny



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Re: [asterisk-users] RTCP NTP Clock skew

[John Millican]

On Thursday June 28 2007 1:19 pm, Jared Smith wrote:
 On 6/28/07, John Millican [EMAIL PROTECTED] wrote:
  Would i be correct in assuming that if i pull a copy of
  1.4.5 from digium this weekend that this message will go away?

 No... you'd have to pull the latest code from the 1.4 branch using
 Subversion, or wait for 1.4.6 to be released.

 -Jared
Thank you for the info.  I missinterpreted Russell's comments to by that it 
would be in the 1.4 stable, silly me.

Internal RTCP NTP clock skew detected: lsr=2362715969, now=2362741181, 
dlsr=65500 (0:999ms), diff=40288

So what sort of badness will this be causing, if it is not fixed in 1.4, by me 
waiting until 1.6?


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Re: [asterisk-users] setup multiple phones for 1 extension

[randulo]

Just add the softphone to the dial command. If it's not connected
nothing will bad happen
and the regular phone will ring. Whenever the softphione is registered
it will ring as well. If the other phone is a SIP phone, you could use
IAX as the softphone with the same username and password. Otherwise,
you'll need to set up another account as someone else said. I'm not
sure how this all works in trixbox, but you would just need to add the
softphone user account and add the phone to the extension. Make any
sense? The whole thing would take about 2 minutes to set up in
straight asterisk terms.

On 6/28/07, Ryan Stille [EMAIL PROTECTED] wrote:

 I have a user who works from home 1 day a week.  On that day I'd like
 for him to be able to connect with a softphone and be reachable by just
 dialing his extension as we normally would.  I could set him up a new
 extension, then he could forward his phone there on those days.  But I'd
 like an easier, more transparent way to do it.

 So I'd like that whenever his softphone is connected (registered is
 the term I believe), that dialing his extension would ring his desk
 phone and his softphone (at the same time).  If the softphone isn't

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Re: [asterisk-users] network routing

[Ed Nuñez]

Thanks, that worked

 

· I was using GATEWAYDEV=eth1

And that was not working.

 

Thanks again

 

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ~Russell
Sent: Thursday, June 28, 2007 3:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] network routing

 

How many GW you need to add ?  if it is one .. then add 

GATEWAY=xxx.xxx.xxx.xxx into /etc/sysconfig/network

thanks
Russell



On 6/29/07, Ed Nuñez [EMAIL PROTECTED] wrote:

I have installed the Asterisk BE B.2.2 image file in a new server.  I need
to make network routing changes.  However in their version of rPath (pound
key) Digium has removed the netconfig command.  I am able to manually add
the route with 

 

Route add default gw xxx.xxx.xxx.xxx however when I reboot I lose my
routing.  Does anyone know which conf file I need to edit in order to make
this routing change permanent?

 

Thank you

 


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Re: [asterisk-users] North American voice BRI - Informal survey

[Jeremy Mann]

 you would think the telcos would be more interested in selling this to
 small/medium businesses that are not ready for a voice pri but it

Since when to the telcos have the consumer's best interest in mind?  They can 
sell you a PRI at full loop cost with a smaller number of channels in the hopes 
you will add to it, they will then charge you an upgrade fee or some other 
inflated installation cost when in reality it is almost 0 work to reprovision, 
pure profit for them.

ATT is/was doing buyback promotions recently for 5 analog lines + a full Data 
T1 for around $425 total(including loop cost), that's a steal and frankly we 
would have been crazy to request BRI service.

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Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware

[Dan Austin]

Greg wrote:
 So, if you ever use a Cisco SIP Phone with an Asterisk 
 server, it's  not possible to localize menus, soft 
 keys, and so on ?

 Not unless someone wants to add support for it in the SIP
 channel, which I doubt.  I would be more than willing to 
 provide the SIP messages that a CallManager sends to 
 accomplish it though.

Localization with CCM happens when the phone boots.  The
initial TFTP config download (xml or older .cfg) includes
a setting to identify the local, which then TFTP downloaded.
Setting up the initial config file (xml or older .cfg) is
not difficult, but without copies of the localization files
on your TFTP server, it will not help much.

With the localization files, the channel driver can send
the button templates and the phone will display the localized
version of the button(s).

Dan

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Re: [asterisk-users] North American voice BRI - Informal survey

[Stephen Bosch]

Jeremy Mann wrote:
 you would think the telcos would be more interested in selling this
 to small/medium businesses that are not ready for a voice pri but
 it
 
 Since when to the telcos have the consumer's best interest in mind?
 They can sell you a PRI at full loop cost with a smaller number of
 channels in the hopes you will add to it, they will then charge you
 an upgrade fee or some other inflated installation cost when in
 reality it is almost 0 work to reprovision, pure profit for them.

People forget that the PSTN network was very expensive to build in the
first place, which is why we had monopolies; the regulation of the
market amounted to a form of subsidy.

With that regulatory impetus gone, there's little incentive for the
telcos to maintain their last-mile wireline infrastructure. There's
precious little money in it. Under those circumstances, they can be
expected to tweak things to make them worthwhile.

I'm not defending them -- I'm just pointing out that this isn't just a
simple case of the damn telcos. We're living in a world of our own
making. It's the price of having cheap long distance.

 ATT is/was doing buyback promotions recently for 5 analog lines + a
 full Data T1 for around $425 total(including loop cost), that's a
 steal and frankly we would have been crazy to request BRI service.

I cannot get partial PRI with fewer than 10 channels around here, so
there's really *no* choice.

-Stephen-

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Re: [asterisk-users] setup multiple phones for 1 extension

[Ryan Stille]

I installed the follow-me module and tried it out, it works great.  I am 
just continually amazed at what asterisk can do.

Another question - I'd like one of the extensions to ring out to a cell 
phone.  I may have the users press '9' or maybe tell them to use 
extension 900 or something, not sure yet.  Whats the best way to forward 
a call out to an external number?

I was thinking I could set it up as an extension, then use follow me, 
tell it to have 0 initial wait time, then in the extension list use 
1234567890#  - the help says suffixing it with a pound sign will route 
the call externally.

Something else I've thought about was playing a message to the user 
asking them if they are sure they want to connect, and if so press 1 - 
then I'll forward out the call. 

-Ryan

Gustavo Hernandez wrote:
 use folow-me
 - Original Message - 
 From: Ryan Stille [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Sent: Thursday, June 28, 2007 4:16 PM
 Subject: [asterisk-users] setup multiple phones for 1 extension


   
 I'll start by saying I'm a trixbox user, and a new one at that, so 
 hopefully you can respond to me on those terms.

 I have a user who works from home 1 day a week.  On that day I'd like 
 for him to be able to connect with a softphone and be reachable by just 
 dialing his extension as we normally would.  I could set him up a new 
 extension, then he could forward his phone there on those days.  But I'd 
 like an easier, more transparent way to do it.

 So I'd like that whenever his softphone is connected (registered is 
 the term I believe), that dialing his extension would ring his desk 
 phone and his softphone (at the same time).  If the softphone isn't 
 connected, then of course just the desk phone would ring.  To do this, 
 do I need to setup his extension to actually be a ring group, and give 
 him two separate extensions, then put those in the ring group?  Is that 
 the best way to do it?  Or could he just log on twice with the same 
 extension/secret?

 Thanks,
 -Ryan



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Re: [asterisk-users] North American voice BRI - Informal survey

[Jon Pounder]

Quoting Stephen Bosch [EMAIL PROTECTED]:

 Jeremy Mann wrote:
 you would think the telcos would be more interested in selling this
 to small/medium businesses that are not ready for a voice pri but
 it

 Since when to the telcos have the consumer's best interest in mind?
 They can sell you a PRI at full loop cost with a smaller number of
 channels in the hopes you will add to it, they will then charge you
 an upgrade fee or some other inflated installation cost when in
 reality it is almost 0 work to reprovision, pure profit for them.

 People forget that the PSTN network was very expensive to build in the
 first place, which is why we had monopolies; the regulation of the
 market amounted to a form of subsidy.

the copper plant was expensive to build yes, which is why not offering  
bri as a mid level service does not make sense - 2x analog lines take  
2pr of copper which taxes the infrastructure more than one bri on one  
pair with the same roughly profit to them as the 2pr solution. yes a  
pri would put even less load on the pairs available, but most  
companies in the 5-50 employee range are not ready for that sort of  
expense when the breakeven point is near the analog lines of the full  
pipe.

physical plant wise what would make sense is just install a  
pri/channelized t1 for anyone with more than a couple lines, then as  
they add and delete lines no physical install has to be done at all,  
channels are just added or dropped from the circuit thats already  
there. its the truck rolls that cost the money, not the hardware on  
the ends at all, and when the pairs run out, then you need a remote  
switch, fibre, pairgain boxes, remote channel banks, etc., etc., to  
continue to offer service in the saturated area.

so go figure ? its a win situation for them from day 1 as far as  
profit per pair, provisioning, and conserving pairs in saturated areas  
to avoid build costs, yet it doesn't happen routinely and its like a  
fight to even make it happen in a special case.











 With that regulatory impetus gone, there's little incentive for the
 telcos to maintain their last-mile wireline infrastructure. There's
 precious little money in it. Under those circumstances, they can be
 expected to tweak things to make them worthwhile.

 I'm not defending them -- I'm just pointing out that this isn't just a
 simple case of the damn telcos. We're living in a world of our own
 making. It's the price of having cheap long distance.

 ATT is/was doing buyback promotions recently for 5 analog lines + a
 full Data T1 for around $425 total(including loop cost), that's a
 steal and frankly we would have been crazy to request BRI service.

 I cannot get partial PRI with fewer than 10 channels around here, so
 there's really *no* choice.

 -Stephen-

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Jon Pounder

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Re: [asterisk-users] North American voice BRI - Informal survey

[Tom]

At 02:37 PM 6/28/2007, you wrote:
  SB == Stephen Bosch [EMAIL PROTECTED] writes:

SB Hi, folks: I remain intrigued by the gap in BRI implementation
SB between North America and Europe, and I wanted to get feedback
SB from the list members on the matter. I'm seriously considering
SB making the leap in our office.

BRI is being phased out in some parts of Europe. Try ordering a new
BRI line from Telia in Sweden...


/Benny


As an ISP who had about 100 BRIs 7 years ago I think I know why.  A 
BRI costs the Telco the same or more to provision than a T1.  They 
both use a good quality copper pair and both need repeaters over a 
certain distance.

We were paying $50 a month for a BRI and $250 to $500 a month for a 
T1 loop depending on mileage 7 years ago.  Newer technologies killed 
our BRI data business.

I think BRIs cost the telco too much in resources to deploy and that 
is why T1 prices have dropped and BRIs have disappeared in favor of 
ADSL for data (which doesn't take any copper since it rides existing 
POTS) and PRIs for voice (which rides a T1).

Tom


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Re: [asterisk-users] network routing

[Joseph Bajin]

or in the same file you can just do a

X.X.X.X via Y.Y.Y.Y

Each new one on a seperate line.

On 6/28/07, Watkins, Bradley [EMAIL PROTECTED] wrote:


  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
  Sent: Thursday, June 28, 2007 3:13 PM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: Re: [asterisk-users] network routing
 
  This allows me to edit the IP Address of the NIC card, but
  not edit my IP routing.
 
 

 In your instance, you're trying to add a default gateway.

 Therefore, in your /etc/sysconfig/network-scripts/ifcfg-ethX file:

 GATEWAY=XXX.XXX.XXX.XXX


 If you need others, create /etc/sysconfig/network-scripts/route-ethX and use 
 this format:


 GATEWAY0=1.2.3.4
 NETMASK0=255.255.255.0
 ADDRESS0=1.2.3.0

 GATEWAY1=1.2.3.5
 NETMASK1=255.255.255.0
 ADDRESS1=1.2.3.5

 And so forth.

 - Brad

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Re: [asterisk-users] Agent Channel SIP transfer

[Marlon Dutra]

On 11/22/06, Xue Liangliang [EMAIL PROTECTED] wrote:
 Hi, we are using asterisk 1.2.13. When callbacklogin agent transfer
 call using SIP phone's transfer feature, he is always in busy status
 and cannot answer any more incoming call from queue until the
 transferee hang up the call.

I'm experiencing the same problem here with Asterisk 1.4.5.

Is there a solution for that?

-- 
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Propus
GnuPG ID: 0x3E2060AC pgp.mit.edu
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http://hackers.propus.com.br/~marlon/

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Re: [asterisk-users] Customized Ring Tone

[Dimitri Volski]

You can use Queues.  Put them in a queue and let them listen to music on 
hold.

Cheers,
Dimitri


GNUbie wrote:
 Hello all,

 I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the
 Digium's Dev Kit that comes with 1 FXO and 1 FXS.  How do I configure my
 home PBX in such a way that whenever someone calls on my trunkline (PSTN)
 number, he/she will hear a customized ring tone, probably playing an MP3
 file, instead of a boring standard ring tone while the extension 
 number that
 is forwarded the call is still ringing?  My current
 /etc/asterisk/extensions.conf file looks like this:

 [general]
 static=yes
 writeprotect=no
 autofallthrough=yes
 clearglobalvars=no

 [pstn]
 exten = s,1,NoOp(Caller ID is ${CALLERID(num)})
 exten = s,2,Dial(Zap/1,15,g2)
 exten = s,n,Congestion

 [local]
 ignorepat = 9
 exten = _9.,1,Dial(Zap/g1/${EXTEN:1})
 exten = _9.,n,Congestion
 exten = 11,1,Dial(Zap/1,20,rt)

 Thank you in advance.

 

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Re: [asterisk-users] Modification of Caller ID based on context

[arkda]

Nice solution Eric, thanks. Very elegant.

On 6/27/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:


Matthew Brothers wrote:
 Hi,

 I have been looking for an example of accomplishing this, but
 I've been unable to locate something similar to what I'm trying
 to do.

 Here's the scenario:

 Users caller ID is set to their internal extension (200-250).
 This is set in sip.conf for each user. Each user has a local DID
 as well (hosted through Vitelity, for example (555)111-). The
  problem is that this extension was being passed to the outside
 world. I currently have a SetCallerID command changing the
 CallerID to our main office number, but some users want their DID
  sent, not the general number.

 The problem is that if their caller ID is set to their DID, when
 users hit redial on their phones internally they dial out and
 back in. I corrected this by putting each DID in extensions.conf
 under their three digit extension, but that seems a bit like a
 kludge obviously.

 I'm looking for a method of sending the internal three digit
 extension only when a user is dialing another user internally,
 otherwise it will send their DID. Is their a method to do this in
  the dial plan? Anyone have an example of how to accomplish this?


 Thanks in advance.


 Mike,

 I have a similar setup (I even use Vitel) and the easiest and
 cleanest method that I have found to accomplish this is with the
 AstDB. You can simply create a cross-reference of DIDs and Internal
 extensions similar to extdid/200 = 555111 ... extdid/250 =
 5551112272 in the AstDB. Then you can change your outgoing dialplan
 to change the caller id based upon this cross reference. Example:


 exten = NXXNXX,n,Set(outgoingCID=MAINNUMBER)

 exten = NXXNXX,n,
 GotoIf($[ ${DB_EXISTS(extdid/${CALLERID(num)})} = 0 ]?makecall)

 exten = NXXNXX,n,
 Set(outgoingCID=${DB(extdid/${CALLERID(num)})})

 exten = NXXNXX,n(makecall),Set(CALLERID(num)=${outgoingCID})

 ...

 You could even simplify your incoming context by cross-referencing
 in the other direction. That is didext/555111 = 200 ...
 didext/5551112272 = 250.

 exten = NXXNXX,n,
 Goto(internal-extensions,${DB(didext/${EXTEN})},1)

 OR you could do something similar with LOCAL channels or with a Dial
 command.

Here is my solution.  I've stripped out most of the unimportant stuff.

Because our carrier charges for PICs on a per-DID basis, we set the
Caller*ID number for long distance calls to be the main number,
regardless of what the person's DID is.   It also allows use of more
than one main number, depending on the device making the call.

The macro-dial-result is not important for this.  It is a macro we use
to figure out what happened to the call based on HANGUPCAUSE and what,
if any tone or message to send the caller, as well as decide if the call
failed and should be sent out a different route.

In sip.conf set up the device like this:

[0004f201e570-a]
callerid=Room, Computer 3726
setvar=DID=9852463726
setvar=BTN=9858982022
accountcode=3726
type=friend
host=dynamic
secret=S
context=toll-access

My extensions.conf looks like this:

[toll-access]
;
; 9-1-nxx-nxx-
exten = _91NXXNXX,1,Set(USE_BTN=yes)
exten = _91NXXNXX,n,Gosub(outgoing-call-fixup,${EXTEN},1)
exten = _91NXXNXX,n,Dial(${PSTN}/${EXTEN:1},,g)
exten = _91NXXNXX,n,Macro(dial-result,SIP/[EMAIL PROTECTED])
;
; 9-1-985-nxx-
exten = _91985NXX,1,Gosub(outgoing-call-fixup,${EXTEN},1)
exten = _91985NXX,n,Dial(${PSTN}/${EXTEN:1},,g)
exten = _91985NXX,n,Macro(dial-result,SIP/[EMAIL PROTECTED])

[outgoing-call-fixup]
;
exten = _X.,1,GotoIf($[${LEN(${CALLERID(num)})} != 10]?check-btn)
exten = _X.,n,Return
exten = _X.,n(check-btn),GotoIf($[${USE_BTN} = yes]?set-btn)
exten = _X.,n,Set(CALLERID(num)=${DID})
exten = _X.,n,Return
exten = _X.,n(set-btn),Set(CALLERID(num)=${BTN})
exten = _X.,n,Return

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[asterisk-users] SPA-2100 Distinctive Ring

[Matt Putnam]

I have been looking into how to setup distinctive ringing on a SPA-2100. So
far the only thing i have been able to find is how to define a distinctive
ring in the spa config. What i cannot figure out is what SIP message i need
to be sending to it in order for it use the ring. I did find out how to add
the sip message for distinctive ring i just dont know what variable needs to
be passed in order for it to work.
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Re: [asterisk-users] Customized Ring Tone

[GNUbie]

Hello Dimitri,

On 6/29/07, Dimitri Volski [EMAIL PROTECTED] wrote:


You can use Queues.  Put them in a queue and let them listen to music on
hold.



How do you do this based on my original /etc/asterisk/extensions.conf that I
have on my home PBX?  I just want that the PSTN caller will hear a music
instead of a ring tone while the callee rings.

Thanks.
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Re: [asterisk-users] SPA-2100 Distinctive Ring

[Luki]

 I did find out how to add the sip message for distinctive ring
 i just dont know what variable needs to be passed in
 order for it to work.

Try: SetVar(_ALERT_INFO=Bellcore-r2);

etc.

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Re: [asterisk-users] Linking Asterisk with another SIP PBX (or SIP Softswitch)

[Noah Miller]

Hi Bilal -

 If I need to do a trunk between Asterisk and another
 SIP softswitch (so Asterisk will send a SIP calls to
 that softswitch), then I have to configure this on the
 sip.conf file

Yes.


 And is it the same
 when I configure iax trunk?

Not exactly the same, but very close.  Here's a page on how to connect
Asterisk to Cisco Call Manager using SIP:

http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration


 Should I determine the context in this case for this
 SIP trunk?

Yes.


- Noah

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Re: [asterisk-users] Shared Extension Appearance

[Russell Bryant]

Mike Ryan wrote:
 My question is: Can SLA give me the same results?  And if so, does it make
 more sense to use SLA to achieve this?  Lastly, if I use SLA, will I also
 have the ability to barge and will I be able to park using the hold button?

The SLA code that is in Asterisk now will not completely support what I would 
consider real shared extensions.  It is intended to map some trunk (either real 
FXO trunk or IP trunk) to buttons on phones.  When any phone is making a call 
using that trunk, that line appearance on all of the other phones shows that it 
is in use.  It allows for barging into calls, as well as parking the call by 
simply putting it on hold.  Once a trunk is on hold, it can be retrieved from 
hold from any of the other phones (stations) that have this line appearance.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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Re: [asterisk-users] Asterisk and IPv6

[Russell Bryant]

Bent Bagger wrote:
 When will these additions make their way into the Asterisk mainstream

It has not yet been merged into the main development tree, but I'm sure it will 
be before Asterisk 1.6 is released.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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Re: [asterisk-users] Query

[Deepak Naidu]

I am not sure what exactly you wish to achieve.  Just a basic SIP--to--SIP call 
or ?
   
  I am not much into the configs, but ya I can tell you that you can try using 
FreePBX or Trixbox kind of setup to write ur Asterisk config file, rather then 
u editing them, as it has macros, context etc... which is too high to me.  But 
the browser interface help a lot understanding the config files later once 
configured via FreePBX.
   
  FreePBX -- Its a tool(software which is wrapper over asterisk which gives a 
web based interface to manage  configure ur asterisk configuration files with 
easy understanding.
   
  tixbox-- Its a kind of Asterisk solution which is combination of 
asterisk+freepbx+linux+crm tools etc.. for quick Asterisk deployment.
   
  I am not sure whether u know all these if yes, hen excuse me.. but ur mail 
sounded u might need this info needed.

[EMAIL PROTECTED] wrote:
  Hi,
I am trying to establish call through sip phone between two PC connected to 
linux box on which asterisk server is running

1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP Adress : 192.168.1.53

Now, I am tying to dial from 1st PC to 2nd PC

I am trying to dial from 1st PC to 2nd PC through asterisk server
The problem is 1st PC is calling directly to 2nd PC not through asterisk server

I am doing the following additions in configuration files

1) sip.conf

[general]
context=sip
bindport=5060 
bindaddr=0.0.0.0 

[phone1]
type=friend
host=192.168.1.149
port=5060
nat=yes
dtmfmode=rfc2833
context=sip

[phone2]
type=friend
host=192.168.1.53
port=5060
nat=yes
dtmfmode=rfc2833
context=sip

2) extensions.conf
exten = 11,1,Dial(SIP/phone2,20,tr)

Now, I am calling from sip phone1 by name [EMAIL PROTECTED]
It is not being called through asterisk server running on linux m/c. It is 
calling directly. As, I am running sip debub but no packet dumping is taking 
place. Can anybody will tell me the error I am doing.
Thanx and regards
sanchal
















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Linux your Life, Don't Window it [[]] 

   { All for the best }



   
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Re: [asterisk-users] Asterisk + hinting presence + macro

[Eric \ManxPower\ Wieling]

Rob Schall wrote:
 Eric ManxPower Wieling wrote:
 Rob Schall wrote:
  
 I currently have about 50 polycom 501 phones on my asterisk setup. 
 The dialplan is set to work with mysql (realtime), and all of the 
 extensions for the phones route through the same macro (stdexten). 
 This all works fine until I tried to set up notify status.

 On voip-info, they say do something like...

 ,hint,SIP/
 ,1,Dial(SIP/)
 blah blah blah

 This functionality works fine. But what if you have a macro
 s,hint,SIP/${ARG1}
 s,1,Dial(SIP/${ARG1}

 this adds a s hint which obviously doesn't work, instead of a hint 
 for  as it should.
 

 Yes.  Put in the correct hint.  There is no reason that 
 ,hint,SIP/ would not work in a macro.

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 So, if I understand you correctly, my macro would look something vaguely 
 like...
 
 [macro-stdexten]
 ${ARG1},hint,SIP/${ARG1}
 s,1,Dial(${ARG1})?
 
 This will work? My understand was that by going into a macro, you were 
 going to be using the s extension. I'm not sure how that hint would 
 get called if its not inside the s extension.

I have no idea, but as I understand it, Hints are separate from extensions.

I guess you could do something like:

[macro-stdexten]
exten = s,1,Goto(${MACRO_EXTEN},1)

exten = _,hint,SIP/${ARG1}
exten = _,1,Dial(${ARG1})

I do this sort of thing in many of my macros that Dial somewhere.  I 
seem to remember something about hints not working for pattern matching. 
or working weirdly.

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Re: [asterisk-users] FAX over T1

[Andres Paglayan]

On Jun 22, 2007, at 3:43 PM, Joe acquisto wrote:

 I have an existing Hylafax system using a mainpine 4 port board, 4  
 POTS lines.

 Have a recently installed Asterisk system, with a dedicated T1  
 line.  (Well, it's really a fonality system).

 What would I need to do, or where is the reading material, for what  
 I need to do, to convert the Hylafax server to use the T1 line?
 Reliably.  Preferably to use DID's as well.

 The current FAX works fine, but there is some desire to get rid of  
 the analog lines.

 Could one add some sort of device in the Asterisk server, to act as  
 FAX extensions, keeping the mainpine on the hylafax?  Like a  
 TDM400p with FSX modules?

 I'm just saying, ya know?  I suppose I have to ask fonality, since  
 it's their box?

 joe a.



did you fix this yet?
I had the same problem,
and worked it out,
contact me off list if you want the how to
(or at least one of the how tos)


Andres Paglayan

--Harmony is more important than being right
Bapak



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Re: [asterisk-users] Asterisk + hinting presence + macro

[Jaswinder Singh]

Sorry i didnt read your mail properly . I thought your problem is with
cdr's. Here's link to cdr problem  :)

http://lists.digium.com/pipermail/asterisk-dev/2007-June/028085.html

see the next message for patch .

On 29/06/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:


Rob Schall wrote:
 Eric ManxPower Wieling wrote:
 Rob Schall wrote:

 I currently have about 50 polycom 501 phones on my asterisk setup.
 The dialplan is set to work with mysql (realtime), and all of the
 extensions for the phones route through the same macro (stdexten).
 This all works fine until I tried to set up notify status.

 On voip-info, they say do something like...

 ,hint,SIP/
 ,1,Dial(SIP/)
 blah blah blah

 This functionality works fine. But what if you have a macro
 s,hint,SIP/${ARG1}
 s,1,Dial(SIP/${ARG1}

 this adds a s hint which obviously doesn't work, instead of a hint
 for  as it should.


 Yes.  Put in the correct hint.  There is no reason that
 ,hint,SIP/ would not work in a macro.

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 So, if I understand you correctly, my macro would look something vaguely
 like...

 [macro-stdexten]
 ${ARG1},hint,SIP/${ARG1}
 s,1,Dial(${ARG1})?

 This will work? My understand was that by going into a macro, you were
 going to be using the s extension. I'm not sure how that hint would
 get called if its not inside the s extension.

I have no idea, but as I understand it, Hints are separate from
extensions.

I guess you could do something like:

[macro-stdexten]
exten = s,1,Goto(${MACRO_EXTEN},1)

exten = _,hint,SIP/${ARG1}
exten = _,1,Dial(${ARG1})

I do this sort of thing in many of my macros that Dial somewhere.  I
seem to remember something about hints not working for pattern matching.
or working weirdly.

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