Re: [Asterisk-Users] three way calling and cisco ata 186

2003-07-08 Thread Pavel Zheltouhov
Thomas Dingermann wrote:
Ok, if this is not working with sip or h.323, maybe it does with mgcp ?
I tried to get ATA and Asterisk working with MGCP, but nothing worked!
Any Howtos available about MGCP/ATA186/Asterisk?
I just try two ATA with asterisk with that configuration files :

;
; MGCP Configuration for Asterisk
;
[general]
port = 2727
bindaddr = 0.0.0.0
allow=ulaw
inbanddtmf=yes
transfer = yes
threewaycalling=yes
[10.0.1.19]
transfer = yes
threewaycalling=yes
host = 10.0.1.19
context = default
line => aaln/1
transfer = 1
line => aaln/2
transfer = 1
line => *
[10.0.1.20]
transfer = yes
threewaycalling=yes
host = 10.0.1.20
context = default
line => aaln/1
transfer = 1
line => aaln/2
transfer = 1
line => *
and extensions.conf

---
exten => 31,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr)
exten => 32,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr)
exten => 33,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr)
exten => 34,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr)
-
Ordinary tasks works good. Call transfer with '#' key work too.
But three way calling not work with stange error :
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
dial to 33
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '3'
-- Executing Dial("MGCP/aaln/[EMAIL PROTECTED]", 
"MGCP/aaln/[EMAIL PROTECTED]|20|tr") in new stack
-- MGCP mgcp_request(aaln/[EMAIL PROTECTED])
-- MGCP cw: 0, dnd: 0, so: 0, sno: 0
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
-- Called aaln/[EMAIL PROTECTED]
-- MGCP/aaln/[EMAIL PROTECTED] is ringing

answer
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP/aaln/[EMAIL PROTECTED] answered MGCP/aaln/[EMAIL PROTECTED]
-- MGCP mgcp_answer(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED]
-- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and 
MGCP/aaln/[EMAIL PROTECTED]

Talking now
Attempt call person 3 :
 hookflash :
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf'
-- Swapping 1 for 0 on aaln/[EMAIL PROTECTED]
-- MGCP Muting 1 on aaln/[EMAIL PROTECTED]
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
now trying dial to other phone ( 600 - echo test )

-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '6'

atfter this, person1 hear 'fastbusy', short beeps !

And other output of asterisk:

-- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and 
MGCP/aaln/[EMAIL PROTECTED]
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0'
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0'
-- Executing Playback("MGCP/aaln/[EMAIL PROTECTED]", "demo-echotest") 
in new stack
-- MGCP mgcp_answer(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED]
-- Playing 'demo-echotest'
-- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and 
MGCP/aaln/[EMAIL PROTECTED]
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf'
-- Swapping 0 for 1 on aaln/[EMAIL PROTECTED]
-- We didn't make one of the calls FLIPFLOP 0 and 1 on aaln/[EMAIL PROTECTED]
-- MGCP Muting 0 on aaln/[EMAIL PROTECTED]
-- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and 
MGCP/aaln/[EMAIL PROTECTED]
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu'
NOTICE[20501]: File chan_mgcp.c, Line 762 (mgcp_fixup): 
mgcp_fixup(MGCP/aaln/[EMAIL PROTECTED], MGCP/aaln/[EMAIL PROTECTED])
WARNING[20501]: File chan_mgcp.c, Line 764 (mgcp_fixup): old channel 
wasn't 0x81065a8 but was (nil)
WARNING[20501]: File channel.c, Line 1847 (ast_do_masquerade): Fixup 
failed on channel MGCP/aaln/[EMAIL PROTECTED], strange things may happen.
NOTICE[20501]: File chan_mgcp.c, Line 762 (mgcp_fixup): 
mgcp_fixup(MGCP/aaln/[EMAIL PROTECTED], MGCP/aaln/[EMAIL PROTECTED])
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu'
  == Spawn extension (default, 600, 1) exited non-zero on 
'MGCP/aaln/[EMAIL PROTECTED]'

--

Any ideas ?

--
Pavel Zheltouhov, Comlink ISP, Voronezh, Russia
phone/fax +7(0732) 727172, http://www.comlink.ru
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Re: [Asterisk-Users] three way calling and cisco ata 186

2003-07-08 Thread Thomas Dingermann
Pavel Zheltouhov wrote:

Ok, if this is not working with sip or h.323, maybe it does with mgcp ?
I tried to get ATA and Asterisk working with MGCP, but nothing worked!
Any Howtos available about MGCP/ATA186/Asterisk?
Thomas

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Re: [Asterisk-Users] three way calling and cisco ata 186

2003-07-08 Thread Pavel Zheltouhov
Thomas Dingermann wrote:
I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and
asterisk  as pbx. I need feature called as 'three way calling' or 
'transfer with consultation'. Registering,calling and 'blind transfer' 
work fine.
Same here - and if you hang up, the call is not transferred...
Something goes wrong when using the Hook-Flash with ATA/Asterisk.
Any solutions out there?
Ok, if this is not working with sip or h.323, maybe it does with mgcp ?

--
Pavel Zheltouhov, Comlink ISP, Voronezh, Russia
phone/fax +7(0732) 727172, http://www.comlink.ru
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Re: [Asterisk-Users] three way calling and cisco ata 186

2003-07-07 Thread Thomas Dingermann
Pavel Zheltouhov wrote:
I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and
asterisk  as pbx. I need feature called as 'three way calling' or 
'transfer with consultation'. Registering,calling and 'blind transfer' 
work fine.

Same here
Is this feature provided by sip clients or by asterisk itself ?
What I have to configure in ATA   and what keys I have to press
on my phones ?
Three way calling is enabled by setting  bits 28-29 to 10
 ( Select the Cisco VG248 Style for mid-call services.
 as described at 
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a00800c4d1f.html#42433 

But it seems not works,I always get conference call with 3 persons.

Same here - and if you hang up, the call is not transferred...
Something goes wrong when using the Hook-Flash with ATA/Asterisk.
Any solutions out there?

Thomas

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[Asterisk-Users] three way calling and cisco ata 186

2003-07-07 Thread Pavel Zheltouhov
I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and
asterisk  as pbx. I need feature called as 'three way calling' or 
'transfer with consultation'. Registering,calling and 'blind transfer' 
work fine.

Is this feature provided by sip clients or by asterisk itself ?
What I have to configure in ATA   and what keys I have to press
on my phones ?
Three way calling is enabled by setting  bits 28-29 to 10
 ( Select the Cisco VG248 Style for mid-call services.
 as described at 
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a00800c4d1f.html#42433
But it seems not works,I always get conference call with 3 persons.

--
Pavel Zheltouhov
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