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Hi Marios, I found the problem, the t38 patch. I
was testing T38 to send and receive fax, there is a patch to sip.c for
that.
Now I recompiled asterisk without
the patch and the transfer problem is solved.
Thanks for you help
Diego
----- Original Message -----
Sent: Wednesday, March 08, 2006 5:54
PM
Subject: RE: [Asterisk-bsd] Zap on hold
problem
I just tried it to see what happens.
No problems for calls from Zap->eyeBeam then pressed
line 2 dialed a sip phone then press xfer press line 1 and the 2 were
connected.
Is the eyeBeam behind a NAT ?
Is asterisk and eyeBeam on the same
network?
Thanks Marios, I made you told me, and it
works fine, but we need a supervised transfer.
It seems as asterisk ignores the invites from
UA when I press line 2 button. The eyebeam does not receive response form
asterisk, and resend the invite three times. ( I see that on diagnostic
log of eyebeam)
This is the invite from eyebeam:
SENDING TO: {ip of asterisk} :5060
INVITE
sip:[EMAIL PROTECTED] of asterisk} SIP/2.0
To: "asterisk"<sip:[EMAIL PROTECTED]
of asterisk}>;tag=as63cf8d5d
From: <sip:[EMAIL PROTECTED] of
eyebeam}:6199>;tag=b10c4f30
Via: SIP/2.0/UDP {ip of
eyebeam}:6199;branch=z9hG4bK-d87543-252324346-1--d87543-;rport
Call-ID:
[EMAIL PROTECTED] of asterisk}
CSeq: 2 INVITE
Contact:
<sip:[EMAIL PROTECTED] of eyebeam}:6199>
Max-Forwards: 70
Allow: INVITE,
ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 3004t
stamp 16741
Content-Length: 273
v=0
o=- 28646833 28659668 IN IP4 {ip of
eyebeam}
s=eyeBeam
c=IN IP4 0.0.0.0
t=0 0
m=audio 9296 RTP/AVP 0
8 101
a=alt:1 1 : 9CAD96D3 7C38BE5D {ip of eyebeam} 9296
a=fmtp:101
0-15
a=rtpmap:101 telephone-event/8000
a=sendonly
Does asterisk know that this invite is for
him? the packet is send to [EMAIL PROTECTED], Is
necesary define "asterisk" name on /etc/hosts? My host name is
ip-pbx.
Diego
----- Original Message -----
Sent: Tuesday, March 07, 2006 4:37
PM
Subject: RE: [Asterisk-bsd] Zap on hold
problem
This is strange.
So you press line 1 again on eyeBeam and it
doesn't get you back to the first call?
Hmm.
Let's try then the
Asterisk transfer instead or the eybeam.
In features.conf change
;blindxfer => #1
To
blindxfer => #
In *CLI> reload
res_features.so
Make a call to Zap->eyeBeam
Answer eyeBeam and
press #
You should hear "Transferring"
Enter another extension once
successful transfer eyeBeam will hangup
If this works then there is no
problem with asterisk and Zap.
Usually on my eyeBeam I press line 2
enter a number (extension or a PSTN number) once the other extension answers
then I press xfer
and the two are
connected.
-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego
Valencia
Sent: Tuesday, March 07, 2006 1:15 PM
To: Asterisk on BSD
discussion
Subject: Re: [Asterisk-bsd] Zap on hold problem
Hi
Marios, I don't have problem transfering sips, I only have problem when
the call is coming form zap channel. There is a setting for zapata
transfers? Theses are my conf:
features.conf
;
; Sample
Parking configuration
;
[general]
parkext =>
700
; What ext. to dial to park
parkpos =>
701-720
; What extensions to park calls on
context =>
parkedcalls ; Which
context parked calls are in
;parkingtime =>
45
; Number of seconds a call can be parked
for
; (default is 45 seconds)
;transferdigittimeout =>
3 ; Number of seconds to wait between digits
when transfering a call
;courtesytone =
beep ;
Sound file to play to the parked
caller
; when someone dials a parked call
;xfersound =
beep
; to indicate an attended transfer is
complete
;xferfailsound =
beeperr ; to indicate a failed
transfer
;adsipark =
yes
; if you want ADSI parking announcements
;findslot =>
next
; Continue to the 'next' parking space.
Defaults to 'first'
available
pickupexten = 8
; Configure the pickup extension. Default is
*8
;featuredigittimeout = 500 ; Max time
(ms) between digits
for
; feature activation. Default is
500
[featuremap]
;blindxfer =>
#1
; Blind transfer
;disconnect =>
*0
; Disconnect
;automon =>
*1
; One Touch Record
;atxfer =>
*2
; Attended transfer
[applicationmap]
;testfeature =>
#9,callee,Playback,tt-monkeys ;Play tt-monkeys
to
sip.conf
[233]
canreinvite=no
username=233
type=friend
context=nacionales
secret=secret233
;subscribecontext=trunklocal
language=es
host=dynamic
[EMAIL PROTECTED],233
disallow=all
allow=g729
allow=ulaw
allow=alaw
[240]
canreinvite=no
username=240
type=friend
context=nacionales
secret=secret240
;subscribecontext=trunklocal
language=es
host=dynamic
[EMAIL PROTECTED],233
disallow=all
allow=g729
allow=ulaw
allow=alaw
extensions.conf:
[macro-stdexten];
;
;
Standard extension macro:
; ${ARG1} - Extension (we
could have used ${MACRO_EXTEN} here as well
; ${ARG2} -
Device(s) to ring
;
exten =>
s,1,Dial(${ARG2},30,t)
; Ring the
interface, 20 seconds maximum
exten =>
s,2,Goto(s-${DIALSTATUS},1)
; Jump based
on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten =>
s-NOANSWER,1,Dial(SIP/1222,30,)
; retorana a
la consola
exten => s-NOANSWER,2,Hangup
;exten
=>
s-BUSY,1,MusicOnHold(ringbusy)
; If busy, send to
voicemail w/ busy announce
exten =>
s-BUSY,1,Hangup
exten =>
_s-.,1,Goto(s-NOANSWER,1)
; Treat
anything else as no answer
[incomingzap]
include
=> internos
exten =>
s,1,Wait,1
; Wait a second, just for fun
;exten =>
s,n,Set(SIP_CODEC=ulaw)
exten =>
s,2,Answer
; Answer the line
exten =>
s,3,Set(TIMEOUT(digit)=5) ; Set Digit Timeout
to 5 seconds
exten => s,4,Set(TIMEOUT(response)=3) ; Set
Response Timeout to 10 seconds
exten =>
s,5,Set(LANGUAGE()=es) ; Set
language to french
exten => s,6(restart),BackGround(welcome) ; Play a
congratulatory message
exten =>
s,7,WaitExten ; Wait
for an extension to be dialed.
exten =>
s,8,Dial(SIP/232,30
zapata.conf
[channels]
faxdetect=incoming
hanguponpolarityswitch=yes
busydetect=yes
busycount=4
immediate
=> no
transfer => yes
cancallforward =>
yes
threewaycalling => yes
callreturn =>
yes
usecallerid=yes
hidecallerid=no
group => 1
context =>
incomingzap
signalling => fxs_ks
amaflags =>
documentation
echocancel=yes
;Cancela el echo producido por las lineas
análogas
echocancelwhenbridged=yes
echotraining=yes
channel
=> 1-2
------------------------------
Call flow:
--
Starting simple switch on 'Zap/2-1'
Mar 5 13:46:41 NOTICE[3477]:
chan_zap.c:6063 ss_thread: Got event 2
(Ring/Answered)...
-- Executing Wait("Zap/2-1", "1") in
new stack
-- Executing Answer("Zap/2-1", "") in new
stack
-- Executing Set("Zap/2-1", "TIMEOUT(digit)=5") in new
stack
-- Digit timeout set to 5
-- Executing
Set("Zap/2-1", "TIMEOUT(response)=3") in new stack
--
Response timeout set to 3
-- Executing Set("Zap/2-1",
"LANGUAGE()=es") in new stack
-- Executing
BackGround("Zap/2-1", "welcome") in new stack
-- Playing
'welcome' (language 'es')
== CDR updated on Zap/2-1
-- Executing Macro("Zap/2-1", "stdexten|233|SIP/233") in new stack (233
is eyebeam)
-- Executing Dial("Zap/2-1", "SIP/233|30|t")
in new stack
-- Called 233
-- SIP/233-7aa8
is ringing
-- SIP/233-7aa8 answered Zap/2-1
-------------> I press "line 2" button
on eyebeam to call to other
extension
-- Started music on hold, class 'default', on
Zap/2-1 ---------> MOH on
ZAP
At this point the caller (PSTN)
is on MOH, but I can't return to call 1 to
transfer it. After a few
minutes the eyebeam says "Failed to place call on
hold"
Thanks
Diego
----- Original Message -----
From: "Marios Andreou" <[EMAIL PROTECTED]>
To:
"'Asterisk on BSD discussion'" <[email protected]>
Sent: Tuesday, March 07, 2006 12:54 PM
Subject: RE:
[Asterisk-bsd] Zap on hold problem
I'm using eyeBeam and I never
had a problem with HOLD and Transfer with
asterisk.
It might be
something with your extensions.conf setup.
Do you have the 't' or 'T'
option in the Dial from the ZAP to the SIP ?
Do you have enabled
transfers in the features ?
-----Original Message-----
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Diego Valencia
Sent: Tuesday, March 07, 2006 9:56 AM
To:
Asterisk on BSD discussion
Cc: Olle E Johansson
Subject: Re:
[Asterisk-bsd] Zap on hold problem
Hi Olle, thanks for you reply. Can
you help me about my problem? I can't
transfer the call when it is coming
from zap channel. I want to do this:
PSTN ---> ZAP ----> SIP
----transfer---> SIP
Is it posible?
When I press hold
button, on the pstn side, starts MOH, but I can't return
to the previous
call any more. The eyebeam says "Failed to place call on
hold".
I see
that the UA recieves "not found" from asterisk when it sends the
"on
hold" INVITE.
I was searching on the net and I can't find a user
with the same problem.
:o( I guess that I'm doing something
wrong.
Thanks for any help.
BR
Diego
-----
Original Message -----
From: "Olle E Johansson" <[EMAIL PROTECTED]>
To:
"Asterisk on BSD discussion" <[email protected]>
Cc: "Olle E Johansson" <[EMAIL PROTECTED]>
Sent:
Monday, March 06, 2006 5:23 PM
Subject: Re: [Asterisk-bsd] Zap on hold
problem
>
> 6 mar 2006 kl. 20.47 skrev Diego
Valencia:
>
>> Hi, anybody knows if is normal the "Ignoring
this INVITE request"?:
>> The call is incoming from zap channel,
this invite is when I put the
>> call on hold, and the UA
does not get a response.
> This means that we are getting a repeated
transmission of an INVITE that
> we already have and are
processing. The second one will be ignored.
>
> /O
>
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list
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> http://lists.digium.com/mailman/listinfo/asterisk-bsd
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