[Asterisk-bsd] Zap on hold problem
Diego Valencia
dvalencia at powervt.com.ar
Tue Mar 7 15:15:00 MST 2006
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:asterisk@{ip of asterisk} SIP/2.0
To: "asterisk"<sip:asterisk@{ip of asterisk}>;tag=as63cf8d5d
From: <sip:233@{ip of eyebeam}:6199>;tag=b10c4f30
Via: SIP/2.0/UDP {ip of eyebeam}:6199;branch=z9hG4bK-d87543-252324346-1--d87543-;rport
Call-ID: 49d9935e6b5a26326b81d3ad187039ff@{ip of asterisk}
CSeq: 2 INVITE
Contact: <sip:233@{ip 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 asterisk at ip, Is necesary define "asterisk" name on /etc/hosts? My host name is ip-pbx.
Diego
----- Original Message -----
From: "Marios Andreou" <marios at comand.net>
To: "'Asterisk on BSD discussion'" <asterisk-bsd at lists.digium.com>
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: asterisk-bsd-bounces at lists.digium.com [mailto:asterisk-bsd-bounces at lists.digium.com] 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
mailbox=233 at default,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
mailbox=233 at default,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" <marios at comand.net>
To: "'Asterisk on BSD discussion'" <asterisk-bsd at lists.digium.com>
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: asterisk-bsd-bounces at lists.digium.com
[mailto:asterisk-bsd-bounces at lists.digium.com] 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" <oej at edvina.net>
To: "Asterisk on BSD discussion" <asterisk-bsd at lists.digium.com>
Cc: "Olle E Johansson" <oej at edvina.net>
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
> _______________________________________________
> Asterisk-BSD mailing list
> Asterisk-BSD at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-bsd
_______________________________________________
Asterisk-BSD mailing list
Asterisk-BSD at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-bsd
_______________________________________________
Asterisk-BSD mailing list
Asterisk-BSD at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-bsd
_______________________________________________
Asterisk-BSD mailing list
Asterisk-BSD at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-bsd
_______________________________________________
Asterisk-BSD mailing list
Asterisk-BSD at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-bsd
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-bsd/attachments/20060307/d91ea6e2/attachment-0001.htm
More information about the Asterisk-BSD
mailing list