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<DIV><FONT face=Arial size=2>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.</FONT></DIV>
<DIV><FONT face=Arial size=2>Now I recompiled asterisk without
the patch and the transfer problem is solved.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks for you help</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Diego</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=marios@comand.net href="mailto:marios@comand.net">Marios Andreou</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-bsd@lists.digium.com
href="mailto:asterisk-bsd@lists.digium.com">'Asterisk on BSD discussion'</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Wednesday, March 08, 2006 5:54
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> RE: [Asterisk-bsd] Zap on hold
problem</DIV>
<DIV><BR></DIV>
<DIV dir=ltr align=left><SPAN class=718475020-08032006><FONT face=Tahoma
color=#0000ff size=2>I just tried it to see what happens.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=718475020-08032006><FONT face=Tahoma
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=718475020-08032006><FONT face=Tahoma
color=#0000ff size=2>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.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=718475020-08032006><FONT face=Tahoma
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=718475020-08032006><FONT face=Tahoma
color=#0000ff size=2>Is the eyeBeam behind a NAT ?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=718475020-08032006><FONT face=Tahoma
color=#0000ff size=2>Is asterisk and eyeBeam on the same
network?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=718475020-08032006><FONT face=Tahoma
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=718475020-08032006></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=718475020-08032006><FONT face=Tahoma
color=#0000ff size=2></FONT></SPAN> </DIV><BR>
<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> <A
href="mailto:asterisk-bsd-bounces@lists.digium.com">asterisk-bsd-bounces@lists.digium.com</A>
[mailto:asterisk-bsd-bounces@lists.digium.com] <B>On Behalf Of </B>Diego
Valencia<BR><B>Sent:</B> Tuesday, March 07, 2006 5:15 PM<BR><B>To:</B>
Asterisk on BSD discussion<BR><B>Subject:</B> Re: [Asterisk-bsd] Zap on hold
problem<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><FONT face=Arial size=2>Thanks Marios, I made you told me, and it
works fine, but we need a supervised transfer.</FONT></DIV>
<DIV><FONT face=Arial size=2>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)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>This is the invite from eyebeam:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>SENDING TO: {ip of asterisk} :5060<BR>INVITE
sip:asterisk@{ip of asterisk} SIP/2.0<BR>To: "asterisk"<sip:asterisk@{ip
of asterisk}>;tag=as63cf8d5d<BR>From: <sip:233@{ip of
eyebeam}:6199>;tag=b10c4f30<BR>Via: SIP/2.0/UDP {ip of
eyebeam}:6199;branch=z9hG4bK-d87543-252324346-1--d87543-;rport<BR>Call-ID:
</FONT><A href="mailto:49d9935e6b5a26326b81d3ad187039ff@{ip"><FONT
face=Arial size=2>49d9935e6b5a26326b81d3ad187039ff@{ip</FONT></A><FONT
face=Arial size=2> of asterisk}<BR>CSeq: 2 INVITE<BR>Contact:
<sip:233@{ip of eyebeam}:6199><BR>Max-Forwards: 70<BR>Allow: INVITE,
ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO<BR>Content-Type: application/sdp<BR>User-Agent: eyeBeam release 3004t
stamp 16741<BR>Content-Length: 273</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=- 28646833 28659668 IN IP4 {ip of
eyebeam}<BR>s=eyeBeam<BR>c=IN IP4 0.0.0.0<BR>t=0 0<BR>m=audio 9296 RTP/AVP 0
8 101<BR>a=alt:1 1 : 9CAD96D3 7C38BE5D {ip of eyebeam} 9296<BR>a=fmtp:101
0-15<BR>a=rtpmap:101 telephone-event/8000<BR>a=sendonly</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Does asterisk know that this invite is for
him? the packet is send to <A href="mailto:asterisk@ip">asterisk@ip</A>, Is
necesary define "asterisk" name on /etc/hosts? My host name is
ip-pbx.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Diego</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>----- Original Message ----- </FONT>
<DIV><FONT face=Arial size=2>From: "Marios Andreou" <</FONT><A
href="mailto:marios@comand.net"><FONT face=Arial
size=2>marios@comand.net</FONT></A><FONT face=Arial size=2>></FONT></DIV>
<DIV><FONT face=Arial size=2>To: "'Asterisk on BSD discussion'"
<</FONT><A href="mailto:asterisk-bsd@lists.digium.com"><FONT face=Arial
size=2>asterisk-bsd@lists.digium.com</FONT></A><FONT face=Arial
size=2>></FONT></DIV>
<DIV><FONT face=Arial size=2>Sent: Tuesday, March 07, 2006 4:37
PM</FONT></DIV>
<DIV><FONT face=Arial size=2>Subject: RE: [Asterisk-bsd] Zap on hold
problem</FONT></DIV></DIV>
<DIV><FONT face=Arial><BR><FONT size=2></FONT></FONT></DIV><FONT face=Arial
size=2>This is strange.<BR>So you press line 1 again on eyeBeam and it
doesn't get you back to the first call?<BR>Hmm.<BR>Let's try then the
Asterisk transfer instead or the eybeam.<BR>In features.conf change
<BR>;blindxfer => #1<BR>To<BR>blindxfer => #<BR><BR>In *CLI> reload
res_features.so<BR><BR>Make a call to Zap->eyeBeam<BR>Answer eyeBeam and
press #<BR>You should hear "Transferring"<BR>Enter another extension once
successful transfer eyeBeam will hangup<BR>If this works then there is no
problem with asterisk and Zap.<BR><BR>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<BR>and the two are
connected.<BR> <BR><BR>-----Original Message-----<BR>From: </FONT><A
href="mailto:asterisk-bsd-bounces@lists.digium.com"><FONT face=Arial
size=2>asterisk-bsd-bounces@lists.digium.com</FONT></A><FONT face=Arial
size=2> [mailto:asterisk-bsd-bounces@lists.digium.com] On Behalf Of Diego
Valencia<BR>Sent: Tuesday, March 07, 2006 1:15 PM<BR>To: Asterisk on BSD
discussion<BR>Subject: Re: [Asterisk-bsd] Zap on hold problem<BR><BR>Hi
Marios, I don't have problem transfering sips, I only have problem when
<BR>the call is coming form zap channel. There is a setting for zapata
<BR>transfers? Theses are my conf:<BR><BR>features.conf<BR><BR>;<BR>; Sample
Parking configuration<BR>;<BR><BR>[general]<BR>parkext =>
700
; What ext. to dial to park<BR>parkpos =>
701-720
; What extensions to park calls on<BR>context =>
parkedcalls ; Which
context parked calls are in<BR>;parkingtime =>
45
; Number of seconds a call can be parked
for<BR>
; (default is 45 seconds)<BR>;transferdigittimeout =>
3 ; Number of seconds to wait between digits
<BR>when transfering a call<BR>;courtesytone =
beep ;
Sound file to play to the parked
caller<BR>
; when someone dials a parked call<BR>;xfersound =
beep
; to indicate an attended transfer is <BR>complete<BR>;xferfailsound =
beeperr ; to indicate a failed
transfer<BR>;adsipark =
yes
; if you want ADSI parking announcements<BR>;findslot =>
next
; Continue to the 'next' parking space. <BR>Defaults to 'first'
available<BR>pickupexten = 8
; Configure the pickup extension. Default is
*8<BR>;featuredigittimeout = 500 ; Max time
(ms) between digits
for<BR>
; feature activation. Default is
500<BR><BR><BR>[featuremap]<BR>;blindxfer =>
#1
; Blind transfer<BR>;disconnect =>
*0
; Disconnect<BR>;automon =>
*1
; One Touch Record<BR>;atxfer =>
*2
; Attended transfer<BR><BR>[applicationmap]<BR>;testfeature =>
#9,callee,Playback,tt-monkeys ;Play tt-monkeys
to<BR><BR>sip.conf<BR><BR>[233]<BR>canreinvite=no<BR>username=233<BR>type=friend<BR>context=nacionales<BR>secret=secret233<BR>;subscribecontext=trunklocal<BR>language=es<BR>host=dynamic<BR></FONT><A
href="mailto:mailbox=233@default,233"><FONT face=Arial
size=2>mailbox=233@default,233</FONT></A><BR><FONT face=Arial
size=2>disallow=all<BR>allow=g729<BR>allow=ulaw<BR>allow=alaw<BR><BR>[240]<BR>canreinvite=no<BR>username=240<BR>type=friend<BR>context=nacionales<BR>secret=secret240<BR>;subscribecontext=trunklocal<BR>language=es<BR>host=dynamic<BR></FONT><A
href="mailto:mailbox=233@default,233"><FONT face=Arial
size=2>mailbox=233@default,233</FONT></A><BR><FONT face=Arial
size=2>disallow=all<BR>allow=g729<BR>allow=ulaw<BR>allow=alaw<BR><BR>extensions.conf:<BR><BR><BR><BR>[macro-stdexten];<BR>;<BR>;
Standard extension macro:<BR>; ${ARG1} - Extension (we
could have used ${MACRO_EXTEN} here as well<BR>; ${ARG2} -
Device(s) to ring<BR>;<BR>exten =>
s,1,Dial(${ARG2},30,t)
; Ring the <BR>interface, 20 seconds maximum<BR>exten =>
s,2,Goto(s-${DIALSTATUS},1)
; Jump based <BR>on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)<BR>exten =>
s-NOANSWER,1,Dial(SIP/1222,30,)
; retorana a <BR>la consola<BR>exten => s-NOANSWER,2,Hangup<BR>;exten
=>
s-BUSY,1,MusicOnHold(ringbusy)
; If busy, send to <BR>voicemail w/ busy announce<BR>exten =>
s-BUSY,1,Hangup<BR>exten =>
_s-.,1,Goto(s-NOANSWER,1)
; Treat <BR>anything else as no answer<BR><BR>[incomingzap]<BR><BR>include
=> internos<BR><BR>exten =>
s,1,Wait,1
; Wait a second, just for fun<BR>;exten =>
s,n,Set(SIP_CODEC=ulaw)<BR>exten =>
s,2,Answer
; Answer the line<BR>exten =>
s,3,Set(TIMEOUT(digit)=5) ; Set Digit Timeout
to 5 seconds<BR>exten => s,4,Set(TIMEOUT(response)=3) ; Set
Response Timeout to 10 seconds<BR>exten =>
s,5,Set(LANGUAGE()=es) ; Set
language to french<BR>exten => s,6(restart),BackGround(welcome) ; Play a
congratulatory message<BR>exten =>
s,7,WaitExten ; Wait
for an extension to be dialed.<BR>exten =>
s,8,Dial(SIP/232,30<BR><BR>zapata.conf<BR><BR>[channels]<BR><BR>faxdetect=incoming<BR>hanguponpolarityswitch=yes<BR>busydetect=yes<BR>busycount=4<BR>immediate
=> no<BR>transfer => yes<BR>cancallforward =>
yes<BR>threewaycalling => yes<BR>callreturn =>
yes<BR>usecallerid=yes<BR>hidecallerid=no<BR>group => 1<BR>context =>
incomingzap<BR>signalling => fxs_ks<BR>amaflags =>
documentation<BR>echocancel=yes
;Cancela el echo producido por las lineas
<BR>análogas<BR>echocancelwhenbridged=yes<BR>echotraining=yes<BR>channel
=> 1-2<BR><BR>------------------------------<BR><BR>Call flow:<BR><BR>--
Starting simple switch on 'Zap/2-1'<BR>Mar 5 13:46:41 NOTICE[3477]:
chan_zap.c:6063 ss_thread: Got event 2
<BR>(Ring/Answered)...<BR> -- Executing Wait("Zap/2-1", "1") in
new stack<BR> -- Executing Answer("Zap/2-1", "") in new
stack<BR> -- Executing Set("Zap/2-1", "TIMEOUT(digit)=5") in new
stack<BR> -- Digit timeout set to 5<BR> -- Executing
Set("Zap/2-1", "TIMEOUT(response)=3") in new stack<BR> --
Response timeout set to 3<BR> -- Executing Set("Zap/2-1",
"LANGUAGE()=es") in new stack<BR> -- Executing
BackGround("Zap/2-1", "welcome") in new stack<BR> -- Playing
'welcome' (language 'es')<BR> == CDR updated on Zap/2-1<BR>
-- Executing Macro("Zap/2-1", "stdexten|233|SIP/233") in new stack (233
<BR>is eyebeam)<BR> -- Executing Dial("Zap/2-1", "SIP/233|30|t")
in new stack<BR> -- Called 233<BR> -- SIP/233-7aa8
is ringing<BR> -- SIP/233-7aa8 answered Zap/2-1
-------------> I press "line 2" button <BR>on eyebeam to call to other
extension<BR> -- Started music on hold, class 'default', on
Zap/2-1 ---------> MOH on <BR>ZAP<BR><BR>At this point the caller (PSTN)
is on MOH, but I can't return to call 1 to <BR>transfer it. After a few
minutes the eyebeam says "Failed to place call on
<BR>hold"<BR><BR><BR>Thanks<BR><BR>Diego<BR><BR>----- Original Message -----
<BR>From: "Marios Andreou" <</FONT><A
href="mailto:marios@comand.net"><FONT face=Arial
size=2>marios@comand.net</FONT></A><FONT face=Arial size=2>><BR>To:
"'Asterisk on BSD discussion'" <</FONT><A
href="mailto:asterisk-bsd@lists.digium.com"><FONT face=Arial
size=2>asterisk-bsd@lists.digium.com</FONT></A><FONT face=Arial
size=2>><BR>Sent: Tuesday, March 07, 2006 12:54 PM<BR>Subject: RE:
[Asterisk-bsd] Zap on hold problem<BR><BR><BR>I'm using eyeBeam and I never
had a problem with HOLD and Transfer with <BR>asterisk.<BR>It might be
something with your extensions.conf setup.<BR><BR>Do you have the 't' or 'T'
option in the Dial from the ZAP to the SIP ?<BR>Do you have enabled
transfers in the features ?<BR><BR><BR>-----Original Message-----<BR>From:
</FONT><A href="mailto:asterisk-bsd-bounces@lists.digium.com"><FONT
face=Arial size=2>asterisk-bsd-bounces@lists.digium.com</FONT></A><FONT
face=Arial size=2> <BR>[mailto:asterisk-bsd-bounces@lists.digium.com] On
Behalf Of Diego Valencia<BR>Sent: Tuesday, March 07, 2006 9:56 AM<BR>To:
Asterisk on BSD discussion<BR>Cc: Olle E Johansson<BR>Subject: Re:
[Asterisk-bsd] Zap on hold problem<BR><BR>Hi Olle, thanks for you reply. Can
you help me about my problem? I can't<BR>transfer the call when it is coming
from zap channel. I want to do this:<BR><BR>PSTN ---> ZAP ----> SIP
----transfer---> SIP<BR><BR>Is it posible?<BR><BR>When I press hold
button, on the pstn side, starts MOH, but I can't return<BR>to the previous
call any more. The eyebeam says "Failed to place call on<BR>hold".<BR>I see
that the UA recieves "not found" from asterisk when it sends the
"on<BR>hold" INVITE.<BR>I was searching on the net and I can't find a user
with the same problem.<BR>:o( I guess that I'm doing something
wrong.<BR><BR>Thanks for any help.<BR><BR>BR<BR><BR>Diego<BR><BR><BR>-----
Original Message ----- <BR>From: "Olle E Johansson" <</FONT><A
href="mailto:oej@edvina.net"><FONT face=Arial
size=2>oej@edvina.net</FONT></A><FONT face=Arial size=2>><BR>To:
"Asterisk on BSD discussion" <</FONT><A
href="mailto:asterisk-bsd@lists.digium.com"><FONT face=Arial
size=2>asterisk-bsd@lists.digium.com</FONT></A><FONT face=Arial
size=2>><BR>Cc: "Olle E Johansson" <</FONT><A
href="mailto:oej@edvina.net"><FONT face=Arial
size=2>oej@edvina.net</FONT></A><FONT face=Arial size=2>><BR>Sent:
Monday, March 06, 2006 5:23 PM<BR>Subject: Re: [Asterisk-bsd] Zap on hold
problem<BR><BR><BR>><BR>> 6 mar 2006 kl. 20.47 skrev Diego
Valencia:<BR>><BR>>> Hi, anybody knows if is normal the "Ignoring
this INVITE request"?:<BR>>> The call is incoming from zap channel,
this invite is when I put the<BR>>> call on hold, and the UA
does not get a response.<BR>> This means that we are getting a repeated
transmission of an INVITE that<BR>> we already have and are
processing. The second one will be ignored.<BR>><BR>> /O<BR>>
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