Re: [asterisk-users] Transfer calls "on demand"

[Daniel Heckl]

You are searching for „Call Pickup“. It is implemented in Asterisk by default.

https://wiki.asterisk.org/wiki/display/AST/Call+Pickup 

Take a look under section „Configuration Options“.

Daniel

> Am 29.12.2015 um 07:53 schrieb Luca Bertoncello :
> 
> Hi list!
> 
> Right now I configured my Asterisk to forward the calls for the number X to
> both phones (mine and the phone of my wife).
> It works, of course, but I'm not enthusiast...
> 
> I see what we have at office: if one phone rings, other phones in the same
> group can "catch the call", so that if a colleague is not present, another
> colleague can catch the call.
> 
> I'd like to have the same procedure at home. I think, Asterisk can do that,
> but I have no idea how to implement this.
> 
> Shortly: what I want is that every phone rings only on calls for the own
> number, and I can catch the call from the other phone, if for example my wife
> is not at home, for example pressing "*5#" or other key combination.
> 
> Thanks a lot for your suggestion!
> 
> Luca Bertoncello
> (lucab...@lucabert.de)
> 
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Re: [asterisk-users] Transfer calls "on demand"

[Luca Bertoncello]

Daniel Heckl  schrieb:

> You are searching for „Call Pickup“. It is implemented in Asterisk by
> default.
> 
> https://wiki.asterisk.org/wiki/display/AST/Call+Pickup
>  Take a look under
> section „Configuration Options“.

Hi, Daniel!

Thanks for your answer...
I'm using Asterisk 1.8.30.0 on an OpenWRT-Router.
I found the configuration for call pickup in the sip.conf and features.conf,
so I tried to activate it...
Unfortunately, unsuccessfully...

So, my sip.conf:

callgroup=1,3-4 ; We are in caller groups 1,3,4
pickupgroup=1,3-5   ; We can do call pick-p for call group 1,3,4,5

my features.conf:

; Pickup Options
;
pickupexten = *8   ; Configure the pickup extension. (default is *8)
;pickupsound = beep ; to indicate a successful pickup (default: no 
sound)
;pickupfailsound = beeperr  ; to indicate that the pickup failed (default: 
no sound)

my users.conf:

[general]
callgroup = 1
pickupgroup = 1

my extensions.conf:

[anika_incoming]
exten => _0049351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
exten => _0049351222,Set(CHANNEL(pickupgroup)=1)
exten => _0049351222,n,Dial(local/222@anika_incoming)
exten => _0351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
exten => _0351222,n,Dial(local/222@anika_incoming)
exten => _222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
exten => _222,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" = "+49" 
]?0${CALLERID(num):3}:${CALLERID(num)})})  ; Damit das "+49" mit "0" ersetzt 
wird
exten => _222,n,Set(CHANNEL(musicclass)=default)
;;;exten => 
_222,n,Dial(SIP/0049351222/1@luca_for_anika_voip_mobile,19,RcxX)
exten => _222,n,Dial(SIP/0049351222,19,RcxX)
exten => _222,n,Verbose(2,Voicemail for Anika)
exten => _222,n,Set(CALLERID(name)=)   
; Damit in der E-Mail der AB nicht den Namen steht
exten => _222,n,VoiceMail(0049351222,us)
exten => _222,n,Hangup

Then I called the 222 with my mobile phone and I tried to get the call
from the other phone, calling the *8.
Unfortunately I get an error (invalid number) on the display of the phone,
and the phone 222 continue to ring.
No error on the log of Asterisk...

Any suggestion?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Transfer calls "on demand"

[Daniel Heckl]

On top of the page: "Call pickup support added in Asterisk 11“

I think that is the problem. I do not know a solution for 1.8, but maybe 
someone other.

> Am 29.12.2015 um 10:20 schrieb Luca Bertoncello :
> 
> Daniel Heckl  schrieb:
> 
>> You are searching for „Call Pickup“. It is implemented in Asterisk by
>> default.
>> 
>> https://wiki.asterisk.org/wiki/display/AST/Call+Pickup
>>  Take a look under
>> section „Configuration Options“.
> 
> Hi, Daniel!
> 
> Thanks for your answer...
> I'm using Asterisk 1.8.30.0 on an OpenWRT-Router.
> I found the configuration for call pickup in the sip.conf and features.conf,
> so I tried to activate it...
> Unfortunately, unsuccessfully...
> 
> So, my sip.conf:
> 
> callgroup=1,3-4 ; We are in caller groups 1,3,4
> pickupgroup=1,3-5   ; We can do call pick-p for call group 1,3,4,5
> 
> my features.conf:
> 
> ; Pickup Options
> ;
> pickupexten = *8   ; Configure the pickup extension. (default is 
> *8)
> ;pickupsound = beep ; to indicate a successful pickup (default: 
> no sound)
> ;pickupfailsound = beeperr  ; to indicate that the pickup failed 
> (default: no sound)
> 
> my users.conf:
> 
> [general]
> callgroup = 1
> pickupgroup = 1
> 
> my extensions.conf:
> 
> [anika_incoming]
> exten => _0049351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _0049351222,Set(CHANNEL(pickupgroup)=1)
> exten => _0049351222,n,Dial(local/222@anika_incoming)
> exten => _0351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _0351222,n,Dial(local/222@anika_incoming)
> exten => _222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _222,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" = "+49" 
> ]?0${CALLERID(num):3}:${CALLERID(num)})})  ; Damit das "+49" mit "0" ersetzt 
> wird
> exten => _222,n,Set(CHANNEL(musicclass)=default)
> ;;;exten => 
> _222,n,Dial(SIP/0049351222/1@luca_for_anika_voip_mobile,19,RcxX)
> exten => _222,n,Dial(SIP/0049351222,19,RcxX)
> exten => _222,n,Verbose(2,Voicemail for Anika)
> exten => _222,n,Set(CALLERID(name)=)  
>  ; Damit in der E-Mail der AB nicht den Namen steht
> exten => _222,n,VoiceMail(0049351222,us)
> exten => _222,n,Hangup
> 
> Then I called the 222 with my mobile phone and I tried to get the call
> from the other phone, calling the *8.
> Unfortunately I get an error (invalid number) on the display of the phone,
> and the phone 222 continue to ring.
> No error on the log of Asterisk...
> 
> Any suggestion?
> 
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
> 
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Re: [asterisk-users] Transfer calls "on demand"

[Doug Lytle]

Luca Bertoncello wrote:

So, my sip.conf:

callgroup=1,3-4 ; We are in caller groups 1,3,4
pickupgroup=1,3-5   ; We can do call pick-p for call group 1,3,4,5


Keep it simple for testing.  My sip.conf on a working Asterisk system below:

[4220](stemplet)
defaultuser=4220
pickupgroup=1
callgroup=1
callerid = Operator <4220>

[4232](stemplet)
defaultuser=4232
mailbox = 4232@sip
context = sip
callgroup=1
pickupgroup=1
callerid = Judy K. <4232>


My features.conf:

; Pickup Options
;
pickupexten = *7; Configure the pickup extension. 
(default is *8)


This has worked for me on all version of Asterisk that we've used; we're 
currently running 11.  In the above, The operator can pickup calls for 
Judy and vise versa using *7.


You'll also want to make sure your phone's digit map understands that *7 
is a valid key combination.  We're using Polycom phones.


Doug


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Re: [asterisk-users] Transfer calls "on demand"

[Luca Bertoncello]

Doug Lytle  schrieb:

> Keep it simple for testing.  My sip.conf on a working Asterisk system below:

IT WORKS!!!

Thanks a lot!
Luca Bertoncello
(lucab...@lucabert.de)

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[asterisk-users] Transfer calls "on demand"

[Luca Bertoncello]

Hi list!

Right now I configured my Asterisk to forward the calls for the number X to
both phones (mine and the phone of my wife).
It works, of course, but I'm not enthusiast...

I see what we have at office: if one phone rings, other phones in the same
group can "catch the call", so that if a colleague is not present, another
colleague can catch the call.

I'd like to have the same procedure at home. I think, Asterisk can do that,
but I have no idea how to implement this.

Shortly: what I want is that every phone rings only on calls for the own
number, and I can catch the call from the other phone, if for example my wife
is not at home, for example pressing "*5#" or other key combination.

Thanks a lot for your suggestion!

Luca Bertoncello
(lucab...@lucabert.de)

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[asterisk-users] Transfer

[Dan Cropp]

I am running Asterisk 13.5.0.

I have the Transfer working when using the chan_sip support.
However, when I try to perform a Transfer using pjsip, it is failing.

The one difference I am seeing in the SIP trace is chan_sip automatically sends 
the Referred-By.  PJSIP does not.
The switch provider I am working with has never seen a REFER without the 
Referred-By line

In both cases, I am performing the Transfer via AMI
EXEC Transfer 

Does Asterisk 13.5.0 PJSIP support require a flag or something to force the 
Referred-By line to automatically be passed when a Transfer is performed?

chan_sip (succeeds)
19:27:32.512123 IP (tos 0x0, ttl 64, id 11492, offset 0, flags [none], proto 
UDP (17), length 630)
192.168.xxx.xxx.sip  192.168.yyy.yyy.sip: SIP, length: 602
REFER sip:3...@192.168.yyy.yyy:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;branch=z9hG4bK58f4bd1d
Max-Forwards: 70
From: sip:3...@192.168.xxx.xxx;tag=as44000cf4
To: sip:3...@192.168.yyy.yyy;tag=7Iy0JkwDC
Contact: sip:3...@192.168.xxx.xxx:5060
Call-ID: jdeuqpak-00...@192.168.yyy.yyy
CSeq: 102 REFER
User-Agent: Asterisk PBX 13.5.0
Date: Thu, 20 Aug 2015 19:27:32 GMT
Refer-To: sip:3...@192.168.yyy.yyy
Referred-By: sip:3...@192.168.xxx.xxx:5060
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Pjsip
18:46:58.386372 IP (tos 0x0, ttl 64, id 38690, offset 0, flags [DF], proto UDP 
(17), length 654)
192.168.xxx.xxx.sip  192.168.yyy.yyy.sip: SIP, length: 626
REFER sip:3...@192.168.yyy.yyy:5060 SIP/2.0
Via: SIP/2.0/UDP 
192.168.xxx.xxx:5060;rport;branch=z9hG4bKPjec41c3b9-d734-482d-82c1-2a6f8d9452a3
From: 
sip:3...@192.168.xxx.xxx;tag=3c10f423-e468-42ea-87a1-658ae106581c
To: sip:3...@192.168.yyy.yyy;tag=WITKDakt
Contact: sip:192.168.xxx.xxx:5060
Call-ID: s6wk6l6q-00...@192.168.yyy.yyy
CSeq: 981 REFER
Event: refer
Expires: 600
Supported: 100rel, timer, replaces, norefersub
Accept: message/sipfrag;version=2.0
Allow-Events: message-summary, presence, dialog, refer
Refer-To: sip:3...@192.168.yyy.yyy
Max-Forwards: 70
User-Agent: Asterisk PBX 13.5.0
Content-Length:  0
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[asterisk-users] Transfer call question

[Nick Awesome]

Hello guys,

I have trunk “1, Internal num “99 and MeetMe “1010

now I calling 99 - 89264959635 via 1

 /pbx/agi.php: [agi_channel] = PJSIP/99-0012
 /pbx/agi.php: [agi_callerid] = 99
 /pbx/agi.php: [agi_calleridname] = 99
 /pbx/agi.php: [agi_context] = dialmap
 /pbx/agi.php: [agi_extension] = 89264959635

then I would like to direct transfer this call to 1010
and when I do that from my phone I getting this agi_request in AGI: 

 /pbx/agi.php: [agi_channel] = PJSIP/1-0013
 /pbx/agi.php: [agi_callerid] = 89264959635
 /pbx/agi.php: [agi_calleridname] = unknown
 /pbx/agi.php: [agi_context] = dialmap
 /pbx/agi.php: [agi_extension] = 1010

There is no information who is transferring that call, so AGI thinks that it is 
inbound call and hangup it because in my case external 89264959635 to internal 
1010 is denied.
is there way do determine that call was transfered from 99 so I can use route 
table of abonent 99 to connect the call properly?
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[asterisk-users] Transfer call placed from console (with chan_alsa)

[Alex]

Hi everyone.

Having experimented a but with a prototype of a system I described in
an earlier thread (Reading DTMF sent by callee during a SIP call), I
decided to implement my requirement by transferring the call to
another extension. This way, the callee can open the door by pressing
#1, and the dial plan for extension 1 takes care of the rest.

This works when I make a typical SIP to SIP call, but it doesn't when
I call from the console, using chan_alsa. I can see that the transfer
feature is inactive:

rasterisk*CLI core show channeltype console
-- Info about channel driver: Console --
  Device State: no
Indication: yes
 Transfer : no
  Capabilities: 0x40 (slin)
   Digit Begin: no
 Digit End: yes
Send HTML : no
 Image Support: no
  Text Support: yes



However, I am unable to find a way to activate it. How can I transfer
placed from the console? Is it possible, in principle?


Alex

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[asterisk-users] transfer capabilities

[Andrew Nowrot]

The company that looks after my clients internal phone system has a
problem with logging in to the PABX using their data modem.

Connection looks like this

ISDN PRA from telco - Asterisk - SIP Trunk to my clients Asterisk -
my clients Asterisk - E1 port to his old PABX

I am planning to use transfer capabilities on DAHDI channel to my
clients PABX. Should I also set it on sip trunk and channel from the
telco in order for it to work?

Best regards

Andrew
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[asterisk-users] Transfer rights for attended transfers

[jg]

Recently I asked a question about possibly unwanted calls due to extended transfer rights after 
attended transfers using DTMF sequences 
(http://lists.digium.com/pipermail/asterisk-users/2013-September/280536.html). Obviously, 
transferring with SIP INVITEs (hold + transfer keys) is not immediately affected by the this, 
but it is not always possible to enforce this.


Meanwhile I have changed the dial options in my dial plans in such a way that they are based on 
the origin of the participating channels regardless of the number of transfers and the local 
intermediate contexts.


I think it would be easier to handle the rights with the device configuration and not at the 
switching points in the dialplan. Depending on the origin of the call the caller or callee or 
both should have transfer rights. For a call from outside (e.g. via an ISDN channel) only the 
callee should have transfer rights. For internal calls typically both parties are allowed to 
transfer and for calls to the outside only the caller should have the rights. At least this is 
the setup I typically use. A simple flag allowtransfer, like for the old sip channel, should 
resolve this. I think it is the devices that matter when creating a bridge and not the current 
context in the dialplan.


This is an example of what is currently happening for an attended transfer when DTMF sequences 
are allowed


Call from outside:

[from-pstn]
exten = _X.,1,Dial(SIP/,,...t...); fine -- only callee can transfer

Attended transfer (Asterisk uses a Local channel to connect):

[from-internal]
exten = 1234,1,Dial(Local/,,Tt...); bad -- from here on the outside caller can do 
whatever he wants


in this case it should be

exten = 1234,1,Dial(Local/,,t...)

Call from inside:

[from-internal]
; e.g.
exten = _X.,1,Dial(DAHDI/r2/${FILTER(0-9,${EXTEN})},,...T...)  ; fine -- only 
caller can transfer

Atx:

[from-internal]
exten = 1234,1,Dial(Local/,,Tt...); bad again

in this case it should be

exten = 1234,1,Dial(Local/,,T...)

If both parties are local then exten = 1234,1,Dial(,,Tt...) is fine.

I would like to hear comments about this, even if I have overlooked something that makes me look 
stupid.


One last remark. It may not be easy to explore this when using 2 Asterisk systems as the DTMF 
sequences are first caught by the originating system (but the entries in features.conf can be 
deleted).


jg

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Re: [asterisk-users] Transfer Fraud

[jg]

I should have mentioned that I am looking at attended transfers.

When a Local channel gets created for the announcement it is in a way on behalf of the caller, 
but has the permissions of an internal channel. Depending on the origin of the call the transfer 
permissions should then be set dynamically. I am currently looking to what degree I can use the 
BRIDGEPEER and TRANSFER_CONTEXT for that purpose. If I can't use them, I always can introduce my 
own persistent variable. In either case I have to manually set the transfer permissions.


Is there a simpler way of taking care of this potential problem, or am I not aware of some 
important concept?


jg

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Re: [asterisk-users] Transfer Fraud

[Adrian Serafini]

On 09/13/2013 04:12 PM, jg wrote:
Is there a general recipe to avoid fraudulent calls under the 
following conditions?


A receptionist transfers calls as a callee (customers are calling) and 
as a caller (boss asks to call and then transfer to him), i.e. the 
Dial cmd for the internal context contains Tt. Then an outside call 
would operate as a Local channel in an internal context after the 
first transfer. If the internal context allows to dial outside, which 
is quite common, then this can be abused by the outside caller.


An obvious solution is to disallow Local channels to call outside 
lines, but there are some possible side effects if Local channels are 
used explicitly. This would require adding a persistent channel 
variable (the ones with __).


create a separate context for outbound calls.

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[asterisk-users] Transfer Fraud

[jg]

Is there a general recipe to avoid fraudulent calls under the following 
conditions?

A receptionist transfers calls as a callee (customers are calling) and as a caller (boss asks to 
call and then transfer to him), i.e. the Dial cmd for the internal context contains Tt. Then 
an outside call would operate as a Local channel in an internal context after the first 
transfer. If the internal context allows to dial outside, which is quite common, then this can 
be abused by the outside caller.


An obvious solution is to disallow Local channels to call outside lines, but there are some 
possible side effects if Local channels are used explicitly. This would require adding a 
persistent channel variable (the ones with __).


I apologize if this type of question has already been asked before.

jg

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Re: [asterisk-users] Transfer Fraud

[jg]

create a separate context for outbound calls.

Wouldn't that be more or less identical to my way? I would have to dispatch the channel to see 
whether it is allowed to enter the outbound context. Maybe I misunderstood something.


jg

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Re: [asterisk-users] Transfer Fraud

[Eric Wieling]

This is one of the disadvantages of using phones without a transfer button.  

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg
Sent: Friday, September 13, 2013 4:52 PM
To: adrian-li...@wombit.com; Asterisk Users Mailing List - Non-Commercial 
Discussion
Subject: Re: [asterisk-users] Transfer Fraud


 create a separate context for outbound calls.

Wouldn't that be more or less identical to my way? I would have to dispatch the 
channel to see whether it is allowed to enter the outbound context. Maybe I 
misunderstood something.

jg

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[asterisk-users] Transfer cmd via AsyncAGI

[Dan Cropp]

Hello,

 

I am using Asterisk 11.0.1 and do not notice any changes regarding the
Transfer on newer Asterisk 11.x versions.

 

I am using AMI and controlling a channel via AsyncAGI.

 

I send a Transfer cmd (such as the following)

 

Action: AGI

ActionID: C8

Channel: SIP/1004-0002

CommandID: C8

Command: EXEC Transfer SIP/1003

 

Destination phone starts ringing.

If it answers the call, everything works fine.  I am notified the
agiexec completed successfully and given a TRANSFERSTATUS of SUCCESS.  I
am also notified when the call is hungup so that I can cleanup
information regarding the call.

 

Event: Hangup

Privilege: call,all

Channel: SIP/1004-0002

Uniqueid: 1367761382.0

CallerIDNum: 1004

CallerIDName: 1004 - Asterisk

ConnectedLineNum: unknown

ConnectedLineName: unknown

AccountCode: 3

Cause: 16

Cause-txt: Normal Clearing

 

 

However, if the destination does not answer the Transfer (SIP REFER) and
I hangup the original call being transferred, it stops ringing the
destination phone and partially notifies me of the completed call.

However, I do not receive any Event: Hangup for the original Channel:
SIP/1004-0002

In fact, I do not receive any information for the agiexec completing.

 

The very last event notification I receive with this channel is the
following

Event: AGIExec

Privilege: agi,all

SubEvent: Start

Channel: SIP/1004-0002

CommandId: 556226156

Command: EXEC Transfer SIP/1003

 

Any suggestions on what I may be doing wrong?

Are there any known fixes for this?

Or should I submit a bug to the developer list?

 

Have a great day!

 

Dan

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Re: [asterisk-users] Transfer only, no outbound calling

[Todd Routhier]

Nathan,

 Yes, SIP.. :-)

I ended up deciding to just not allow attended transfer at all since it
seemed so hard to deal with. If someone really wants attended transfer they
can put the call on hold, dial using the other line then transfer the call
on the other line if they want the call on the other end. Same thing, just
one more step.

I am just going to set a var in sip.conf so when people try to dial out
direct, it will catch it in the dial plan and kill the call. With blind
transfer I can set a var on the way in and it's held onto nicely and I can
allow the transfer based on that.

Again, thanks for your detailed response.



On Tue, Apr 16, 2013 at 9:59 PM, Nathan Anderson nath...@fsr.com wrote:

 On Tuesday, April 16, 2013 6:25 PM, Todd Routhier wrote:

  New Problem, now operators can pick up the previous inbound only line and
  dial out to anything that matches the patterns I have defined in the
  context for their extension in sip.conf.
 
  What I really need to make work here is Attended-Transfer since that is
  what is desired by those using the system.

 I'll assume we are talking about SIP extensions here.

 What is doing the actual transfer?  Is it Asterisk (res_features /
 features.conf), or the phones themselves?

 If it is the phones themselves, you're probably out of luck because in an
 attended transfer scenario, the transferor has to send a regular ol' INVITE
 to the transfer target before sending a REFER to the transferee, and so
 there's really no way that Asterisk can know whether that INVITE to the
 transfer target is someone in the middle of attempting an attended
 transfer, or someone trying to place a regular outbound call.  Your only
 hope would be to sniff the SIP traffic between your handsets and Asterisk,
 and see if there is a SIP header difference that is detectable between what
 your phones generate for an attended transfer vs. an outbound call.  If
 there is, you can use the ${SIP_HEADER()} function in your dialplan to
 check for the presence of that difference in order to determine whether a
 call is an attended transfer or not.

 If you have the option of using Asterisk's built-in attended transfer
 feature (features.conf + passing option 't' to the Dial() command that
 calls a given extension for an inbound call) instead of a button on your
 phones, you can override which context a transfer target's number is
 executed in by overriding the global variable TRANSFER_CONTEXT.  So you can
 create a new stub context that sets your variable to let you know that this
 is a transfer and then jumps to the SIP client's normal context, and set
 TRANSFER_CONTEXT=your_new_context under the [globals] section of
 extensions.conf.  Check for the presence of your variable in the SIP
 client's context, and act accordingly.

 Note that in either scenario, as long as you allow attended transfers, the
 system can be gamed by people.  For example, assuming that extensions can
 call other extensions, someone who wants to make an unsanctioned outbound
 call simply walks over to a vacant phone in another cubicle, calls their
 own phone/extension, rushes back to answer it, and then initiates an
 attended transfer that they never end up completing (they just talk to the
 person they initiated the transfer to the whole time).

 Hope this helps,

 --
 Nathan Anderson
 First Step Internet, LLC
 nath...@fsr.com

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[asterisk-users] Transfer only, no outbound calling

[Todd Routhier]

OK, it's been a while since I drank from the pool of wisdom hear on the
list.

After cracking my head against the wall for a few days trying to figure
this out, I have decided to swallow my pride and take the drink.

So, on to my question:

I have some agents/operators setup in sip.conf which point to a context
where I have just about disabled outbound calls (only specific numbers can
be dialed).

The purpose of this is to allow the inbound calls to come in, then if the
operator has a need, they transfer the call to a pre-defined extension
which lives in the limited context defined in sip.conf.

This has worked for some time to restrict outbound calling and where calls
can be transferred to.

Now I would like to open up the numbers the inbound calls can be
transferred to. So, easy enough I thought and I went on my merry way adding
the regular patterns to the context such as NXXNXX and so on.

Hooray, now the operators can transfer anywhere.

New Problem, now operators can pick up the previous inbound only line and
dial out to anything that matches the patterns I have defined in the
context for their extension in sip.conf.

What I really need to make work here is Attended-Transfer since that is
what is desired by those using the system.

It seems that any variables I try to set on the way in don't carry through
too well during an attended transfer.

Basically, I need the ability to know for sure at the point the call ends
up in the outbound context (defined in sip.conf) if the call is actually a
transfer from an inbound call or if it's a direct dial outbound call with
no incoming call attached. If I can figure out how to know this for sure, I
can just do a GoToIf type of thing in the outbound context that just kills
the call if there is no proof that it's a transfer.

I hope this makes sense, please let me know if more info is needed.

Running Asterisk 1.8.8.0.

A huge thanks in advance to the list for any help with this, it's driving
me batty.

Regards,
 Todd R.
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Re: [asterisk-users] Transfer only, no outbound calling

[Nathan Anderson]

On Tuesday, April 16, 2013 6:25 PM, Todd Routhier wrote:

 New Problem, now operators can pick up the previous inbound only line and
 dial out to anything that matches the patterns I have defined in the
 context for their extension in sip.conf.  
 
 What I really need to make work here is Attended-Transfer since that is
 what is desired by those using the system. 

I'll assume we are talking about SIP extensions here.

What is doing the actual transfer?  Is it Asterisk (res_features / 
features.conf), or the phones themselves?

If it is the phones themselves, you're probably out of luck because in an 
attended transfer scenario, the transferor has to send a regular ol' INVITE to 
the transfer target before sending a REFER to the transferee, and so there's 
really no way that Asterisk can know whether that INVITE to the transfer target 
is someone in the middle of attempting an attended transfer, or someone trying 
to place a regular outbound call.  Your only hope would be to sniff the SIP 
traffic between your handsets and Asterisk, and see if there is a SIP header 
difference that is detectable between what your phones generate for an attended 
transfer vs. an outbound call.  If there is, you can use the ${SIP_HEADER()} 
function in your dialplan to check for the presence of that difference in order 
to determine whether a call is an attended transfer or not.

If you have the option of using Asterisk's built-in attended transfer feature 
(features.conf + passing option 't' to the Dial() command that calls a given 
extension for an inbound call) instead of a button on your phones, you can 
override which context a transfer target's number is executed in by overriding 
the global variable TRANSFER_CONTEXT.  So you can create a new stub context 
that sets your variable to let you know that this is a transfer and then jumps 
to the SIP client's normal context, and set TRANSFER_CONTEXT=your_new_context 
under the [globals] section of extensions.conf.  Check for the presence of your 
variable in the SIP client's context, and act accordingly.

Note that in either scenario, as long as you allow attended transfers, the 
system can be gamed by people.  For example, assuming that extensions can call 
other extensions, someone who wants to make an unsanctioned outbound call 
simply walks over to a vacant phone in another cubicle, calls their own 
phone/extension, rushes back to answer it, and then initiates an attended 
transfer that they never end up completing (they just talk to the person they 
initiated the transfer to the whole time).

Hope this helps,

-- 
Nathan Anderson
First Step Internet, LLC
nath...@fsr.com

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[asterisk-users] Transfer by transfer button, distinguish in dialplan

[Roel van Meer]

Hi list!

We're using Asterisk in a setup where you can transfer a call via the 
Asterisk feature, or via the Transfer button on a SIP phone. Both work.


However, in my dialplan I cannot distinguish normal calls from calls made 
by pressing the Transfer button on a phone.


To clarify, I would like to know the difference in the call from A to B, in 
these two situations:


1 (normal call)
A calls B

2 (transferred call)
C calls A
A presses Transfer button and calls B

Does anyone know a way how to accomplish this? Is it at all possible?

Thanks in advance for any hints,

Roel

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Re: [asterisk-users] Transfer call issue

[Phil Daws]

Is anybody else experiencing this problem ?

-- 
Thanks, Phil

- Original Message -
 Hello,
 
 a client attempted to transfer a call today which failed and returned
 the channel back to her.  When this happened on the console we saw:
 
 Got OK on REFER Notify message
 
 the version that we are running is 1.8.9.2.  Are you aware of any
 none issues please with this version as I could not find anything in
 Jira ?
 --
 Thanks, Phil
 
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[asterisk-users] Transfer call issue

[Phil Daws]

Hello, 

a client attempted to transfer a call today which failed and returned the 
channel back to her.  When this happened on the console we saw:

Got OK on REFER Notify message 

the version that we are running is 1.8.9.2.  Are you aware of any none issues 
please with this version as I could not find anything in Jira ?
-- 
Thanks, Phil

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Re: [asterisk-users] Transfer CDRs

[[Digital^Dude] ®]

Please share if anyone has encountered this cdr issue with call transfer.

On Fri, May 18, 2012 at 5:32 PM, [Digital^Dude] ®
millennium@gmail.comwrote:

 Hello,

 I'm using attended call transfer in asterisk 1.8.11.0 on a CentOS machine.
 Each CDR entry of calls that are transferred is repeated once. Every field
 including uniqueid, calldate, billsec, duration, src, dst, channel,
 dstchannel is exactly the same.
 Besides adding a constraint in the database table, isn't there any way I
 can resolve this call transfer cdr duplication issue in asterisk csv cdrs?


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[asterisk-users] Transfer CDRs

[[Digital^Dude] ®]

Hello,

I'm using attended call transfer in asterisk 1.8.11.0 on a CentOS machine.
Each CDR entry of calls that are transferred is repeated once. Every field
including uniqueid, calldate, billsec, duration, src, dst, channel,
dstchannel is exactly the same.
Besides adding a constraint in the database table, isn't there any way I
can resolve this call transfer cdr duplication issue in asterisk csv cdrs?
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Re: [asterisk-users] Transfer to fax

[Warren Selby]

On Tuesday, March 13, 2012, Kevin P. Fleming kpflem...@digium.com wrote:
 On 03/13/2012 05:45 PM, Eric Wieling wrote:

 The faxdetect option is documented in the 1.8 sip.conf.sample.

 Right, I forgot about that. Now I've really confused things.

 /me heads back to his hole


It was actually added to chan sip in 1.6.2, I remember that being a selling
point on a 1.6.2 upgrade for a client of mine about a year and a half ago.

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--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

So I'm still trying to get this to work... (I'm top posting, but the details 
are below, if you want/need background info)

I'd like Asterisk to detect incoming faxes and redirect them elsewhere.  The 
details aren't important, as long as I get the detection working. 

I've added this to my sip.conf file.  Probably overkill, but I'll tune it once 
it works:

[general]
faxdetect=both

My sip registrations are all in a Mysql RT database, so I added this column to 
my table:

faxdetect char(3) default 'no'

I've set faxdetect to 'yes' for the devices that I expect to be receiving fax 
calls.

I did a sip reload from the console after adding and updating this column.

I've added a fax extension to the appropriate context in extensions.conf:
exten = fax,1,noop(I hear a fax!)

Since most of my dialplan is in an AGI script, I've added this to the code 
that handles my test number:

$main::agi-answer();
$main::agi-exec(ringing);
$main::agi-exec(wait,5);


So, now that all of this is in place, I call the extension from my fax 
machine... and I don't get any indication on the console that Asterisk heard a 
fax.  My extension simply rings and I answer it.

What am  missing?

TIA,
Mike Diehl.


On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote:
 On 02/24/2012 05:20 PM, Mike Diehl wrote:
  On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:
  On 02/24/2012 05:00 PM, Mike Diehl wrote:
  On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
  On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
  On 02/24/2012 03:32 PM, Mike Diehl wrote:
  Hi all,
  
  I've got a user that has one phone number an wants to be able to us
  it for both voice and fax.
  
  When a fax call comes in, he wants to do some incantation on the
  keypad and have the call go to the fax machine.
  
  As I see it, he has 3 options.
  
  1.  (blind?) Transfer it to the fax extension.
  
  2.  Use features.conf to create a key sequence, say *2, to
  dial/transfer to a fax extension.
  
  3.  Use fax detect (SIP) to do it automatically.  However I'm also
  using FFA, so I suspect these are mutually exclusive.
  
  They are not. Enabling faxdetect should do exactly what you want; it
  will redirect the call to the 'fax' extension in the current context,
  and you can then Dial() your FAX machine (or send the call to
  ReceiveFAX).
  
  Thank you.  Then, that's what I'll do.
  
  On second though, I think my suggestion that FFA and fax detect were
  mutually exclusive stemmed from the idea that a call that was being
  originated/answered/handled by FFA would have it's call disconnected
  and redirected by fax detect.
  
  If this is the case, it changes my dial plan logic, and I'm not sure I
  fully understand what changes I'll need to make.
  
  For all I know, it might even simplify things by isolating all fax
  handling in one block.
  
  Well, first you should not have faxdetect enabled on outbound channels.
  That takes care of the 'originating' part.
  
  If you have an inbound channel that you *know* you are sending to
  ReceiveFAX, then you can just disable faxdetect on that channel before
  doing so (this is why we made 'faxdetect' configurable from the
  dialplan). Alternatively, you can just let calls that you *know* are
  going to go to ReceiveFAX (dedicated FAX DIDs, for example) just 'idle'
  in the dialplan listening to silence until faxdetect kicks in and sends
  them to ReceiveFAX.
  
  Note that the usage of FFA is not relevant here; whether you are using
  Fax for Asterisk, the free version of it, or res_fax_spandsp, the
  behavior and scenarios would be the same.
  
  Very nice.
  
  Sounds like I need to add a faxdetect column to my SIP real-time
  configuration. Once I've done a sip reload or pruned/loaded my user
  agents, I should be good to go!
  
  Didn't know faxdetect was configurable in the dialplan...  Pointer to how
  to do it?
 
 The CHANNEL() dialplan function with the 'faxdetect' option. Not sure
 which releases have it; it might only be Asterisk 10.

-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Transfer to fax

[Danny Nicholas]

#1 you might need a progress() statement after answer
#2 what does sip show peer xxx look like on this peer?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Tuesday, March 13, 2012 4:18 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Transfer to fax

So I'm still trying to get this to work... (I'm top posting, but the details
are below, if you want/need background info)

I'd like Asterisk to detect incoming faxes and redirect them elsewhere.  The
details aren't important, as long as I get the detection working. 

I've added this to my sip.conf file.  Probably overkill, but I'll tune it
once it works:

[general]
faxdetect=both

My sip registrations are all in a Mysql RT database, so I added this column
to my table:

faxdetect char(3) default 'no'

I've set faxdetect to 'yes' for the devices that I expect to be receiving
fax calls.

I did a sip reload from the console after adding and updating this column.

I've added a fax extension to the appropriate context in extensions.conf:
exten = fax,1,noop(I hear a fax!)

Since most of my dialplan is in an AGI script, I've added this to the code
that handles my test number:

$main::agi-answer();
$main::agi-exec(ringing);
$main::agi-exec(wait,5);


So, now that all of this is in place, I call the extension from my fax
machine... and I don't get any indication on the console that Asterisk heard
a fax.  My extension simply rings and I answer it.

What am  missing?

TIA,
Mike Diehl.


On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote:
 On 02/24/2012 05:20 PM, Mike Diehl wrote:
  On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:
  On 02/24/2012 05:00 PM, Mike Diehl wrote:
  On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
  On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
  On 02/24/2012 03:32 PM, Mike Diehl wrote:
  Hi all,
  
  I've got a user that has one phone number an wants to be able 
  to us it for both voice and fax.
  
  When a fax call comes in, he wants to do some incantation on 
  the keypad and have the call go to the fax machine.
  
  As I see it, he has 3 options.
  
  1.  (blind?) Transfer it to the fax extension.
  
  2.  Use features.conf to create a key sequence, say *2, to 
  dial/transfer to a fax extension.
  
  3.  Use fax detect (SIP) to do it automatically.  However I'm 
  also using FFA, so I suspect these are mutually exclusive.
  
  They are not. Enabling faxdetect should do exactly what you 
  want; it will redirect the call to the 'fax' extension in the 
  current context, and you can then Dial() your FAX machine (or 
  send the call to ReceiveFAX).
  
  Thank you.  Then, that's what I'll do.
  
  On second though, I think my suggestion that FFA and fax detect 
  were mutually exclusive stemmed from the idea that a call that was 
  being originated/answered/handled by FFA would have it's call 
  disconnected and redirected by fax detect.
  
  If this is the case, it changes my dial plan logic, and I'm not 
  sure I fully understand what changes I'll need to make.
  
  For all I know, it might even simplify things by isolating all fax 
  handling in one block.
  
  Well, first you should not have faxdetect enabled on outbound channels.
  That takes care of the 'originating' part.
  
  If you have an inbound channel that you *know* you are sending to 
  ReceiveFAX, then you can just disable faxdetect on that channel 
  before doing so (this is why we made 'faxdetect' configurable from 
  the dialplan). Alternatively, you can just let calls that you 
  *know* are going to go to ReceiveFAX (dedicated FAX DIDs, for example)
just 'idle'
  in the dialplan listening to silence until faxdetect kicks in and 
  sends them to ReceiveFAX.
  
  Note that the usage of FFA is not relevant here; whether you are 
  using Fax for Asterisk, the free version of it, or res_fax_spandsp, 
  the behavior and scenarios would be the same.
  
  Very nice.
  
  Sounds like I need to add a faxdetect column to my SIP real-time 
  configuration. Once I've done a sip reload or pruned/loaded my user 
  agents, I should be good to go!
  
  Didn't know faxdetect was configurable in the dialplan...  Pointer 
  to how to do it?
 
 The CHANNEL() dialplan function with the 'faxdetect' option. Not sure 
 which releases have it; it might only be Asterisk 10.

-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Tuesday 13 March 2012 3:21:58 pm Danny Nicholas wrote:
 #1 you might need a progress() statement after answer

I'll try that.  Thank you.

 #2 what does sip show peer xxx look like on this peer?

I'm testing against my office phone, a Polycom 501:

  * Name   : 0004F211F1D0-2
  Realtime peer: Yes, cached
  Secret   : Set
  MD5Secret: Not set
  Remote Secret: Not set
  Context  : customers
  Subscr.Cont. : Not set
  Language : 
  Accountcode  : 1
  AMA flags: Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup: 
  Pickupgroup  : 
  Mailbox  : 7001@context
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic  : Yes
  Callerid : Mike Diehl 5051234567
  MaxCallBR: 384 kbps
  Expire   : 172
  Insecure : no
  Nat  : Always
  ACL  : Yes
  T.38 support : Yes
  T.38 EC mode : FEC
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: Yes
  Text Support : No
  Ign SDP ver  : Yes
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode : rfc2833
  Timer T1 : 500
  Timer B  : 32000
  ToHost   : 
  Addr-IP : 173.10.242.192 Port 1811
  Defaddr-IP  : 0.0.0.0 Port 5060
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 0004F211F1D0-2
  SIP Options  : (none)
  Codecs   : 0x4 (ulaw)
  Codec Order  : (ulaw:20)
  Auto-Framing :  No 
  100 on REG   : Yes
  Status   : OK (88 ms)
  Useragent: PolycomSoundPointIP-SPIP_501-UA/3.1.4.0070
  Reg. Contact : sip:0004F211F1D0-2@10.0.1.81
  Qualify Freq : 6 ms
  Variables:
 line_id = 0004F211F1D0-2
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess : 90 secs
  Parkinglot   : 



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
 Sent: Tuesday, March 13, 2012 4:18 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Transfer to fax
 
 So I'm still trying to get this to work... (I'm top posting, but the
 details are below, if you want/need background info)
 
 I'd like Asterisk to detect incoming faxes and redirect them elsewhere. 
 The details aren't important, as long as I get the detection working.
 
 I've added this to my sip.conf file.  Probably overkill, but I'll tune it
 once it works:
 
 [general]
 faxdetect=both
 
 My sip registrations are all in a Mysql RT database, so I added this column
 to my table:
 
 faxdetect char(3) default 'no'
 
 I've set faxdetect to 'yes' for the devices that I expect to be receiving
 fax calls.
 
 I did a sip reload from the console after adding and updating this column.
 
 I've added a fax extension to the appropriate context in extensions.conf:
 exten = fax,1,noop(I hear a fax!)
 
 Since most of my dialplan is in an AGI script, I've added this to the code
 that handles my test number:
 
 $main::agi-answer();
 $main::agi-exec(ringing);
 $main::agi-exec(wait,5);
 
 
 So, now that all of this is in place, I call the extension from my fax
 machine... and I don't get any indication on the console that Asterisk
 heard a fax.  My extension simply rings and I answer it.
 
 What am  missing?
 
 TIA,
 Mike Diehl.
 
 On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote:
  On 02/24/2012 05:20 PM, Mike Diehl wrote:
   On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:
   On 02/24/2012 05:00 PM, Mike Diehl wrote:
   On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
   On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
   On 02/24/2012 03:32 PM, Mike Diehl wrote:
   Hi all,
   
   I've got a user that has one phone number an wants to be able
   to us it for both voice and fax.
   
   When a fax call comes in, he wants to do some incantation on
   the keypad and have the call go to the fax machine.
   
   As I see it, he has 3 options.
   
   1.  (blind?) Transfer it to the fax extension.
   
   2.  Use features.conf to create a key sequence, say *2, to
   dial/transfer to a fax extension.
   
   3.  Use fax detect (SIP) to do it automatically.  However I'm
   also using FFA, so I suspect these are mutually exclusive.
   
   They are not. Enabling faxdetect should do exactly what you
   want; it will redirect the call to the 'fax' extension in the
   current context, and you can then Dial() your FAX machine (or
   send the call to ReceiveFAX).
   
   Thank you.  Then, that's what I'll do.
   
   On second though, I think my suggestion that FFA and fax detect
   were mutually exclusive stemmed from the idea that a call that was
   being originated/answered/handled by FFA would have it's call
   disconnected and redirected by fax detect.
   
   If this is the case, it changes my dial plan logic, and I'm not
   sure I fully understand what changes I'll need to make.
   
   For all I know, it might

Re: [asterisk-users] Transfer to fax

[Kevin P. Fleming]

On 03/13/2012 04:18 PM, Mike Diehl wrote:

So I'm still trying to get this to work... (I'm top posting, but the details
are below, if you want/need background info)

I'd like Asterisk to detect incoming faxes and redirect them elsewhere.  The
details aren't important, as long as I get the detection working.

I've added this to my sip.conf file.  Probably overkill, but I'll tune it once
it works:

[general]
faxdetect=both


This will have no effect; see below.


My sip registrations are all in a Mysql RT database, so I added this column to
my table:

faxdetect char(3) default 'no'

I've set faxdetect to 'yes' for the devices that I expect to be receiving fax
calls.


'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi). 
It's a feature that can be enabled on a channel via the CHANNEL() 
dialplan function. In the dialplan itself, you'd use something like this:


exten = 1234,5,Set(CHANNEL(faxdetect)=yes)

To do this in a configuration file, so that it will be applied to 
channels as soon as they are created, use 'setvar':


[peer1]
setvar=CHANNEL(faxdetect)=yes

I'm not sure how this would be done using Realtime configuration, but it 
should be possible. I'd encourage you to test it out using a 
non-Realtime peer first, just to make sure that it works the way you expect.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote:
 On 03/13/2012 04:18 PM, Mike Diehl wrote:

  I've set faxdetect to 'yes' for the devices that I expect to be receiving
  fax calls.
 
 'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi).
 It's a feature that can be enabled on a channel via the CHANNEL()
 dialplan function. In the dialplan itself, you'd use something like this:
 exten = 1234,5,Set(CHANNEL(faxdetect)=yes)

This function was implemented somewhere in the 10.x code base, I believe.  I'm 
running 1.6.x.  So, it sounds like I need to plan an upgrade in order to get 
this to work.

 To do this in a configuration file, so that it will be applied to
 channels as soon as they are created, use 'setvar':
 
 [peer1]
 setvar=CHANNEL(faxdetect)=yes
 
 I'm not sure how this would be done using Realtime configuration, but it
 should be possible. I'd encourage you to test it out using a
 non-Realtime peer first, just to make sure that it works the way you
 expect.

I've used setvar in my RT config and it works very well.

WRT the upgrade, I've gone in and made some code changes to the voicemail 
module which I'll have to port over to version 10.x.  Sounds like I should 
sign up to be a developer so I can pass those patches on...

-- 

Take care and have fun,
Mike Diehl.

--
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Re: [asterisk-users] Transfer to fax

[Larry Moore]

On 14/03/2012 5:18 AM, Mike Diehl wrote:

So I'm still trying to get this to work... (I'm top posting, but the details
are below, if you want/need background info)

I'd like Asterisk to detect incoming faxes and redirect them elsewhere.  The
details aren't important, as long as I get the detection working.

I've added this to my sip.conf file.  Probably overkill, but I'll tune it once
it works:

[general]
faxdetect=both

My sip registrations are all in a Mysql RT database, so I added this column to
my table:

faxdetect char(3) default 'no'

I've set faxdetect to 'yes' for the devices that I expect to be receiving fax
calls.

I did a sip reload from the console after adding and updating this column.

I've added a fax extension to the appropriate context in extensions.conf:
exten =  fax,1,noop(I hear a fax!)

Since most of my dialplan is in an AGI script, I've added this to the code
that handles my test number:

$main::agi-answer();
$main::agi-exec(ringing);
$main::agi-exec(wait,5);


So, now that all of this is in place, I call the extension from my fax
machine... and I don't get any indication on the console that Asterisk heard a
fax.  My extension simply rings and I answer it.

What am  missing?



In your peer config set directmedia=no and faxdetect=cng, Asterisk needs 
to be in the path to hear the CNG tones.


Larry.

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Re: [asterisk-users] Transfer to fax

[Kevin P. Fleming]

On 03/13/2012 04:56 PM, Mike Diehl wrote:

On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote:

On 03/13/2012 04:18 PM, Mike Diehl wrote:



I've set faxdetect to 'yes' for the devices that I expect to be receiving
fax calls.


'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi).
It's a feature that can be enabled on a channel via the CHANNEL()
dialplan function. In the dialplan itself, you'd use something like this:
exten =  1234,5,Set(CHANNEL(faxdetect)=yes)


This function was implemented somewhere in the 10.x code base, I believe.  I'm
running 1.6.x.  So, it sounds like I need to plan an upgrade in order to get
this to work.


Right, so prior to that version, the *only* channel driver that had 
'faxdetect' functionality was chan_dahdi.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Tuesday 13 March 2012 4:04:31 pm Kevin P. Fleming wrote:
 On 03/13/2012 04:56 PM, Mike Diehl wrote:
  On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote:
  On 03/13/2012 04:18 PM, Mike Diehl wrote:
  I've set faxdetect to 'yes' for the devices that I expect to be
  receiving fax calls.
  
  'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi).
  It's a feature that can be enabled on a channel via the CHANNEL()
  dialplan function. In the dialplan itself, you'd use something like
  this: exten =  1234,5,Set(CHANNEL(faxdetect)=yes)
  
  This function was implemented somewhere in the 10.x code base, I believe.
   I'm running 1.6.x.  So, it sounds like I need to plan an upgrade in
  order to get this to work.
 
 Right, so prior to that version, the *only* channel driver that had
 'faxdetect' functionality was chan_dahdi.

So, I have a few long nights ahead of me!

Thanks for your time.

-- 

Take care and have fun,
Mike Diehl.

--
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Re: [asterisk-users] Transfer to fax

[Eric Wieling]

The faxdetect option is documented in the 1.8 sip.conf.sample.  

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Tuesday, March 13, 2012 6:17 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Transfer to fax

On Tuesday 13 March 2012 4:04:31 pm Kevin P. Fleming wrote:
 On 03/13/2012 04:56 PM, Mike Diehl wrote:
  On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote:
  On 03/13/2012 04:18 PM, Mike Diehl wrote:
  I've set faxdetect to 'yes' for the devices that I expect to be 
  receiving fax calls.
  
  'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi).
  It's a feature that can be enabled on a channel via the CHANNEL() 
  dialplan function. In the dialplan itself, you'd use something like
  this: exten =  1234,5,Set(CHANNEL(faxdetect)=yes)
  
  This function was implemented somewhere in the 10.x code base, I believe.
   I'm running 1.6.x.  So, it sounds like I need to plan an upgrade in 
  order to get this to work.
 
 Right, so prior to that version, the *only* channel driver that had 
 'faxdetect' functionality was chan_dahdi.

So, I have a few long nights ahead of me!

Thanks for your time.

-- 

Take care and have fun,
Mike Diehl.

--
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Re: [asterisk-users] Transfer to fax

[Kevin P. Fleming]

On 03/13/2012 05:45 PM, Eric Wieling wrote:

The faxdetect option is documented in the 1.8 sip.conf.sample.


Right, I forgot about that. Now I've really confused things.

/me heads back to his hole

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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[asterisk-users] Transfer to fax

[Mike Diehl]

Hi all,

I've got a user that has one phone number an wants to be able to us it for 
both voice and fax.

When a fax call comes in, he wants to do some incantation on the keypad and 
have the call go to the fax machine.

As I see it, he has 3 options.

1.  (blind?) Transfer it to the fax extension.  

2.  Use features.conf to create a key sequence, say *2, to dial/transfer to a 
fax extension.

3.  Use fax detect (SIP) to do it automatically.  However I'm also using FFA, 
so I suspect these are mutually exclusive.

Does anyone have any insight as to which would work better?

TIA,

-- 

Take care and have fun,
Mike Diehl.

--
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Re: [asterisk-users] Transfer to fax

[Danny Nicholas]

Why not use AMD (Answering Machine Detect)?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Friday, February 24, 2012 3:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Transfer to fax

Hi all,

I've got a user that has one phone number an wants to be able to us it for
both voice and fax.

When a fax call comes in, he wants to do some incantation on the keypad and
have the call go to the fax machine.

As I see it, he has 3 options.

1.  (blind?) Transfer it to the fax extension.  

2.  Use features.conf to create a key sequence, say *2, to dial/transfer to
a fax extension.

3.  Use fax detect (SIP) to do it automatically.  However I'm also using
FFA, so I suspect these are mutually exclusive.

Does anyone have any insight as to which would work better?

TIA,

-- 

Take care and have fun,
Mike Diehl.

--
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Re: [asterisk-users] Transfer to fax

[Kevin P. Fleming]

On 02/24/2012 03:32 PM, Mike Diehl wrote:

Hi all,

I've got a user that has one phone number an wants to be able to us it for
both voice and fax.

When a fax call comes in, he wants to do some incantation on the keypad and
have the call go to the fax machine.

As I see it, he has 3 options.

1.  (blind?) Transfer it to the fax extension.

2.  Use features.conf to create a key sequence, say *2, to dial/transfer to a
fax extension.

3.  Use fax detect (SIP) to do it automatically.  However I'm also using FFA,
so I suspect these are mutually exclusive.


They are not. Enabling faxdetect should do exactly what you want; it 
will redirect the call to the 'fax' extension in the current context, 
and you can then Dial() your FAX machine (or send the call to ReceiveFAX).


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
 On 02/24/2012 03:32 PM, Mike Diehl wrote:
  Hi all,
  
  I've got a user that has one phone number an wants to be able to us it
  for both voice and fax.
  
  When a fax call comes in, he wants to do some incantation on the keypad
  and have the call go to the fax machine.
  
  As I see it, he has 3 options.
  
  1.  (blind?) Transfer it to the fax extension.
  
  2.  Use features.conf to create a key sequence, say *2, to dial/transfer
  to a fax extension.
  
  3.  Use fax detect (SIP) to do it automatically.  However I'm also using
  FFA, so I suspect these are mutually exclusive.
 
 They are not. Enabling faxdetect should do exactly what you want; it
 will redirect the call to the 'fax' extension in the current context,
 and you can then Dial() your FAX machine (or send the call to ReceiveFAX).

Thank you.  Then, that's what I'll do.

-- 

Take care and have fun,
Mike Diehl.

--
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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
 On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
  On 02/24/2012 03:32 PM, Mike Diehl wrote:
   Hi all,
   
   I've got a user that has one phone number an wants to be able to us it
   for both voice and fax.
   
   When a fax call comes in, he wants to do some incantation on the keypad
   and have the call go to the fax machine.
   
   As I see it, he has 3 options.
   
   1.  (blind?) Transfer it to the fax extension.
   
   2.  Use features.conf to create a key sequence, say *2, to
   dial/transfer to a fax extension.
   
   3.  Use fax detect (SIP) to do it automatically.  However I'm also
   using FFA, so I suspect these are mutually exclusive.
  
  They are not. Enabling faxdetect should do exactly what you want; it
  will redirect the call to the 'fax' extension in the current context,
  and you can then Dial() your FAX machine (or send the call to
  ReceiveFAX).
 
 Thank you.  Then, that's what I'll do.

On second though, I think my suggestion that FFA and fax detect were mutually 
exclusive stemmed from the idea that a call that was being 
originated/answered/handled by FFA would have it's call disconnected and 
redirected by fax detect.

If this is the case, it changes my dial plan logic, and I'm not sure I fully 
understand what changes I'll need to make.

For all I know, it might even simplify things by isolating all fax handling in 
one block.

Comments?

-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Transfer to fax

[Kevin P. Fleming]

On 02/24/2012 05:00 PM, Mike Diehl wrote:

On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:

On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:

On 02/24/2012 03:32 PM, Mike Diehl wrote:

Hi all,

I've got a user that has one phone number an wants to be able to us it
for both voice and fax.

When a fax call comes in, he wants to do some incantation on the keypad
and have the call go to the fax machine.

As I see it, he has 3 options.

1.  (blind?) Transfer it to the fax extension.

2.  Use features.conf to create a key sequence, say *2, to
dial/transfer to a fax extension.

3.  Use fax detect (SIP) to do it automatically.  However I'm also
using FFA, so I suspect these are mutually exclusive.


They are not. Enabling faxdetect should do exactly what you want; it
will redirect the call to the 'fax' extension in the current context,
and you can then Dial() your FAX machine (or send the call to
ReceiveFAX).


Thank you.  Then, that's what I'll do.


On second though, I think my suggestion that FFA and fax detect were mutually
exclusive stemmed from the idea that a call that was being
originated/answered/handled by FFA would have it's call disconnected and
redirected by fax detect.

If this is the case, it changes my dial plan logic, and I'm not sure I fully
understand what changes I'll need to make.

For all I know, it might even simplify things by isolating all fax handling in
one block.


Well, first you should not have faxdetect enabled on outbound channels. 
That takes care of the 'originating' part.


If you have an inbound channel that you *know* you are sending to 
ReceiveFAX, then you can just disable faxdetect on that channel before 
doing so (this is why we made 'faxdetect' configurable from the 
dialplan). Alternatively, you can just let calls that you *know* are 
going to go to ReceiveFAX (dedicated FAX DIDs, for example) just 'idle' 
in the dialplan listening to silence until faxdetect kicks in and sends 
them to ReceiveFAX.


Note that the usage of FFA is not relevant here; whether you are using 
Fax for Asterisk, the free version of it, or res_fax_spandsp, the 
behavior and scenarios would be the same.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:
 On 02/24/2012 05:00 PM, Mike Diehl wrote:
  On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
  On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
  On 02/24/2012 03:32 PM, Mike Diehl wrote:
  Hi all,
  
  I've got a user that has one phone number an wants to be able to us it
  for both voice and fax.
  
  When a fax call comes in, he wants to do some incantation on the
  keypad and have the call go to the fax machine.
  
  As I see it, he has 3 options.
  
  1.  (blind?) Transfer it to the fax extension.
  
  2.  Use features.conf to create a key sequence, say *2, to
  dial/transfer to a fax extension.
  
  3.  Use fax detect (SIP) to do it automatically.  However I'm also
  using FFA, so I suspect these are mutually exclusive.
  
  They are not. Enabling faxdetect should do exactly what you want; it
  will redirect the call to the 'fax' extension in the current context,
  and you can then Dial() your FAX machine (or send the call to
  ReceiveFAX).
  
  Thank you.  Then, that's what I'll do.
  
  On second though, I think my suggestion that FFA and fax detect were
  mutually exclusive stemmed from the idea that a call that was being
  originated/answered/handled by FFA would have it's call disconnected and
  redirected by fax detect.
  
  If this is the case, it changes my dial plan logic, and I'm not sure I
  fully understand what changes I'll need to make.
  
  For all I know, it might even simplify things by isolating all fax
  handling in one block.
 
 Well, first you should not have faxdetect enabled on outbound channels.
 That takes care of the 'originating' part.
 
 If you have an inbound channel that you *know* you are sending to
 ReceiveFAX, then you can just disable faxdetect on that channel before
 doing so (this is why we made 'faxdetect' configurable from the
 dialplan). Alternatively, you can just let calls that you *know* are
 going to go to ReceiveFAX (dedicated FAX DIDs, for example) just 'idle'
 in the dialplan listening to silence until faxdetect kicks in and sends
 them to ReceiveFAX.
 
 Note that the usage of FFA is not relevant here; whether you are using
 Fax for Asterisk, the free version of it, or res_fax_spandsp, the
 behavior and scenarios would be the same.

Very nice.  

Sounds like I need to add a faxdetect column to my SIP real-time configuration. 
 
Once I've done a sip reload or pruned/loaded my user agents, I should be good 
to go!

Didn't know faxdetect was configurable in the dialplan...  Pointer to how to do 
it?

BTW, FFA rocks!



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Re: [asterisk-users] Transfer to fax

[Kevin P. Fleming]

On 02/24/2012 05:20 PM, Mike Diehl wrote:

On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:

On 02/24/2012 05:00 PM, Mike Diehl wrote:

On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:

On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:

On 02/24/2012 03:32 PM, Mike Diehl wrote:

Hi all,

I've got a user that has one phone number an wants to be able to us it
for both voice and fax.

When a fax call comes in, he wants to do some incantation on the
keypad and have the call go to the fax machine.

As I see it, he has 3 options.

1.  (blind?) Transfer it to the fax extension.

2.  Use features.conf to create a key sequence, say *2, to
dial/transfer to a fax extension.

3.  Use fax detect (SIP) to do it automatically.  However I'm also
using FFA, so I suspect these are mutually exclusive.


They are not. Enabling faxdetect should do exactly what you want; it
will redirect the call to the 'fax' extension in the current context,
and you can then Dial() your FAX machine (or send the call to
ReceiveFAX).


Thank you.  Then, that's what I'll do.


On second though, I think my suggestion that FFA and fax detect were
mutually exclusive stemmed from the idea that a call that was being
originated/answered/handled by FFA would have it's call disconnected and
redirected by fax detect.

If this is the case, it changes my dial plan logic, and I'm not sure I
fully understand what changes I'll need to make.

For all I know, it might even simplify things by isolating all fax
handling in one block.


Well, first you should not have faxdetect enabled on outbound channels.
That takes care of the 'originating' part.

If you have an inbound channel that you *know* you are sending to
ReceiveFAX, then you can just disable faxdetect on that channel before
doing so (this is why we made 'faxdetect' configurable from the
dialplan). Alternatively, you can just let calls that you *know* are
going to go to ReceiveFAX (dedicated FAX DIDs, for example) just 'idle'
in the dialplan listening to silence until faxdetect kicks in and sends
them to ReceiveFAX.

Note that the usage of FFA is not relevant here; whether you are using
Fax for Asterisk, the free version of it, or res_fax_spandsp, the
behavior and scenarios would be the same.


Very nice.

Sounds like I need to add a faxdetect column to my SIP real-time configuration.
Once I've done a sip reload or pruned/loaded my user agents, I should be good
to go!

Didn't know faxdetect was configurable in the dialplan...  Pointer to how to do
it?


The CHANNEL() dialplan function with the 'faxdetect' option. Not sure 
which releases have it; it might only be Asterisk 10.


--
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Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote:
 On 02/24/2012 05:20 PM, Mike Diehl wrote:
  On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:
  On 02/24/2012 05:00 PM, Mike Diehl wrote:
  On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
  On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
  On 02/24/2012 03:32 PM, Mike Diehl wrote:
  Hi all,
  
  I've got a user that has one phone number an wants to be able to us
  it for both voice and fax.
  
  When a fax call comes in, he wants to do some incantation on the
  keypad and have the call go to the fax machine.
  
  As I see it, he has 3 options.
  
  1.  (blind?) Transfer it to the fax extension.
  
  2.  Use features.conf to create a key sequence, say *2, to
  dial/transfer to a fax extension.
  
  3.  Use fax detect (SIP) to do it automatically.  However I'm also
  using FFA, so I suspect these are mutually exclusive.
  
  They are not. Enabling faxdetect should do exactly what you want; it
  will redirect the call to the 'fax' extension in the current context,
  and you can then Dial() your FAX machine (or send the call to
  ReceiveFAX).
  
  Thank you.  Then, that's what I'll do.
  
  On second though, I think my suggestion that FFA and fax detect were
  mutually exclusive stemmed from the idea that a call that was being
  originated/answered/handled by FFA would have it's call disconnected
  and redirected by fax detect.
  
  If this is the case, it changes my dial plan logic, and I'm not sure I
  fully understand what changes I'll need to make.
  
  For all I know, it might even simplify things by isolating all fax
  handling in one block.
  
  Well, first you should not have faxdetect enabled on outbound channels.
  That takes care of the 'originating' part.
  
  If you have an inbound channel that you *know* you are sending to
  ReceiveFAX, then you can just disable faxdetect on that channel before
  doing so (this is why we made 'faxdetect' configurable from the
  dialplan). Alternatively, you can just let calls that you *know* are
  going to go to ReceiveFAX (dedicated FAX DIDs, for example) just 'idle'
  in the dialplan listening to silence until faxdetect kicks in and sends
  them to ReceiveFAX.
  
  Note that the usage of FFA is not relevant here; whether you are using
  Fax for Asterisk, the free version of it, or res_fax_spandsp, the
  behavior and scenarios would be the same.
  
  Very nice.
  
  Sounds like I need to add a faxdetect column to my SIP real-time
  configuration. Once I've done a sip reload or pruned/loaded my user
  agents, I should be good to go!
  
  Didn't know faxdetect was configurable in the dialplan...  Pointer to how
  to do it?
 
 The CHANNEL() dialplan function with the 'faxdetect' option. Not sure
 which releases have it; it might only be Asterisk 10.

Version 10 is on my radar, but I'm using 1.6.3 now. I'll do some research.  If 
it's not in 1.6, I'll just try to make this work in the sip config, for now.

Thanks for your time.

-- 

Take care and have fun,
Mike Diehl.

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[asterisk-users] Transfer to VoiceMail Asterisk 1.6

[motty.cruz]

Hello,
I'm using Asterisk 1.6 with Polycom SoundPoint 650, everything is running
fine except that I can't program a button on Polycom to transfer inbound
call to Voicemail directly. 

I have the following in my extension.conf 

exten = _547xx,1,Voicemail(${EXTEN:1}@default,u)

Reception can transfer directly to VoiceMail when dialing digit 5 I want to
make a softkey on Polycom 650 does anybody know how to accomplish tranfering
directly to VoiceMail? 

Thanks, 
Motty


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[asterisk-users] Transfer beep w/ Polycom phone

[Mike Diehl]

Hi all.

When a user transfers a call by pressing the transfer soft button on their 
phone, I'd like it to beep at them when the transfer is complete.  I've got 
it turned on in features.conf:

xfersound = beep   ; to indicate an attended transfer is complete 
xferfailsound = beeperr; to indicate a failed transfer 

However, it seems that transfer is a function of the phone, not Asterisk.  Is 
there any way I can configure the Polycom phones to either use the Asterisk 
function, or to make a beep when a transfer completes?

TIA,

-- 

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Re: [asterisk-users] Transfer beep w/ Polycom phone

[Danny Nicholas]

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Mike Diehl
 Sent: Monday, April 25, 2011 4:30 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Transfer beep w/ Polycom phone
 
 Hi all.
 
 When a user transfers a call by pressing the transfer soft button on
 their
 phone, I'd like it to beep at them when the transfer is complete.  I've
 got
 it turned on in features.conf:
 
 xfersound = beep   ; to indicate an attended transfer is
 complete
 xferfailsound = beeperr; to indicate a failed transfer
 
 However, it seems that transfer is a function of the phone, not Asterisk.
 Is
 there any way I can configure the Polycom phones to either use the
 Asterisk
 function, or to make a beep when a transfer completes?
 
 TIA,
 
 --
 
 Take care and have fun,
 Mike Diehl.
[Danny Nicholas] 
The Polycom soft button can be a bear.  I'd recommend looking at sip debug
on the transfers and modifying features.conf accordingly since I'll bet your
users would NOT like to lose their soft transfer key.


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[asterisk-users] Transfer feature dialing out after one digit

[Hose]

Because some users have requested transfer beep confirmations I've
switched our phones over to using the asterisk transfer feature instead
of the built in transfer functions of the phones.  While testing it was
working fine, but I changed something in features.conf and suddenly any
time I hit transfer (*2), I can only enter one digit before asterisk
immediately tries to dial that extension.  

Features.conf is extremely barebones so I'm not sure where any problems
would come up:

[general]
transferdigittimeout = 3 
xfersound = beep

[featuremap]
blindxfer = #
atxfer = *2 

Tried change transferdigittimeout to something insanely long like 100,
but still the same thing.  I even fully restarted asterisk (just in case
it was some weird fluke that crept in).  

This is 1.6.2.17.2.  Someone else posted something extremely similar,
though it's in version 1.8.2.3:

http://forums.digium.com/viewtopic.php?f=1t=77154


hose

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Re: [asterisk-users] Transfer feature dialing out after one digit

[Hose]

What you say...Hose (hose+aster...@bluemaggottowel.com):

 Because some users have requested transfer beep confirmations I've
 switched our phones over to using the asterisk transfer feature instead
 of the built in transfer functions of the phones.  While testing it was
 working fine, but I changed something in features.conf and suddenly any
 time I hit transfer (*2), I can only enter one digit before asterisk
 immediately tries to dial that extension.  
 
 Features.conf is extremely barebones so I'm not sure where any problems
 would come up:
 
 [general]
 transferdigittimeout = 3 
 xfersound = beep
 
 [featuremap]
 blindxfer = #
 atxfer = *2 
 
 Tried change transferdigittimeout to something insanely long like 100,
 but still the same thing.  I even fully restarted asterisk (just in case
 it was some weird fluke that crept in).  
 
 This is 1.6.2.17.2.  Someone else posted something extremely similar,
 though it's in version 1.8.2.3:
 
 http://forums.digium.com/viewtopic.php?f=1t=77154
 
 hose

Looks like this might be an asterisk bug with the dial command,
specifically, the T option, allowing the calling user to execute
transfers.  The lowercase t option (allowing the *called* user to
execute transfers) works fine in all cases; the erroneous behavior of
having the calling user attempt a transfer but having it registered way
before the featureddigittimeout value expires happens whether the
lowercase t option is enabled or not.

Can anyone confirm or refute with their version?  You'd just have to add
the T option to a dial command, call / answer an extension, and then try and
execute a feature transfer from the calling extension.

hose

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[asterisk-users] Transfer Device Data

[Elliot Murdock]

Hello!

I am trying to find out the device name and/or other identifying data
to be used in a context when a device transfers the call to new a
phone number.  From running tests, it looks like the account code
variable (${CDR(accountcode)}) is set to the account code of the
device that placed the original call, so if the callee device (not the
original calling device) is making the transfer to a new number, the
account code will not be correct, since it will be the account code of
the calling device, but not the called device.

How do I find out which device is making the transfer?

Thanks for any suggestions!
Elliot

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Re: [asterisk-users] Transfer Device Data

[C F]

${BLINDTRANSFER} should hold the device name of the one doing the
blind transfer.


On Sat, Feb 12, 2011 at 6:06 PM, Elliot Murdock murdo...@gmail.com wrote:
 Hello!

 I am trying to find out the device name and/or other identifying data
 to be used in a context when a device transfers the call to new a
 phone number.  From running tests, it looks like the account code
 variable (${CDR(accountcode)}) is set to the account code of the
 device that placed the original call, so if the callee device (not the
 original calling device) is making the transfer to a new number, the
 account code will not be correct, since it will be the account code of
 the calling device, but not the called device.

 How do I find out which device is making the transfer?

 Thanks for any suggestions!
 Elliot

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Re: [asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target

[Doug Lytle]

John Reynolds wrote:
Has anyone seen or heard of this? Know how to resolve to expected 
behavior?  I appreciate any pointers. 


John,

Without seeing any of your dial plan or any of the output from your 
console during the failed transfer, nobody is going to be able to help.


Why don't you start by posting the relevant part of your code that does 
the dialing and shows up the console output during a test transfer?


Doug


--
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Re: [asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target

[John Reynolds]

On Sat, Dec 18, 2010 at 6:51 AM, Doug Lytle supp...@drdos.info wrote:

 John Reynolds wrote:

 Has anyone seen or heard of this? Know how to resolve to expected
 behavior?  I appreciate any pointers.


 John,

 Without seeing any of your dial plan or any of the output from your console
 during the failed transfer, nobody is going to be able to help.

 Why don't you start by posting the relevant part of your code that does the
 dialing and shows up the console output during a test transfer?

 Doug



Doug and Darrick,

Thanks, I work on this a bit more and get back with more info.

John R.
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[asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target

[John Reynolds]

The setup is this:
2 sip handsets (a Cisco 7960 and a 7961) exten 401/402
1 fxs/dahdi cordless phone, exten 201
rhino fxo/fxs analog card
asterisk 1.4.31

This is running on a Soekris 5501 with Astlinux 0.7.2

While I do have FXO capabilities, no POTS lines are connected.

When a call comes in (VoIP, either SIP or IAX) it is usually answered on one
of the SIP Cisco phones(x 401 or 402). If it is for my wife, then I would
like to transfer the call to the fxs/dahdi analog cordless phone (x 201). At
one time this worked, but about a year or so ago it stopped.

What is happening now is that the call comes in (x 401), is transferred via
the cisco transfer soft button to (x 201), ... during this time the caller
was put on hold or rather was automatically connected to the MOH process...
, When (x 201) answers the phone, they are connected to the MOH process and
cannot hear or talk to the original caller.

In testing, if I leave the (x 201) call open, the original outside call is
kept open as well (the original caller hears nothing). A look at the active
sessions confirms this. When either (x 201) or original caller hang up, the
call/connection is terminated.

I can transfer calls from one Cisco to the other without issue.

I have looked around at my configs, but don't see anything that would cause
this... but truthfully I don't even know where to begin with something like
this. I checked the logs to see if there was something helpful there but did
not see anything. My only though is that it is something with the way the
Cisco internal transfer process happens... but again, I don't know where to
begin to test that theory.

Has anyone seen or heard of this? Know how to resolve to expected behavior?
I appreciate any pointers.

John R.
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[asterisk-users] Transfer (sip - dahdi) results in moh for dahdi

[John Reynolds]

I have had this problem for a while, so I can't be sure when it started or
what was changed.

The setup is this:
2 sip handsets (a Cisco 7960 and a 7961) exten 401/402
1 fxs/dahdi cordless phone, exten 201
rhino fxo/fxs analog card
asterisk 1.4.31

This is running on a Soekris 5501 with Astlinux 0.7.2

While I do have FXO capabilities, no POTS lines are connected.

When a call comes in (VoIP, either SIP or IAX) it is usually answered on one
of the SIP Cisco phones(x 401 or 402). If it is for my wife, which it
usually is, and she is walking around the house, then I would like to
transfer the call to the fxs/dahdi analog cordless phone (x 201). At one
time this worked, but about a year or so ago it stopped.

What is happening now is that the call comes in (x 401), is transferred via
the cisco transfer soft button to (x 201), ... during this time the caller
was put on hold or rather was automatically connected to the MOH process...
, When (x 201) answers the phone, they are connected to the MOH process and
cannot hear or talk to the original caller.

In testing, if I leave the (x 201) call open, the original outside call is
kept open as well. A look at the active sessions confirms this. When
either (x 201) or original caller hang up, the call/connection is
terminated.

I can transfer calls from one Cisco to the other without issue; and if my
laptop, with the softphone installed, had not just taken a turn for the
worst, I would test Cisco to Bria/Counterpath and let you know how that
would work.

I have looked around at my configs, but don't see anything that would cause
this... but truthfully I don't even know where to begin with something like
this. I checked the logs to see if there was something helpful there but did
not see anything. My only though is that it is something with the way the
Cisco internal transfer process happens... but again, I don't know where to
begin to test that theory.

Has anyone seen or heard of this? Know how to resolve to expected behavior?
I appreciate any pointers.

John R.
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[asterisk-users] Transfer + speed dial button problem?

[Gerard]

  Hi everyone,
I'm having a bit of an issue after upgrading from asterisk ~1.2.24 to 
1.6.2.11,
with the old version when the user would go to transfer a call, they 
would press Transfer, then the speed dial button for the extension, 
optionally introduce the call, and then press Transfer again to complete 
the transfer.
Now, with the new version, when you hit the speed dial button, asterisk 
treats it as a new call rather then the input for the transfer, I'm not 
sure if it's a configuration option, or something to do with chan_sccp.. 
(i.e. I should ask on the sccp list instead)
I'm using Cisco 7960 phones with the 7914 12 button add-on. Transferring 
works absolutely fine if you just type the extension number rather then 
using the speed dial button.

I'm running:
Asterisk 1.6.2.11 built by root @ phonesys2 on a i686 running Linux on 
2010-08-22 16:52:42 UTC
and
SCCP channel Release: v2 - 1792 (built by 'root' on 'Mon Aug 23 17:42:15 
CDT 2010')
chan_sccp v3 crashed too much to be useful, so I went back to v2 for now.

Any input would be appreciated!
Thanks,
-Gerard Saraber
gsara...@rarcoa.com


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[asterisk-users] Transfer to non registered extension creates call hangup

[Rushikesh]

Hi list,


I have a small problem happening due to call transfer.

Whenever the call gets transfered to a remote extension ( which is not 
registered to asterisk ) it results in hangup().

When asterisk tries to dial the other extension it results in failure 
making the call cut down :(

Is there any way by which I can send the call back to the person who 
transfered it?


Regards,
Rushikesh

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[asterisk-users] Transfer fails

[Jonas Kellens]

Hello list,

this is the dialplan :

snip
exten = s,n,Dial(SIP/test1SIP/test2,,t)
snip

exten = 10,1,Dial(SIP/test1)
exten = 20,1,Dial(SIP/test2)


So there is an incoming call that rings SIPaccounts test1 and test2.
Account test1 answers and wants to transfer the call to test2.
Transfer is : #20

This is what the CLI shows :

[Jul  2 10:55:30] -- Executing [...@from-test:1] 
Dial(SIP/test1-010e, SIP/test2) in new stack
[Jul  2 10:55:30] WARNING[7604]: app_dial.c:1296 dial_exec_full: Unable 
to create channel of type 'SIP' (cause 20 - Unknown)

[Jul  2 10:55:30]   == Everyone is busy/congested at this time (1:0/0/1)

...and the call is disconnected.

When I call the extension 20 directly from SIPaccount test1, the CLI 
shows no problem :


[Jul  2 10:55:02] -- Executing [...@from-test:1] 
Dial(SIP/test1-010c, SIP/test2) in new stack

[Jul  2 10:55:02] -- Called test2
[Jul  2 10:55:02] -- SIP/test2-010d is ringing


So why can I call extension 20 (test2) directly but not transfer a call 
to it ??



Jonas.
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Re: [asterisk-users] Transfer fails

[Danny Nicholas]

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, July 02, 2010 4:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Transfer fails

 

Hello list,

this is the dialplan :

snip
exten = s,n,Dial(SIP/test1SIP/test2,,t)
snip

exten = 10,1,Dial(SIP/test1)
exten = 20,1,Dial(SIP/test2)


So there is an incoming call that rings SIPaccounts test1 and test2.
Account test1 answers and wants to transfer the call to test2.
Transfer is : #20

This is what the CLI shows :

[Jul  2 10:55:30] -- Executing [...@from-test:1]
Dial(SIP/test1-010e, SIP/test2) in new stack
[Jul  2 10:55:30] WARNING[7604]: app_dial.c:1296 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
[Jul  2 10:55:30]   == Everyone is busy/congested at this time (1:0/0/1)

...and the call is disconnected.

When I call the extension 20 directly from SIPaccount test1, the CLI shows
no problem :

[Jul  2 10:55:02] -- Executing [...@from-test:1]
Dial(SIP/test1-010c, SIP/test2) in new stack
[Jul  2 10:55:02] -- Called test2
[Jul  2 10:55:02] -- SIP/test2-010d is ringing


So why can I call extension 20 (test2) directly but not transfer a call to
it ??


Jonas.

 

-- 

A good possibility is that you have an over-restrictive call-limit (or
whatever it's called in your branch) that is filling the bucket on the
incoming call and not allowing a transfer.

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Re: [asterisk-users] Transfer fails

[Jonas Kellens]

Danny,

thank you for you feedback.

I have the following setting in sip.conf :

limitonpeer = yes

and for every sip peer definition I have :

asterisk*CLI sip show peer test1

  * Name   : test1
  Realtime peer: Yes, cached
  Secret   : Set
  MD5Secret: Not set
  Context  : from-TEST
  Subscr.Cont. : Not set
snip
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 4
snip


With a call limit of 4, I think it must be possible to transfer a call, 
no ?!



Jonas.


On 07/02/2010 03:02 PM, Danny Nicholas wrote:


A good possibility is that you have an over-restrictive call-limit (or 
whatever it's called in your branch) that is filling the bucket on 
the incoming call and not allowing a transfer.


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Re: [asterisk-users] Transfer calls using ##

[hin lee]

Thanks for replying Noah.  I'm using FreePBX web interface and have a ring 
group that rings 4 phones as the operator.  I do know that the context type is 
from-internal but when it rings as below, the context type becomes 
from-pstn.  Can you tell me where exactly to go and change in the FreePBX web 
interface?

Thanks!





From: Noah Miller noahisaacmil...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Wed, May 5, 2010 1:35:38 PM
Subject: Re: [asterisk-users] Transfer calls using ##

 I have a question about the blind transfer using ##. This works great on our
 cordless phone, but there have been occasions that we can't transfer using
 ##. I was able to reproduce the issue by doing the following:

 1) Call in from the outside line,
 2) Ask the operator to transfer me to an extension using ##.
 3) Get the voice mail greeting of the individual.
 4) Hit 0 for the operator before the greeting completed.
 5) Ask the operator to transfer me again using ##.
 6) Operator can't transfer and I can hear the pressing of the keys.

 Why can't I transfer the call the second time around? How can I fix this?

The dial statement in your 'o' extension must have the 't' flag.


- Noah

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Re: [asterisk-users] Transfer calls using ##

[Noah Miller]

 I have a question about the blind transfer using ##. This works great on our
 cordless phone, but there have been occasions that we can't transfer using
 ##. I was able to reproduce the issue by doing the following:

 1) Call in from the outside line,
 2) Ask the operator to transfer me to an extension using ##.
 3) Get the voice mail greeting of the individual.
 4) Hit 0 for the operator before the greeting completed.
 5) Ask the operator to transfer me again using ##.
 6) Operator can't transfer and I can hear the pressing of the keys.

 Why can't I transfer the call the second time around? How can I fix this?

The dial statement in your 'o' extension must have the 't' flag.


- Noah

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[asterisk-users] Transfer calls using ##

[hin lee]

I have a question about the blind transfer 
using ##.  This works great on our cordless phone, but there have been 
occasions that we can't transfer using ##.  I was able to reproduce the 
issue by doing the following:

1) Call in from the outside line,
2) Ask the operator to transfer me to an extension using ##.
3) Get the voice mail greeting of the individual.
4) Hit 0 for the operator before the greeting completed. 
5) Ask the operator to transfer me again using ##.
6) Operator can't transfer and I can hear the pressing of the keys.

Why can't I transfer the call the second time around?  How can I fix 
this?


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[asterisk-users] transfer two gsm mobile calls

[Francesc Perez i Botella]

Hello:

when using any fct or sip to gsm gateway, is possible to transfer an
incomming call to another number automatically from asterisk

say

incoming call (gsm gateway)
answer;
mobile user dial 0XXX
then retention 4 SEND KEY
then dial XXX
then x answer
then recover first call 4 send key
then transfer 2 send key...

my question is how can you emualte send key to gsm gayeway , is it device
dependant

tanks
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[asterisk-users] Transfer after pickup

[Benny Amorsen]

I am probably just being stupid again, but...

I have some non-SIP phones which are set up for doing transfers by DTMF,
by simply adding T or t to the appropriate Dial options. This works
quite well in general.

They can also do non-directed call pickup with *8. However, after a call
pickup they can't transfer the call by DTMF -- there is no Dial command
where I can add the t or T option.

How do I configure *8 to allow blind and attended transfers for the
person who dialled *8?


/Benny



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[asterisk-users] Transfer Issue with IAX Trunk

[Lutgring, Sam]

I have an IAX trunk configured between 2 Asterisk servers.  Everything is 
working great except if the caller presses # during the call.  If they press # 
the local PBX comes on and says transferring and tries to transfer to a blank 
extension.  Does anyone know how to turn this off?  There is no extension 
defined for # in the dial plan.

Thanks for your thoughts on this.


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Re: [asterisk-users] Transfer Issue with IAX Trunk

[Doug Lytle]

Lutgring, Sam wrote:

 I have an IAX trunk configured between 2 Asterisk servers.  Everything 
 is working great except if the caller presses # during the call.  If 
 they press # the local PBX comes on and says transferring and tries to 
 transfer to a blank extension.  Does anyone know how to turn this 
 off?  There is no extension defined for # in the dial plan.


core show application dial:

 t- Allow the called party to transfer the calling party by sending the
   DTMF sequence defined in features.conf.
 T- Allow the calling party to transfer the called party by sending the
   DTMF sequence defined in features.conf.

Doug


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Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Transfer Issue with IAX Trunk

[Administrator TOOTAI]

Doug Lytle a écrit :
 Lutgring, Sam wrote:
   
 I have an IAX trunk configured between 2 Asterisk servers.  Everything 
 is working great except if the caller presses # during the call.  If 
 they press # the local PBX comes on and says transferring and tries to 
 transfer to a blank extension.  Does anyone know how to turn this 
 off?  There is no extension defined for # in the dial plan.

 

 core show application dial:

  t- Allow the called party to transfer the calling party by sending the
DTMF sequence defined in features.conf.
  T- Allow the calling party to transfer the called party by sending the
DTMF sequence defined in features.conf.
   
Or change the config in features.conf

-- 
Daniel

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Re: [asterisk-users] transfer option and pressing #

[Brent Davidson]

Alex Samad wrote:
 Hi

 I have setup forwarding - xfering - where you press # and then the
 extension. I add t to the dial cmd.

 My problem is that when you call something like internet banking they
 want #, but when # is pressed asterisk gets it instead. is there a way
 around this ?

 I haven't been able to get asterisk to listen to flash either 


 Alex
   
The easiest solution would probably be to look in features.conf and 
change the option for forwarding to require two consecutive # presses.

The other option would be to put an explicit dial rule for the numbers 
that need the # bypass and have them omit T and from the dial command.

You could also set up a dat abase with a simple web front end for your 
users to enter numbers that need to have the transfer function bypassed 
and do something like this (I use AEL so this is in AEL Format)

macro specialDial (ext) {
if (${DB_EXISTS(bypass/${ext})}) {
   Dial (${TRUNK}/${ext});// Dial without transfer
} else {
   Dial (${TRUNK}/${ext},,T); // Dial With Transfer
}
}

This is assuming you create a table called Bypass in your Asterisk 
Database and add the number to the database.

Good luck,
Brent



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Re: [asterisk-users] transfer option and pressing #

[Alex Samad]

On Mon, Jul 13, 2009 at 11:50:00AM -0500, Brent Davidson wrote:
 Alex Samad wrote:
  Hi
 
  I have setup forwarding - xfering - where you press # and then the
  extension. I add t to the dial cmd.
 
  My problem is that when you call something like internet banking they
  want #, but when # is pressed asterisk gets it instead. is there a way
  around this ?
 
  I haven't been able to get asterisk to listen to flash either 
 
 
  Alex

 The easiest solution would probably be to look in features.conf and 
 change the option for forwarding to require two consecutive # presses.

actually when into features and change all the options to *digit
instead of #

 
 The other option would be to put an explicit dial rule for the numbers 
 that need the # bypass and have them omit T and from the dial command.
 
 You could also set up a dat abase with a simple web front end for your 
 users to enter numbers that need to have the transfer function bypassed 

this is a home system for now (also testbed)

 and do something like this (I use AEL so this is in AEL Format)
 
 macro specialDial (ext) {
 if (${DB_EXISTS(bypass/${ext})}) {
Dial (${TRUNK}/${ext});// Dial without transfer
 } else {
Dial (${TRUNK}/${ext},,T); // Dial With Transfer
 }
 }
 
 This is assuming you create a table called Bypass in your Asterisk 
 Database and add the number to the database.
 
 Good luck,
 Brent
 
 
 
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[asterisk-users] transfer option and pressing #

[Alex Samad]

Hi

I have setup forwarding - xfering - where you press # and then the
extension. I add t to the dial cmd.

My problem is that when you call something like internet banking they
want #, but when # is pressed asterisk gets it instead. is there a way
around this ?

I haven't been able to get asterisk to listen to flash either 


Alex

-- 
Why is it there are so many more horses' asses than there are horses?


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Re: [asterisk-users] transfer option and pressing #

[Alex Balashov]

Alex Samad wrote:

 Hi
 
 I have setup forwarding - xfering - where you press # and then the
 extension. I add t to the dial cmd.

No, that's simply the order of evaluation.  If the caller is inside an 
Asterisk application that listens for #, it is going to be intercepted 
and preempted instead of passed on.  Your only solution is to reassign 
the role of # to some other DTMF tone, if that's possible with the app 
that you're using.

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

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[asterisk-users] Transfer dropping calls

[Valter Nogueira]

When doing transfers the call drops as follows:

1. I receive a call (internal or not)
2. I dial *2, wait for transfer sound plus dialtone
3. I dial for destinantion person, who pickups the phone
4. We talk to each other
5. I hangup my phone and the call drops

if I dial * when talking with destination person a got the original call
back

The same occurs with blind transfers

We are using Asterisk 1.4.25.1 and X-lite softphones.

Thanks people

Valter
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Re: [asterisk-users] Transfer call from analog telephone

[Daniel Bareiro]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Daniel Bareiro wrote:

 As I've commented in a previous message, after dial *60 (of *600 to Echo
 test), I obtain like a tone cut in three parts followed of a continuous tone,
 causing that I'm incapable to dial the extension completely. The
 waitfordigit appears after to hangup. The cell_number seems to be some
 number that I has dial previously. Testing again with a SIP extension, this
 problem does not happen.

 Also it draws attention to me that the DTMF has a duration of 0ms.

 It is peculiar... after to have a restart of Asterisk, I can dial without
 problems to *600. This is Asterisk log corresponding to the successful
 communication with the extension: 

 - --
 [Jun  4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '*' 
 received on DAHDI/2-1, duration 0 ms
 [Jun  4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted 
 without begin '*' on DAHDI/2-1
 [Jun  4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end 
 passthrough '*' on DAHDI/2-1
 [Jun  4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '6' 
 received on DAHDI/2-1, duration 0 ms
 [Jun  4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted 
 without begin '6' on DAHDI/2-1
 [Jun  4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end 
 passthrough '6' on DAHDI/2-1
 [Jun  4 23:03:31] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '0' 
 received on DAHDI/2-1, duration 0 ms
 [Jun  4 23:03:31] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted 
 without begin '0' on DAHDI/2-1
 [Jun  4 23:03:31] DTMF[28905]: channel.c:2282 __ast_read: DTMF end 
 passthrough '0' on DAHDI/2-1
 [Jun  4 23:03:31] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '0' 
 received on DAHDI/2-1, duration 0 ms
 [Jun  4 23:03:31] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted 
 without begin '0' on DAHDI/2-1
 [Jun  4 23:03:31] DTMF[28905]: channel.c:2282 __ast_read: DTMF end 
 passthrough '0' on DAHDI/2-1
 -- Executing [*...@phones:1] Answer(DAHDI/2-1, ) in new stack
 [Jun  4 23:03:31] DEBUG[28905]: chan_dahdi.c:3174 dahdi_answer: Took 
 DAHDI/2-1 off hook
 -- Executing [*...@phones:2] Playback(DAHDI/2-1, demo-echotest) in 
 new stack
 -- DAHDI/2-1Playing 'demo-echotest' (language 'es')
  == Spawn extension (phones, *600, 2) exited non-zero on 'DAHDI/2-1'
 -- Hungup 'DAHDI/2-1'
 - --

 As you will see, the duration is always of 0 ms (also when I dial to the cell
 phone). After this I make several tests. To dial from cell phone to the analog
 phone and I did not have problems in to call immediately to *600 after to have
 dial to the cell phone in each opportunity. But if from my extension 201 I
 dial the analog phone and after that from my analog phone I dial to *600, it
 happens the same of problem of not to be able to dial beyond *60. Log of the
 CLI for this situation is the following one:

 - --
 [Jun  4 23:08:45] DTMF[29017]: channel.c:2229 __ast_read: DTMF end '*' 
 received on DAHDI/2-1, duration 0 ms
 [Jun  4 23:08:45] DTMF[29017]: channel.c:2271 __ast_read: DTMF end accepted 
 without begin '*' on DAHDI/2-1
 [Jun  4 23:08:45] DTMF[29017]: channel.c:2282 __ast_read: DTMF end 
 passthrough '*' on DAHDI/2-1
 [Jun  4 23:08:46] DTMF[29017]: channel.c:2229 __ast_read: DTMF end '6' 
 received on DAHDI/2-1, duration 0 ms
 [Jun  4 23:08:46] DTMF[29017]: channel.c:2271 __ast_read: DTMF end accepted 
 without begin '6' on DAHDI/2-1
 [Jun  4 23:08:46] DTMF[29017]: channel.c:2282 __ast_read: DTMF end 
 passthrough '6' on DAHDI/2-1
 [Jun  4 23:08:46] DTMF[29017]: channel.c:2229 __ast_read: DTMF end '0' 
 received on DAHDI/2-1, duration 0 ms
 [Jun  4 23:08:46] DTMF[29017]: channel.c:2271 __ast_read: DTMF end accepted 
 without begin '0' on DAHDI/2-1
 [Jun  4 23:08:46] DTMF[29017]: channel.c:2282 __ast_read: DTMF end 
 passthrough '0' on DAHDI/2-1
 -- Blacklisting number 201
 [Jun  4 23:08:54] DEBUG[29017]: chan_dahdi.c:6244 ss_thread: waitfordigit 
 returned  0...
 -- Hungup 'DAHDI/2-1'
 - --

Testing some more I could verify than if I changed the number for echo test to
*700 instead of *600, the problem of not being able to dial beyond *60
disappears. Investigating a little in Internet and reading the source code, I
found the following in the line 2834 of chan_mgcp.c file:

- -
2834   } else if (!ast_strlen_zero(p-lastcallerid)  !strcmp(p-dtmf_buf, 
*60)) {
2835   if (option_verbose  2) {
2836   ast_verbose(VERBOSE_PREFIX_3 Blacklisting number 
%s\n, p-lastcallerid);
2837   }
2838   res = 

Re: [asterisk-users] Transfer call from analog telephone

[Daniel Bareiro]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi Tilghman and Grygoriy.

Tilghman Lesher escribió:

 I was testing both the recall key and uncomment the following lines
 in the features.conf file:

 blindxfer = #1
 atxfer = *2

 verifying previously that the extension uses the arguments tT with
 the Dial application and to include the context featuremap in the
 context in which I have defined the extensions (internal).

 The telephone of the end with which the conversation is staying
 listens the tones to try doing the transfer, but Asterisk does not
 give the dial tone after *2 / #1 or the recall key.

 Remember that the time between the two digits is VERY short.  You must
 press those two digits in quick succession or else the requested
 feature code will not activate.

I made sure to make it sufficiently fast, but still increasing
featuredigittimeout, it did not work.

I am not sure if it will have some relation, but also found another
difficulty when the dial from my analog telephone.

When doing a echo test from an SIP extension, I don't have problems,
but, sometimes, with an analog telephone when trying to dial the
extension to realise the echo test (*600), after to have dial *60, a
tone cut in three parts is listened to soon a continuous tone, doing
impossible to be able to dial the extension completely. Sometimes it
works well, but sometimes it happens, that is something that draws
attention to me and, as it mentioned, from a SIP extension I'm not
having this problem.

This is what I get in the Asterisk CLI after to dial *60:

- --
-- Starting simple switch on 'DAHDI/2-1'
-- Blacklisting number 201
- --


I do not believe that it is something own of the analogical telephone.
Yesterday, exactly, I was testing with another telephone (of my work) to
discard that it could be something of the house telephone, and it happens the
same exactly.

Making the changes in logger.conf to also see the dialing DTMF tones,
they seem to be correctly passed:

- --
-- Starting simple switch on 'DAHDI/2-1'
[Jun  4 06:47:16] DTMF[8669]: channel.c:2229 __ast_read: DTMF end '*' received 
on DAHDI/2-1, duration 0 ms
[Jun  4 06:47:16] DTMF[8669]: channel.c:2271 __ast_read: DTMF end accepted 
without begin '*' on DAHDI/2-1
[Jun  4 06:47:16] DTMF[8669]: channel.c:2282 __ast_read: DTMF end passthrough 
'*' on DAHDI/2-1
[Jun  4 06:47:16] DTMF[8669]: channel.c:2229 __ast_read: DTMF end '6' received 
on DAHDI/2-1, duration 0 ms
[Jun  4 06:47:16] DTMF[8669]: channel.c:2271 __ast_read: DTMF end accepted 
without begin '6' on DAHDI/2-1
[Jun  4 06:47:16] DTMF[8669]: channel.c:2282 __ast_read: DTMF end passthrough 
'6' on DAHDI/2-1
[Jun  4 06:47:17] DTMF[8669]: channel.c:2229 __ast_read: DTMF end '0' received 
on DAHDI/2-1, duration 0 ms
[Jun  4 06:47:17] DTMF[8669]: channel.c:2271 __ast_read: DTMF end accepted 
without begin '0' on DAHDI/2-1
[Jun  4 06:47:17] DTMF[8669]: channel.c:2282 __ast_read: DTMF end passthrough 
'0' on DAHDI/2-1
-- Blacklisting number cell_number
[Jun  4 06:47:21] DEBUG[8669]: chan_dahdi.c:6244 ss_thread: waitfordigit 
returned  0...
-- Hungup 'DAHDI/2-1'
-- Starting simple switch on 'DAHDI/2-1'
[Jun  4 06:47:26] DEBUG[8670]: chan_dahdi.c:6244 ss_thread: waitfordigit 
returned  0...
-- Hungup 'DAHDI/2-1'
- --

As I've commented in a previous message, after dial *60 (of *600 to Echo
test), I obtain like a tone cut in three parts followed of a continuous tone,
causing that I'm incapable to dial the extension completely. The
waitfordigit appears after to hangup. The cell_number seems to be some
number that I has dial previously. Testing again with a SIP extension, this
problem does not happen.

Also it draws attention to me that the DTMF has a duration of 0ms.

It is peculiar... after to have a restart of Asterisk, I can dial without
problems to *600. This is Asterisk log corresponding to the successful
communication with the extension: 

- --
[Jun  4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '*' received 
on DAHDI/2-1, duration 0 ms
[Jun  4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted 
without begin '*' on DAHDI/2-1
[Jun  4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end passthrough 
'*' on DAHDI/2-1
[Jun  4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '6' received 
on DAHDI/2-1, duration 0 ms
[Jun  4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted 
without begin '6' on DAHDI/2-1
[Jun  4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end passthrough 
'6' on DAHDI/2-1
[Jun  4 23:03:31] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '0' 

Re: [asterisk-users] Transfer call from analog telephone

[Grygoriy Dobrovolskyy]

 Remember that the time between the two digits is VERY short.  You must
 press
 those two digits in quick succession or else the requested feature code
 will
 not activate.

 -

Or set featuredigittimeout longer.
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[asterisk-users] Transfer call from analog telephone

[Daniel Bareiro]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi all!

I'm trying to doing a transfer from an analog extension to a SIP
extension but until the moment I was not successful.

I was testing both the recall key and uncomment the following
lines in the features.conf file:

blindxfer = #1
atxfer = *2

verifying previously that the extension uses the arguments tT with the
Dial application and to include the context featuremap in the context
in which I have defined the extensions (internal).

The telephone of the end with which the conversation is staying listens
the tones to try doing the transfer, but Asterisk does not give the dial
tone after *2 / #1 or the recall key.

I copy my configuration files after to have reverted the changes. If some
other data is necessary, don't doubt in consulting to me. The lines that I
added to the configuration files created in the installation are those that
are underneath DGB.

## /etc/asterisk/features.conf 

[general]
parkext = 700  ; What extension to dial to park
parkpos = 701-720  ; What extensions to park calls on. These needs 
to be
; numeric, as Asterisk starts from the start 
position
; and increments with one for the next parked 
call.
context = parkedcalls  ; Which context parked calls are in
; (default is 45 seconds)
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
; one of: parked, caller, both  (default is 
caller)
; one of: callee, caller, both, no (default is 
both)
; one of: callee, caller, both, no (default is 
no)
; one of: callee, caller, both, no (default is 
no)
; one of: callee, caller, both, no (default is 
no)
; Defaults to 'first' available
; as long as the class is not set on the 
channel directly
; using Set(CHANNEL(musicclass)=whatever) in 
the dialplan

; (default is 3 seconds)
; feature activation  (default is 1000 ms)


[featuremap]

[applicationmap]

## /etc/asterisk/extensions.conf

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

[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest   ; IAXtel username/password
TRUNK=Zap/G2; Trunk interface
TRUNKMSD=1  ; MSD digits to strip (usually 
1 or 0)

[default]

; DGB
[internal]
exten = _2xx,1,Dial(SIP/${EXTEN},15,tTm)
exten = _2xx,2,VoiceMail(${ext...@voicemail)
exten = _2xx,3,Playback(vm-goodbye)
exten = _2xx,4,Hangup

exten = *98,1,Answer
exten = *98,2,Wait(1)
exten = *98,3,VoiceMailMain(${caller...@voicemail)
exten = *98,4,Hangup

exten = *600,1,Answer
exten = *600,2,Playback(demo-echotest)
exten = *600,3,Echo
exten = *600,4,Playback(demo-echodone)
exten = *600,5,Hangup

exten = _9.,1,Dial(DAHDI/1/${EXTEN:1})
exten = _9.,2,Hangup

exten = 1010,1,Dial(DAHDI/2,15,tTm)
exten = 1010,2,Hangup

include = phones

[phones]
include = internal

[incoming]


exten = s,1,Dial(SIP/201,15,tTm)
exten = s,2,Hangup

## /etc/asterisk/chan_dahdi.conf

[trunkgroups]

[channels]
context=default
switchtype=national
signalling=fxo_ls
rxwink=300  ; Atlas seems to use long (250ms) winks

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1

immediate=no
busydetect=yes

; DGB
language=es
defaultzone=es
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
inmediate=no

context=phones
signalling=fxo_ks
channel = 2   ; Telephone attached to port 2
context=incoming
signalling=fxs_ks  ; Use FXS signalling for an FXS channel
channel = 1   ; PSTN attached to port 1

##


Which can be the problem or what configuration can be lacking?

Thanks in avance.

Regards,
Daniel

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Re: [asterisk-users] Transfer call from analog telephone

[Tilghman Lesher]

On Monday 01 June 2009 04:52:14 Daniel Bareiro wrote:
 I was testing both the recall key and uncomment the following
 lines in the features.conf file:

 blindxfer = #1
 atxfer = *2

 verifying previously that the extension uses the arguments tT with the
 Dial application and to include the context featuremap in the context
 in which I have defined the extensions (internal).

 The telephone of the end with which the conversation is staying listens
 the tones to try doing the transfer, but Asterisk does not give the dial
 tone after *2 / #1 or the recall key.

Remember that the time between the two digits is VERY short.  You must press
those two digits in quick succession or else the requested feature code will
not activate.

-- 
Tilghman

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[asterisk-users] TRANSFER EVENT ON QUEUE_LOG

[Sebastian]

Hi,

 

Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6.

I make an attended transfer (asterisk feature), and I cant see the event.

 

Any idea? Should I submit a bug report?

 

 

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Re: [asterisk-users] TRANSFER EVENT ON QUEUE_LOG

[Alex Balashov]

Sebastian wrote:
 Hi,
 
  
 
 Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6.
 
 I make an attended transfer (asterisk feature), and I cant see the event.
 
  
 
 Any idea? Should I submit a bug report?

If you do, be sure to headline it in all caps.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] TRANSFER EVENT ON QUEUE_LOG

[Sebastian]

Forget about this.

Is still working.

 

 

From: Sebastian [mailto:s...@adinet.com.uy] 
Sent: viernes, 13 de marzo de 2009 10:05 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: TRANSFER EVENT ON QUEUE_LOG

 

Hi,

 

Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6.

I make an attended transfer (asterisk feature), and I cant see the event.

 

Any idea? Should I submit a bug report?

 

 

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Re: [asterisk-users] TRANSFER EVENT ON QUEUE_LOG

[Sebastian]

I made another post, it is working, I have queue_log to mysql db and I have
a trigger that made the insert fail.

Sorry for the post!.

Regards


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: viernes, 13 de marzo de 2009 10:14 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TRANSFER EVENT ON QUEUE_LOG

Sebastian wrote:
 Hi,
 
  
 
 Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6.
 
 I make an attended transfer (asterisk feature), and I cant see the event.
 
  
 
 Any idea? Should I submit a bug report?

If you do, be sure to headline it in all caps.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] Transfer Asterisk 1.6 Telephone IP

[Daviramos Roussenq Fortunato]

Hi,

My IP phone has an option to send the flash via DTMF.
Enable the sending and the DEBUG I get the following:

 [ TYPE: Control (4) SUBCLASS: Flash (9) ] [SIP/34314730-b7714f38]
[Feb 10 15:12:51] WARNING[29203]: chan_sip.c:5350 sip_indicate: Don't know
how to indicate condition 9
[Feb 10 15:12:51] WARNING[29203]: channel.c:2858 ast_indicate_data: Unable
to handle indication 9 for 'SIP/13649-2001-b7718f20'

How do I set the flash to work as *2.

[featuremap]
  blindxfer=##
  atxfer=*2
  automon=*1
  disconnect=**


2009/2/9 Daviramos Roussenq Fortunato daviramo...@gmail.com

 Hi List.

   I have a small problem in using the transfer key transfer of IP Phone in
 Asterisk 1.6, I think I spend some detail in the configuration but can not
 find.

   What happens is, when I do a transfer using the Transfer button, the
 phone, does not play the music on hold, which is waiting on the phone is
 silent, and I have the same settings on a 1.4 server, and the music plays
 correctly when using the same phone. When using the * 2 to transfer the
 connection or a softfone, the music plays correctly on this server with
 Asterisk 1.6.

   What the detail is missing in my configuration?

   My Configuration

   [featuremap]
   blindxfer=##
   atxfer=*2
   automon=*1
   disconnect=**

 I made a DEBUG to use the channel when the two key TRANSFER Server 1.4 and
 1.6.

 Command:
 core set debug channel SIP/2720-b7d28d70

 DEBUG no 1.4:
  [ TYPE: Control (4) SUBCLASS: Unknown control '16' (16) ]
 [SIP/2720-b7d28d70]
 -- Started music on hold, class 'default', on SIP/2001-08a56cf8
  [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/2720-b7d28d70]

 DEBUG no 1.6:
 When tightening the TRANSFER button on the console does not show anything,
 but when any other key grip CLI appears in decimal value of the
 corresponding key.

 I'm using dtmfmode=rfc2833


 How you help me?

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[asterisk-users] Transfer Asterisk 1.6 Telephone IP

[Daviramos Roussenq Fortunato]

Hi List.

  I have a small problem in using the transfer key transfer of IP Phone in
Asterisk 1.6, I think I spend some detail in the configuration but can not
find.

  What happens is, when I do a transfer using the Transfer button, the
phone, does not play the music on hold, which is waiting on the phone is
silent, and I have the same settings on a 1.4 server, and the music plays
correctly when using the same phone. When using the * 2 to transfer the
connection or a softfone, the music plays correctly on this server with
Asterisk 1.6.

  What the detail is missing in my configuration?

  My Configuration

  [featuremap]
  blindxfer=##
  atxfer=*2
  automon=*1
  disconnect=**

I made a DEBUG to use the channel when the two key TRANSFER Server 1.4 and
1.6.

Command:
core set debug channel SIP/2720-b7d28d70

DEBUG no 1.4:
 [ TYPE: Control (4) SUBCLASS: Unknown control '16' (16) ]
[SIP/2720-b7d28d70]
-- Started music on hold, class 'default', on SIP/2001-08a56cf8
 [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/2720-b7d28d70]

DEBUG no 1.6:
When tightening the TRANSFER button on the console does not show anything,
but when any other key grip CLI appears in decimal value of the
corresponding key.

I'm using dtmfmode=rfc2833


How you help me?
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[asterisk-users] Transfer in Asterisk 1.6

[Daviramos Roussenq Fortunato]

Hi, All


How to enable the transfer in Asterisk 1.6?

Not found the module res_features.so.
blindxfer = #
atxfer = *2
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Re: [asterisk-users] Transfer/Park Question.

[Brent Davidson]

Daniel Hazelbaker wrote:
 On Oct 9, 2008, at 2:59 PM, Brent Davidson wrote:

   
 Short answer: currently no.

 Medium answer: I just rolled out 60+ Snom phones (300s and 320s) and  
 we do call parking with DTMF.  People were used to just hitting PARK  
 and their phone displaying the park extension (old NEC system).  I  
 didn't tell anybody anything except it will speak the extension back  
 to you and nobody has complained about hearing the DTMF digits.  We  
 chose a 3 digit code (#92 I believe) to try an alleviate the  
 possibility of somebody accidently parking a call  while filling out a  
 DTMF based form/menu system, but in theory you could assign just * to  
 park and only deal with 1 tone.  Just be aware that if the user needs  
 to hit * for anything else, they won't be able to use it.

 Long answer: Snom phones support text messages to the phone that  
 automatically display.  I am looking for a way to use that in  
 conjunction with Snom's ParkOrbit feature (which does work, you just  
 don't hear the extension).  Basically Asterisk would do a normal park  
 and then trigger a SIP NOTIFY message to the parkING phone that says  
 Parked: 701.  The message can be cleared by the user by pressing X,  
 or ideally Asterisk would auto-clear the message after 10 seconds (or  
 whatever).

 In theory I can do the long answer now with a Manager application,  
 but I don't like the idea of relying on an external application.  If  
 it crashes or locks up for whatever reason then suddenly people get  
 parked and nobody knows where.

 Also be aware that in 1.2.x and 1.4.x, if you park a call and then  
 pick it up, you can't park it again.  At least not with the DTMF  
 method.  I borrowed a patch from the 1.6 branch that fixes this and  
 made it applicable to 1.4.20.1, well I borrowed part of it.  The  
 entire patch let you configure who could park etc., I wanted both  
 sides to always park so I just took the 2 or 3 lines that were needed  
 for that.  If you are interested I can e-mail it to you directly.

 Regards,
 Daniel

   
I wasn't aware of the inability to re-park calls in 1.4  That could have 
been a nasty surprise.  I would be very interested in the patch that 
fixes that.

Thanks,
Brent

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Re: [asterisk-users] Transfer/Park Question.

[Doug Lytle]

Brent Davidson wrote:
Also be aware that in 1.2.x and 1.4.x, if you park a call and then  
pick it up, you can't park it again.  At least not with the DTMF  

  

I wasn't aware of the inability to re-park calls in 1.4  That could have 
been a nasty surprise.  I would be very interested in the patch that 
fixes that.
  


I don't remember where I got it (Might have been the bug tracker) that 
works fine under the current 1.4.x.  I had to do a minor change to get 
it to apply.


Copy into Asterisk source directory

patch -p0 *.patch

rm *.patch

make
make install


Doug


--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.

Index: res/res_features.c
===
--- res/res_features.c	(revision 84405)
+++ res/res_features.c	(working copy)
@@ -1670,7 +1670,7 @@
 	}
 	if (con) {
 		char returnexten[AST_MAX_EXTENSION];
-		snprintf(returnexten, sizeof(returnexten), %s|30|t, peername);
+		snprintf(returnexten, sizeof(returnexten), %s|30|tk, peername);
 		ast_add_extension2(con, 1, peername, 1, NULL, NULL, Dial, strdup(returnexten), ast_free, registrar);
 	}
 	set_c_e_p(chan, parking_con_dial, peername, 1);
@@ -1927,6 +1927,7 @@
 		memset(config, 0, sizeof(struct ast_bridge_config));
 		ast_set_flag((config.features_callee), AST_FEATURE_REDIRECT);
 		ast_set_flag((config.features_caller), AST_FEATURE_REDIRECT);
+		ast_set_flag((config.features_caller), AST_FEATURE_PARKCALL);
 		res = ast_bridge_call(chan, peer, config);
 
 		pbx_builtin_setvar_helper(chan, PARKEDCHANNEL, peer-name);

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Re: [asterisk-users] Transfer/Park Question.

[Brent Davidson]

Doug Lytle wrote:
 Brent Davidson wrote:
 Also be aware that in 1.2.x and 1.4.x, if you park a call and then  
 pick it up, you can't park it again.  At least not with the DTMF 
   
 I wasn't aware of the inability to re-park calls in 1.4  That could 
 have been a nasty surprise.  I would be very interested in the patch 
 that fixes that.
   

 I don't remember where I got it (Might have been the bug tracker) that 
 works fine under the current 1.4.x.  I had to do a minor change to get 
 it to apply.

 Copy into Asterisk source directory

 patch -p0 *.patch

 rm *.patch

 make
 make install


 Doug
Ok, the patch is working great.  Any idea what would make the one step 
parking not work?  I've tried several DTMF combinations in features.conf 
and none of them seem to work when manually dialed or when bound as a 
DTMF code to a key.

So far I've tried the following under [featuremap] in features.conf:

parkcall = *5
parkcall = #72
parkcall = *9
parkcall = #75

I don't even see any acknowledgment of the DTMF tones showing up on the 
console.

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Re: [asterisk-users] Transfer/Park Question.

[Doug Lytle]

Brent Davidson wrote:
 Ok, the patch is working great.  Any idea what would make the one step 
 parking not work?  I've tried several DTMF combinations in features.conf 
   


Check your featuredigittimeout, it defaults to 1/2 second.  You may need 
to increase it.

I have mine set to ## to activate, easier to do it quickly.

Doug



-- 
 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Transfer/Park Question.

[Daniel Hazelbaker]

On Oct 10, 2008, at 1:00 PM, Brent Davidson wrote:

 Doug Lytle wrote:
 I don't remember where I got it (Might have been the bug tracker)  
 that
 works fine under the current 1.4.x.  I had to do a minor change to  
 get
 it to apply.

 Copy into Asterisk source directory

 patch -p0 *.patch

 rm *.patch

 make
 make install


 Doug
 Ok, the patch is working great.  Any idea what would make the one step
 parking not work?  I've tried several DTMF combinations in  
 features.conf
 and none of them seem to work when manually dialed or when bound as a
 DTMF code to a key.

 So far I've tried the following under [featuremap] in features.conf:

 parkcall = *5
 parkcall = #72
 parkcall = *9
 parkcall = #75

 I don't even see any acknowledgment of the DTMF tones showing up on  
 the
 console.

You won't.  The patch I sent you off-list is incomplete, this one is  
better. I forgot I fixed the parked has timed out option in another  
patch before I fixed this part.  Anyway, make sure when you dial you  
put k in the dial options (K too if you want both sides to park).   
It used to be tied to the t option I believe and then got moved out  
to k at some point.  Other than that, it should work.

Daniel


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Re: [asterisk-users] Transfer/Park Question.

[Brent Davidson]

Doug Lytle wrote:

Brent Davidson wrote:
  
Ok, the patch is working great.  Any idea what would make the one step 
parking not work?  I've tried several DTMF combinations in features.conf 
  




Check your featuredigittimeout, it defaults to 1/2 second.  You may need 
to increase it.


I have mine set to ## to activate, easier to do it quickly.

Doug

  
I checked that.  I've got mine set to 800 and all of my other 2-digit 
features work (transfer, blind transfer, etc).  The only one that 
doesn't is the parkcall feature.
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Re: [asterisk-users] Transfer/Park Question.

[Brent Davidson]

Daniel Hazelbaker wrote:



You won't.  The patch I sent you off-list is incomplete, this one is  
better. I forgot I fixed the parked has timed out option in another  
patch before I fixed this part.  Anyway, make sure when you dial you  
put k in the dial options (K too if you want both sides to park).   
It used to be tied to the t option I believe and then got moved out  
to k at some point.  Other than that, it should work.


Daniel

  


That was it.  Needed to add the k options.  All is working now.  Also, I 
don't think I got a patch from you off-list.  The one I got from you was 
the asterisk-1.4.20.1-callreparking.patch.  Got another one from Doug 
Lytle called multi-park.patch.
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[asterisk-users] Transfer/Park Question.

[Brent Davidson]

I've got a situation where I need to use a transfer to the parking lot 
as hold, but am not going to use BLF indicators on the phone to pick up 
the parked calls so I need to hear the 3-digit extension after the 
transfer.  I'm using Snom 300 phones and have tried setting a 
programmable button to Key Event F_TRANSFER 700, which successfully does 
the transfer but cuts off audio so you don't hear the extension to 
dial.   Same with setting a Park Orbit.  I can use the DTMF button type 
to send the transfer command and then the extension but then the person 
doing the parking hears all of the tones, which is annoying.

Is there any way to set up the transfer silently and still get the 
parking slot extension back?

Thanks,
Brent Davidson

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Re: [asterisk-users] Transfer/Park Question.

[Daniel Hazelbaker]

On Oct 9, 2008, at 2:59 PM, Brent Davidson wrote:

 I've got a situation where I need to use a transfer to the parking lot
 as hold, but am not going to use BLF indicators on the phone to pick  
 up
 the parked calls so I need to hear the 3-digit extension after the
 transfer.  I'm using Snom 300 phones and have tried setting a
 programmable button to Key Event F_TRANSFER 700, which successfully  
 does
 the transfer but cuts off audio so you don't hear the extension to
 dial.   Same with setting a Park Orbit.  I can use the DTMF button  
 type
 to send the transfer command and then the extension but then the  
 person
 doing the parking hears all of the tones, which is annoying.

 Is there any way to set up the transfer silently and still get the
 parking slot extension back?

Short answer: currently no.

Medium answer: I just rolled out 60+ Snom phones (300s and 320s) and  
we do call parking with DTMF.  People were used to just hitting PARK  
and their phone displaying the park extension (old NEC system).  I  
didn't tell anybody anything except it will speak the extension back  
to you and nobody has complained about hearing the DTMF digits.  We  
chose a 3 digit code (#92 I believe) to try an alleviate the  
possibility of somebody accidently parking a call  while filling out a  
DTMF based form/menu system, but in theory you could assign just * to  
park and only deal with 1 tone.  Just be aware that if the user needs  
to hit * for anything else, they won't be able to use it.

Long answer: Snom phones support text messages to the phone that  
automatically display.  I am looking for a way to use that in  
conjunction with Snom's ParkOrbit feature (which does work, you just  
don't hear the extension).  Basically Asterisk would do a normal park  
and then trigger a SIP NOTIFY message to the parkING phone that says  
Parked: 701.  The message can be cleared by the user by pressing X,  
or ideally Asterisk would auto-clear the message after 10 seconds (or  
whatever).

In theory I can do the long answer now with a Manager application,  
but I don't like the idea of relying on an external application.  If  
it crashes or locks up for whatever reason then suddenly people get  
parked and nobody knows where.

Also be aware that in 1.2.x and 1.4.x, if you park a call and then  
pick it up, you can't park it again.  At least not with the DTMF  
method.  I borrowed a patch from the 1.6 branch that fixes this and  
made it applicable to 1.4.20.1, well I borrowed part of it.  The  
entire patch let you configure who could park etc., I wanted both  
sides to always park so I just took the 2 or 3 lines that were needed  
for that.  If you are interested I can e-mail it to you directly.

Regards,
Daniel

 Thanks,
 Brent Davidson

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[asterisk-users] Transfer a call without announce : no sound

[Nicolas Ross]

When we receive a call from outside (via a sangoma 104d card) and we do a
blind transfer, that is without anouncing to the called party , we have no
sound either way.

Exemple :

I take my cell phone to call my * box, it rings on my aastra 9113i phone, I
answer.
Then I hit the xfer buton, make my second call to another extention (it can
be either a aastra phone, nortel phone trough ciel portico, whatever.
As soon it rings I hangup or hit the xfer buton again.
Then the bridged call between the other extension and the zap channel have
no sound either way.

If I wait for the called party to answer and announce the transfer, all is
fine.

I've had report of sound one way also, but I wasn't able to reproduce.

Here's the log from my console :

   -- SIP/224-09e0f098 answered Zap/1-1
-- Started music on hold, class 'default', on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] Macro(SIP/224-09e1d728,
ael-std-exten|225|SIP/225) in new stack
-- Executing [EMAIL PROTECTED]:1] Set(SIP/224-09e1d728,
ext=225) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/224-09e1d728,
dev=SIP/225) in new stack
-- Executing [EMAIL PROTECTED]:3] Answer(SIP/224-09e1d728, ) in
new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(SIP/224-09e1d728,
Nicolas Ross 224) in new stack
-- Executing [EMAIL PROTECTED]:5] Wait(SIP/224-09e1d728, 0.5)
in new stack
-- Executing [EMAIL PROTECTED]:6] Dial(SIP/224-09e1d728,
SIP/225|15) in new stack
-- Called 225
-- SIP/225-09e73388 is ringing
-- Stopped music on hold on Zap/1-1
  == Spawn extension (macro-ael-std-exten, s, 6) exited non-zero on
'SIP/224-09e1d728ZOMBIE' in macro 'ael-std-exten'
  == Spawn extension (macro-ael-std-exten, s, 6) exited non-zero on
'SIP/224-09e1d728ZOMBIE'
-- SIP/225-09e73388 answered Zap/1-1

Any ideas ? 


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[asterisk-users] Transfer via AMI

[Nicholas Blasgen]

I have a call between two people.  I know their channel identifier.  I want
to trasfer a call away from one person and pass it to another person.

To start, let's talk about a blind transfer.  My system places both outgoing
calls to people and bridges them together (cheaper, works via AGI).

Action: Redirect
Channel: prospect
ExtraChannel: 0
Exten: SIP/transfer_to
Context: default
Priority: 1

So that works just fine.  I'm having an issue however that when the person
who was orginally talking decides to hang up his call, Asterisk disconnects
the other line as well, as if the ownership of that line is still controled
by the orginal process.  I'd love to solve that problem.  Maybe putting the
SIP/transfer_to into the ExtraChannel and then transfering them to a
conference room.  Suggestions welcome.  Could also be that AGI maintains
control of any channels it creates and when the main calling line dies, it
kills all the others even if they've been transfered away.

Okay, in the end, I'd like this to be assisted transfer.  Place the party on
hold, call another party, and then bridge the two together.  Whenever a
channel is taken away from the current person, the call status is returned
and my AGI script can continue.  So I think it should be fine.  Has anyone
done anything like this?  Any pointers would be great.

PS: (update since I wrote this original message a while back), via the web,
you click a link.  That creates a CALL file which calls your number.  Once
connected, it passes it to an extension that spawns an AGI program.  That
AGI program looks in the database for the number you wanted to call and
places that phone call.  You than chat with that person and decide that
you're done with that call and want to go onto your next phone call.  I use
the Asterisk Manager Interface (AMI) to perform a Redirect on the person
you're talking to.  Doing this causes the AGI script to continue.

-- 
Nicholas Blasgen
[EMAIL PROTECTED]
408.497.9796 (c)
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Re: [asterisk-users] Transfer

[Adrian Marsh]

Thanks Sherwood,

But how do I send back a 302, once I'm already in the dialplan (hasn't
asterisk already sent back a 200 OK by this point??)

Adrian

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: 23 May 2008 17:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfer

Adrian Marsh wrote:

 Hi All,

 In my old telco days (SS7), if I was wanting to hand back a call to 
 the network for transfer to a different PSTN number, there was a 
 specific SS7 action I could take, which send the call back to the 
 network, which in turn then routed the call appropriately. It added a 
 transfer-number into the SS7 headers so that the originating number, 
 dialed number and transfer number all stayed to specs, and everyone 
 was happy.

 In SIP/Asterisk, it seems that I have to hairpin/trombone the call (to

 have at least the control packets go via my SIP server), and use a 
 Dial out to the far end.

 So - is there a way of handing the call back to the network in
asterisk ?

 My detailed problem is this: When a call comes in, I want to send it 
 onto users mobiles, if I hairpin the call that's OK, except the CLI 
 needs to be that of the originator (from the USERS point of view) so 
 they can decide if they want to accept the call.

 Here in the UK, this is where the issues begin... the carriers here 
 don't like it if your sending CLI for other countries, that don't 
 match what they think they should receive from that connecting 
 carrier. Eg, if a call coming to them is 13 digits, but they only 
 expect 11 from that carrier, then they cut the digits. This turns a US

 originated call into a Southampton UK originated call!

 So I was hoping that handing the call back to the network in the 
 traditional sense would make it their problem and not mine... lol

 Thanks,


 Adrian

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http://www.google.com/search?hl=enq=asterisk+302+redirect+sipbtnG=Sear
ch
I believe you're looking for a 302 Redirect? Sorry if you're not, but
that sounds like what you want


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Re: [asterisk-users] Transfer

[Adrian Marsh]

Thanks Sherwood,

But how do I send back a 302, once I'm already in the dialplan (hasn't
asterisk already sent back a 200 OK by this point??)

Adrian

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: 23 May 2008 17:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfer

Adrian Marsh wrote:

 Hi All,

 In my old telco days (SS7), if I was wanting to hand back a call to 
 the network for transfer to a different PSTN number, there was a 
 specific SS7 action I could take, which send the call back to the 
 network, which in turn then routed the call appropriately. It added a 
 transfer-number into the SS7 headers so that the originating number, 
 dialed number and transfer number all stayed to specs, and everyone 
 was happy.

 In SIP/Asterisk, it seems that I have to hairpin/trombone the call (to

 have at least the control packets go via my SIP server), and use a 
 Dial out to the far end.

 So - is there a way of handing the call back to the network in
asterisk ?

 My detailed problem is this: When a call comes in, I want to send it 
 onto users mobiles, if I hairpin the call that's OK, except the CLI 
 needs to be that of the originator (from the USERS point of view) so 
 they can decide if they want to accept the call.

 Here in the UK, this is where the issues begin... the carriers here 
 don't like it if your sending CLI for other countries, that don't 
 match what they think they should receive from that connecting 
 carrier. Eg, if a call coming to them is 13 digits, but they only 
 expect 11 from that carrier, then they cut the digits. This turns a US

 originated call into a Southampton UK originated call!

 So I was hoping that handing the call back to the network in the 
 traditional sense would make it their problem and not mine... lol

 Thanks,


 Adrian

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http://www.google.com/search?hl=enq=asterisk+302+redirect+sipbtnG=Sear
ch
I believe you're looking for a 302 Redirect? Sorry if you're not, but
that sounds like what you want


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[asterisk-users] Transfer

[Adrian Marsh]

Hi All,

 

In my old telco days (SS7), if I was wanting to hand back a call to the
network for transfer to a different PSTN number, there was a specific
SS7 action I could take, which send the call back to the network, which
in turn then routed the call appropriately. It added a transfer-number
into the SS7 headers so that the originating number, dialed number and
transfer number all stayed to specs, and everyone was happy.

 

In SIP/Asterisk, it seems that I have to hairpin/trombone the call (to
have at least the control packets go via my SIP server), and use a Dial
out to the far end.

 

So - is there a way of handing the call back to the network in asterisk
?

 

My detailed problem is this:   When a call comes in, I want to send it
onto users mobiles, if I hairpin the call that's OK, except the CLI
needs to be that of the originator (from the USERS point of view) so
they can decide if they want to accept the call.

 

Here in the UK, this is where the issues begin...  the carriers here
don't like it if your sending CLI for other countries, that don't match
what they think they should receive from that connecting carrier. Eg, if
a call coming to them is 13 digits, but they only expect 11 from that
carrier, then they cut the digits. This turns a US originated call into
a Southampton UK originated call!

 

So I was hoping that handing the call back to the network in the
traditional sense would make it their problem and not mine... lol

 

Thanks,


Adrian

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