[asterisk-users] Asterisk call transfers not working

[Danny Nicholas]

Now I need to see what is in [from-internal-custom] and
[from-internal-additional]

 

Or you could just add these lines to [from-internal-xfer]

Exten => _X,1,Dial(SIP/${EXTEN},30,iKkTtt)

Exten => _XX,1,Dial(SIP/${EXTEN},30,iKkTt)

 

If you have 3 or 4 digit extensions you would need these lines

Exten => _XXX,1,Dial(SIP/${EXTEN},30,iKkTtt)

Exten => _XXXX,1,Dial(SIP/${EXTEN},30,iKkTt)

 

Sorry for the "cap" Exten - outlook did that.

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ramiro Paz
Sent: Wednesday, October 19, 2011 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk call transfers not working

 

Hi Danny:

Thanks for your response.

[from-internal-xfer]
include => from-internal-custom
include => from-internal-additional ; auto-generated
exten => s,1,Macro(hangupcall)
exten => h,1,Macro(hangupcall)

I have to tell you that we use Freepbx 2.9. I hope you can help me to solve
this issue. Let's say I'am an asterisk newbie because my asterisk knowledge
is just basic. This is the first time I got this system working and I really
liked it. Thanks for your time.

Ramiro PAZ
MASTERLINE LOGISTICS


On Wed, Oct 19, 2011 at 11:39 AM, Danny Nicholas <danny at debsinc.com> wrote:

What does your context [from-internal-xfer] look like? (it should either
resemble or have an include for [default] context).

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ramiro Paz
Sent: Wednesday, October 19, 2011 10:33 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Asterisk call transfers not working

 

Hello:

We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0
running. Everything seems to be ok but call transfers. This is the issue:

A, B, C and D are in FXS ports.
1) A calls B. B anwers.
2) B tries to transfer the call to C dialing *2 (code for attended
transfer). 
3) A hears MOH. B dials number C.
4) Asterisk says the dialed number is incorrect or non existing.

We tried with blind transfers and the same problem.
This is the Asterisk CLI log when making a call transfer:
    
-- <DAHDI/19-1> Playing 'pbx-transfer.gsm' (language 'es')
[Oct 19 09:00:21] WARNING[18521]: features.c:2319 builtin_atxfer: No digits
dialed for atxfer.
    -- <DAHDI/19-1> Playing 'pbx-invalid.gsm' (language 'es')
    -- <DAHDI/19-1> Playing 'pbx-transfer.gsm' (language 'es')
[Oct 19 09:00:50] WARNING[18521]: features.c:2319 builtin_atxfer: No digits
dialed for atxfer.
    -- <DAHDI/19-1> Playing 'pbx-invalid.gsm' (language 'es')
    -- <DAHDI/19-1> Playing 'pbx-transfer.gsm' (language 'es')
[Oct 19 09:01:52] WARNING[18521]: features.c:2319 builtin_atxfer: No digits
dialed for atxfer.

or

-- <DAHDI/17-1> Playing 'pbx-transfer.gsm' (language 'es')
[Oct  8 10:15:56] WARNING[3840]: features.c:2315 builtin_atxfer: Extension
'41' does not exist in context 'from-internal-xfer'
    -- <DAHDI/17-1> Playing 'pbx-invalid.gsm' (language 'es')
    -- <DAHDI/17-1> Playing 'pbx-transfer.gsm' (language 'es')
[Oct  8 10:16:11] WARNING[3840]: features.c:2315 builtin_atxfer: Extension
'41' does not exist in context 'from-internal-xfer'
    -- <DAHDI/17-1> Playing 'pbx-invalid.gsm' (language 'es')
    -- <DAHDI/17-1> Playing 'pbx-transfer.gsm' (language 'es')
[Oct  8 10:16:27] WARNING[3840]: features.c:2315 builtin_atxfer: Extension
'4' does not exist in context 'from-internal-xfer'
    -- <DAHDI/17-1> Playing 'pbx-invalid.gsm' (language 'es')

I'd really appreciate your help. Thanks in advance.

Ramiro PAZ
MASTERLINE LOGISTICS


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