[asterisk-users] Problem with call transfer from one server to another server

[akhilesh chand]

Server B(child server)

*chan_dahdi.conf*

[trunkgroups]

[channels]
group=1
context=outbound
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
faxdetect=both


callprogress=no
progzone=in
pulsedial=yes
;busydetect=yes

callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.5
txgain=0.5
callgroup=1
pickupgroup=1

pritimer => t309,6000

immediate=no

switchtype=euroisdn

context=outgoing
signalling=pri_cpe
pridialplan=unknown
group=1
channel => 1-15,17-31

context=outgoing
signalling=pri_cpe
pridialplan=unknown
group=2
channel => 32-46,48-62

context=outgoing
signalling=pri_cpe
pridialplan=unknown
group=3
channel => 63-77,79-93

context=outgoing
signalling=pri_cpe
pridialplan=unknown
group=4
channel => 94-108,110-124

*Sip.conf*

[general]
pear=type
context=hunt_incoming
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=all
nat=yes
callerid = LITE
externip=
externhost=
autocreatepeer=yes
autodomain=yes
localnet=192.168.14.112/255.255.255.0
canreinvite=yes
language=En
allowtransfer=yes
realm=telunet
domain=192.168.14.112
maxexpiry=3600
defaultexpiry=200
useragent=LITE PBX
usereqphone = yes
dtmfmode = rfc2833
alwaysauthreject = no
regcontext=sipregistrations
rtptimeout=3600
rtpholdtimeout=300
rtcachefriends=yes
;--------------------------- SIP DEBUGGING
---------------------------------------------------
sipdebug = yes
registertimeout=60
registerattempts=5
callgroup=1
pickupgroup=1
callevents=yes

Disallow=all
Allow=all
;Allow=ulaw
;Allow=gsm
Canreinvite=no

;register => <username>:<password>:<username>@<Sip Proxy IP or domain name>


[authentication]



[4001]
type=friend
context=outbound
defaultuser=4001
secret=4001
callerid="EXT1"
host=dynamic
nat=no
dtfmode=rfc2833
disallow=all
subscribecontext=outbound
canreinvite=no
allow=all

[4002]
type=friend
context=outbound
defaultuser=4002
secret=4002
callerid="EXT2"
host=dynamic
nat=no
dtfmode=rfc2833
disallow=all
subscribecontext=outbound
canreinvite=no
allow=all




On Sun, Oct 20, 2013 at 11:14 AM, Mitul Limbani <mitul at enterux.in> wrote:

> Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd
> link here.
>
> Mitul
> On Oct 20, 2013 11:07 AM, "akhilesh chand" <omakhileshchand at gmail.com>
> wrote:
>
>> Dear All,
>>
>> I have pri with E1 facility that have 30 line and 100 pri number which is
>> provided by service provider.Number started like 23568561,23568562,23568563
>> and so on. Service provider provide last four digit number for did mapping
>> like 4561,4562,4563.
>>
>>
>> exten => 8561,1,Dial(SIP/4001 at 192.168.14.110,120,tT)
>> exten => 8561,n,hangup()
>>
>> exten => 8562,1,Dial(SIP/4001 at 192.168.14.110,120,tT)
>> exten => 8562,n,hangup()
>>
>> Call comes into first server successful.But problem with second server
>> when call came into second server i got following error:
>>
>> * chan_sip.c:20063 handle_request_invite: Call from '' to extension
>> '4001' rejected because extension not found.*
>>
>> In one more scenario:
>>
>> when i create one extension and call forwarding with this extension that
>> time I'm able to transfer call successful the code is given below:
>>
>> exten => 5001,1,Dial(SIP/4001 at 192.168.14.110,120,tT)
>> exten => 5001,n,hangup()
>>
>>
>> Regards
>> Akhilesh
>>
>> --
>> _____________________________________________________________________
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>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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