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

2013-10-19 Thread 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 => ::@


[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  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" 
> 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@192.168.14.110,120,tT)
>> exten => 8561,n,hangup()
>>
>> exten => 8562,1,Dial(SIP/4001@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@192.168.14.110,120,tT)
>> exten => 5001,n,hangup()
>>
>>
>> Regards
>> Akhilesh
>>
>> --
>> _
>> -- 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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- 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

asterisk-users mailing list
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   http://lists.digium.com/mailman/listinfo/asterisk-users

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

2013-10-19 Thread akhilesh chand
Server A ( which contain pri line)

*chan_dahdi.conf*

[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=no
echocancelwhenbridged=no
rxgain=0.5
txgain=0.5
relaxdtmf=yes
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]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=all
nat=yes
callerid = LITE
externip=
externhost=
autocreatepeer=yes
autodomain=yes
localnet=192.168.53.197/255.255.255.0
canreinvite=yes
language=En
allowtransfer=yes
realm=telunet
domain=192.168.53.197
maxexpiry=3600
defaultexpiry=200
useragent=LITE PBX
usereqphone = yes
dtmfmode = rfc2833
alwaysauthreject = no
regcontext=sipregistrations
rtptimeout=60
rtpholdtimeout=300
rtcachefriends=yes
;--- SIP DEBUGGING
---
sipdebug = yes
registertimeout=60
registerattempts=5
callgroup=1
pickupgroup=1
callevents=yes

;register => ::@


[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

[4003]
type=friend
context=outbound
defaultuser=4003
secret=4003
callerid="EXT3"
host=dynamic
nat=no
dtfmode=rfc2833
disallow=all
subscribecontext=outbound
canreinvite=no
allow=all

[4004]
type=friend
context=outbound
defaultuser=4004
secret=4004
callerid="EXT4"
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  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" 
> 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@192.168.14.110,120,tT)
>> exten => 8561,n,hangup()
>>
>> exten => 8562,1,Dial(SIP/4001@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@192.168.14.110,120,tT)
>> exten => 5001,n,hangup()
>>
>>
>> Regards
>> Akhilesh
>>
>> --
>> _
>> -- 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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

2013-10-19 Thread Mitul Limbani
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" 
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@192.168.14.110,120,tT)
> exten => 8561,n,hangup()
>
> exten => 8562,1,Dial(SIP/4001@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@192.168.14.110,120,tT)
> exten => 5001,n,hangup()
>
>
> Regards
> Akhilesh
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

2013-10-19 Thread akhilesh chand
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@192.168.14.110,120,tT)
exten => 8561,n,hangup()

exten => 8562,1,Dial(SIP/4001@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@192.168.14.110,120,tT)
exten => 5001,n,hangup()


Regards
Akhilesh
-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users