I have integrate SER + Asterisk http://linuxbug.org/index_files/projects.html

Satish Patel
feel free contact :- satish.lx@gmail.com

service password-encryption
service timestamps debug

hostname Aliwei

enable password 7 z1SUMF6JUPe1pf5P

username admin password 7 y1CRyR4p8kJudB
username sapian password 7 VhVcZNYcNKqjggK
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8

voice vad-time 300
voice rtp start-port 10001
voice pattern sip

interface fastethernet 0/0
ip address 172.16.2.11 255.255.255.0

interface ethernet 0/0
ip address 192.168.140.11 255.255.255.0

interface async 0/0

line vty 0 31

sip-ua
transport udp
ip local 172.16.2.11
ip media 172.16.2.11
registrar ipv4:172.16.2.1
mode end-to-end
remote-port 5060
enable-use-rport
ring-mode ring-mode-183
enable-same-number

ip route 0.0.0.0 0.0.0.0 172.16.2.1
ip name-server 172.16.2.1

voice-port 1/0
port-type FXO
busytone 480 300 300
signal groundStart
pre-dial-delay 1
enable-rfc2833
fxo-number 3113343083
local_zone_number 183

voice-port 1/1
port-type FXO
busytone 400 300 300
pre-dial-delay 1
enable-rfc2833
local_zone_number 183

voice-port 1/2
port-type FXO
pre-dial-delay 1
enable-rfc2833

voice-port 1/3
port-type FXO
pre-dial-delay 1
enable-rfc2833

voice-port 1/4
port-type FXO
pre-dial-delay 1
enable-rfc2833

voice-port 1/5
port-type FXO
pre-dial-delay 1
enable-rfc2833

voice-port 1/6
port-type FXO
pre-dial-delay 1
enable-rfc2833

voice-port 1/7
port-type FXO
pre-dial-delay 1
enable-rfc2833

dial-peer voice 1 pots
description fxo0
destination-pattern T
port 1/0
no register e164
no vad
forward-digits all
caller-id

dial-peer voice 2 pots
destination-pattern T
port 1/1
no register e164
no vad
forward-digits all
caller-id

dial-peer voice 3 pots
destination-pattern T
port 1/2
no register e164
no vad
caller-id
call-forward 673
shutdown

dial-peer voice 4 pots
destination-pattern T
port 1/3
no register e164
no vad
caller-id
call-forward 674
shutdown

dial-peer voice 5 pots
destination-pattern T
port 1/4
no register e164
no vad
caller-id
call-forward 675
shutdown

dial-peer voice 6 pots
destination-pattern T
port 1/5
no register e164
no vad
caller-id
call-forward 676
shutdown

dial-peer voice 7 pots
destination-pattern T
port 1/6
no register e164
no vad
caller-id
call-forward 677
shutdown

dial-peer voice 8 pots
destination-pattern T
port 1/7
no register e164
no vad
caller-id
call-forward 678
shutdown

dial-peer voice 100 voip
destination-pattern T
codec g711alaw
voice-class codec 1
session target ipv4:172.16.2.1
dtmf-relay cisco-rtp
no vad
transfer-mode blind

ntp source ethernet 0/0
ntp server 172.16.2.1
ntp clock-period 0
media-wait-for-connect
tone-generate busy f1 440 f2 0 f3 0 f4 0 on-time 500 off-time 500 numCad 1 repCounter 0 on_time_2 0 off_time_2 0 on_time_3 0 off_time_3 0 tonePwr1 12 tonePwr2 0 tonePwr3 0 tonePwr4 0
tone-generate ring-back f1 440 f2 0 f3 0 f4 0 on-time 500 off-time 500 numCad 1 repCounter 0 on_time_2 0 off_time_2 0 on_time_3 0 off_time_3 0 tonePwr1 12 tonePwr2 0 tonePwr3 0 tonePwr4 0
tone-generate dial f1 380 f2 0 f3 0 f4 0 on-time 2400 off-time 2400 numCad 1 repCounter 1 on_time_2 0 off_time_2 0 on_time_3 0 off_time_3 0 tonePwr1 2 tonePwr2 0 tonePwr3 0 tonePwr4 0
trunk-to-voip
voice-bind sip

end

I have to integarte SER with Asterisk means SIP client Register on SER and make call to PSTN with the help of asterisk and also can do meet me confrance and many PBX feature is it possible and How

hi my name is habib and looking to this informations plz:
first i am new user of sip and voip i am in canada and i want to call our office out of canada in dubai for example if i have an internet connection or a dsl connection in my office there i ahve a ata device here in canada and dubai and i have sip software on both computer what do i need to call dubai over voip or sip and i want to connect my land line in dubai to my device or modem if my computer so i can make calls to dubai via voip or sip but i will pay local call not international because i am using my dubai land line is that posssible or i am dreaming i hope i can find something thx for u all plz send me email to hayik@hotmail.com

Have any company use Ser to provide Prepaid internet phone service?
my company is looking for Prepaid internet phone solution.
Thank you, phanvanduc@gmail.com

R there any detailed docs on Ser+asterisk integration .

If you're trying to implement SER on your VoIP network, feel free to contact us voipinfo@idv.net or through our website http://www.idv.net
We have extensive background in implementing SER with Cisco voice gateways, Asterisk, ATAs, etc.
We can also help with custom development.

we are looking for a linux based programmer for improving our SIP proxy Project
sip professional programmer please contact me at grkashani@yahoo.com

Hi,

Can anyone expand on this comment (SER supports SIP connections with more features and more scalability than Asterisk. Normally, SER would be used in conjunction with Asterisk when a SIP phone needed to connect to the PSTN) so that a first-time Asterisk novice can understand the logic of how SER and Asterisk work together to connect to the PSTN? What functions/processes is Asterisk responsible for and what functions/processes is SER responsible for? What handoffs are going on? Thank You, dave_rep@yahoo.com

Hi,
We are investigating into integrating the two. Do you know if Asterisk supports VXML? If not, do you have any pointers or info on how the integration can be done?

Thanks.