VoIP Mailing List Archives

[sams at ticoon.com]

Hello, everyone.

Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.

For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to call F, and it will ring. Voice from softphone to F carries
over and I can hear it; however, no voice from F to softphone will carry. I
have been experimenting with different codec and other cisco/asterisk config
tips from the web. None had worked so far.

If anyone have experienced such problem and knows how to solve this, I will
be eternally grateful.

< sip.conf >
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
disallow = all
nat=yes
canreinvite=yes
allowguest=no
allow=ulaw
allow=alaw
allow=g711
allow=g729
allow=gsm
allow=ilbc
[2000]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[2001]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[2002]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[2003]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[xxx.xxx.xxx.yyy]
context=pstn-incoming
type=friend
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very

[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001
insecure=very

< extensions.conf >
[my-phones]
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)
exten => 2002,1,Dial(SIP/2002)
exten => 2003,1,Dial(SIP/2003)
exten => 6000,1,MeetMe(600,i,54321)
;include => lan-phones

[bogon-calls]
exten => _.,1,Congestion

[pstn-incoming]
include => lan-phones

[local-phones]
include => lan-phones
include => pstn-outbound

[pstn-outbound]
; Calls starting with 9 have the 9 stripped & are then routed out to the
PSTN
exten => _9.,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx.yyy) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
;exten => _9XXXX,1,Dial,SIP/${EXTEN}@xxx.xxx.xxx.yyy ; IP address of Cisco
gateway
;exten => _9XXXX,2,Congestion
exten => _9.,2,Congestion

[lan-phones]
exten => 1001,1,Dial(SIP/1001,20)
exten => 1001,2,Voicemail(u1001)
exten => 1001,3,Answer(SIP/1001)
exten => 1001,102,Voicemail(b1001)
exten => 1001,103,Hangup

< Cisco 2611 config >

Building configuration...

Current configuration : 2030 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname fxroute
!
logging queue-limit 100
enable secret
enable password
!
clock timezone GMT 0
ip subnet-zero
no ip routing
!
!
!
ip audit notify log
ip audit po max-events 100
!
!
!
!
!
voice rtp send-recv
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 gsmefr
codec preference 4 gsmfr
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
mta receive maximum-recipients 0
!
!
!
!
interface Ethernet0/0
ip address xxx.xxx.xxx.yyy 255.255.255.0
no ip route-cache
no ip mroute-cache
full-duplex
no cdp enable
!
interface Ethernet0/1
no ip address
no ip route-cache
no ip mroute-cache
shutdown
half-duplex
no cdp enable
!
ip http server
no ip http secure-server
ip classless
!
!
!
!
call rsvp-sync
!
voice-port 1/0/0
input gain 10
output attenuation 10
no comfort-noise
connection plar opx 1001
station-id number 100
caller-id enable
!
voice-port 1/0/1
input gain 10
output attenuation 10
no comfort-noise
caller-id enable
!
voice-port 1/1/0
!
voice-port 1/1/1
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 100 pots
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 8
port 1/0/0
!
dial-peer voice 2 voip
destination-pattern 1...
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx:5060
session transport udp
dtmf-relay h245-alphanumeric
clid strip
no vad
!
dial-peer voice 1 pots
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:xxx.xxx.xxx.xxx
!
!
!
telephony-service
transfer-pattern ....
transfer-system full-blind
!
!
line con 0
exec-timeout 0 0
line aux 0
line vty 0 4
password
login
!
!
end

Thank you

Sang-Kil (Sam) Suh
System administrator

--
Ticoon Technology Inc.

[samtam888 at gmail.com]

Are you using NAT?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Voice only works from one way.

Hello, everyone.

Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.

For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to call F, and it will ring. Voice from softphone to F carries
over and I can hear it; however, no voice from F to softphone will carry. I
have been experimenting with different codec and other cisco/asterisk config
tips from the web. None had worked so far.

If anyone have experienced such problem and knows how to solve this, I will
be eternally grateful.

< sip.conf >
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
disallow = all
nat=yes
canreinvite=yes
allowguest=no
allow=ulaw
allow=alaw
allow=g711
allow=g729
allow=gsm
allow=ilbc
[2000]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[2001]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[2002]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[2003]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[xxx.xxx.xxx.yyy]
context=pstn-incoming
type=friend
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very

[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001
insecure=very

< extensions.conf >
[my-phones]
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)
exten => 2002,1,Dial(SIP/2002)
exten => 2003,1,Dial(SIP/2003)
exten => 6000,1,MeetMe(600,i,54321)
;include => lan-phones

[bogon-calls]
exten => _.,1,Congestion

[pstn-incoming]
include => lan-phones

[local-phones]
include => lan-phones
include => pstn-outbound

[pstn-outbound]
; Calls starting with 9 have the 9 stripped & are then routed out to the
PSTN
exten => _9.,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx.yyy) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
;exten => _9XXXX,1,Dial,SIP/${EXTEN}@xxx.xxx.xxx.yyy ; IP address of Cisco
gateway
;exten => _9XXXX,2,Congestion
exten => _9.,2,Congestion

[lan-phones]
exten => 1001,1,Dial(SIP/1001,20)
exten => 1001,2,Voicemail(u1001)
exten => 1001,3,Answer(SIP/1001)
exten => 1001,102,Voicemail(b1001)
exten => 1001,103,Hangup

< Cisco 2611 config >

Building configuration...

Current configuration : 2030 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname fxroute
!
logging queue-limit 100
enable secret
enable password
!
clock timezone GMT 0
ip subnet-zero
no ip routing
!
!
!
ip audit notify log
ip audit po max-events 100
!
!
!
!
!
voice rtp send-recv
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 gsmefr
codec preference 4 gsmfr
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
mta receive maximum-recipients 0
!
!
!
!
interface Ethernet0/0
ip address xxx.xxx.xxx.yyy 255.255.255.0
no ip route-cache
no ip mroute-cache
full-duplex
no cdp enable
!
interface Ethernet0/1
no ip address
no ip route-cache
no ip mroute-cache
shutdown
half-duplex
no cdp enable
!
ip http server
no ip http secure-server
ip classless
!
!
!
!
call rsvp-sync
!
voice-port 1/0/0
input gain 10
output attenuation 10
no comfort-noise
connection plar opx 1001
station-id number 100
caller-id enable
!
voice-port 1/0/1
input gain 10
output attenuation 10
no comfort-noise
caller-id enable
!
voice-port 1/1/0
!
voice-port 1/1/1
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 100 pots
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 8
port 1/0/0
!
dial-peer voice 2 voip
destination-pattern 1...
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx:5060
session transport udp
dtmf-relay h245-alphanumeric
clid strip
no vad
!
dial-peer voice 1 pots
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:xxx.xxx.xxx.xxx
!
!
!
telephony-service
transfer-pattern ....
transfer-system full-blind
!
!
line con 0
exec-timeout 0 0
line aux 0
line vty 0 4
password
login
!
!
end

Thank you

Sang-Kil (Sam) Suh
System administrator

--
Ticoon Technology Inc.


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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Register Now: http://www.astricon.net

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[sams at ticoon.com]

Yes, both Asterisk and Cisco are behind Nat.
On 6/20/08 3:26 PM, "Sam Tam" <samtam888 at gmail.com> wrote:

[fgarcia at systeamusa.com]

I was never able to get it to work that way. When I had Asterisk in NAT I
was able to make calls, but most of the times they were one way voice.



I was able to get two-way voice when I configured the remote phone using
STUN and Symetrical RTP. However, the calls dropped every 19-20 seconds. I
read several threads online, but nobody explained the requirements in
details. Everything works fine if you have a public IP address or DMZ on
Asterisk.



Good luck and please let me know if you get it up and running.



Fidel Garcia

System Engineer



sysTeam.

7205 NW 19th Street, Suite 302
Miami, Florida 33126

Email: fgarcia at systeamusa.com

Tel: (305)-477-7303 Fax: (305)-477-0013

http://www.systeamusa.com



From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Friday, June 20, 2008 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice only works from one way.



Yes, both Asterisk and Cisco are behind Nat.
On 6/20/08 3:26 PM, "Sam Tam" <samtam888 at gmail.com> wrote:


Are you using NAT?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Voice only works from one way.

Hello, everyone.

Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.

For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to call F, and it will ring. Voice from softphone to F carries
over and I can hear it; however, no voice from F to softphone will carry. I
have been experimenting with different codec and other cisco/asterisk config
tips from the web. None had worked so far.

If anyone have experienced such problem and knows how to solve this, I will
be eternally grateful.

< sip.conf >
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
disallow = all
nat=yes
canreinvite=yes
allowguest=no
allow=ulaw
allow=alaw
allow=g711
allow=g729
allow=gsm
allow=ilbc


[2000]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[2001]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[2002]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[2003]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[xxx.xxx.xxx.yyy]
context=pstn-incoming
type=friend
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very

[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001
insecure=very

< extensions.conf >
[my-phones]
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)
exten => 2002,1,Dial(SIP/2002)
exten => 2003,1,Dial(SIP/2003)
exten => 6000,1,MeetMe(600,i,54321)
;include => lan-phones

[bogon-calls]
exten => _.,1,Congestion

[pstn-incoming]
include => lan-phones

[local-phones]
include => lan-phones
include => pstn-outbound

[pstn-outbound]
; Calls starting with 9 have the 9 stripped & are then routed out to the
PSTN
exten => _9.,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx.yyy) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
;exten => _9XXXX,1,Dial,SIP/${EXTEN}@xxx.xxx.xxx.yyy ; IP address of Cisco
gateway
;exten => _9XXXX,2,Congestion
exten => _9.,2,Congestion

[lan-phones]
exten => 1001,1,Dial(SIP/1001,20)
exten => 1001,2,Voicemail(u1001)
exten => 1001,3,Answer(SIP/1001)
exten => 1001,102,Voicemail(b1001)
exten => 1001,103,Hangup

< Cisco 2611 config >

Building configuration...

Current configuration : 2030 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname fxroute
!
logging queue-limit 100
enable secret
enable password
!
clock timezone GMT 0
ip subnet-zero
no ip routing
!
!
!
ip audit notify log
ip audit po max-events 100
!
!
!
!
!
voice rtp send-recv
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 gsmefr
codec preference 4 gsmfr
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
mta receive maximum-recipients 0
!
!
!
!
interface Ethernet0/0
ip address xxx.xxx.xxx.yyy 255.255.255.0
no ip route-cache
no ip mroute-cache
full-duplex
no cdp enable
!
interface Ethernet0/1
no ip address
no ip route-cache
no ip mroute-cache
shutdown
half-duplex
no cdp enable
!
ip http server
no ip http secure-server
ip classless
!
!
!
!
call rsvp-sync
!
voice-port 1/0/0
input gain 10
output attenuation 10
no comfort-noise
connection plar opx 1001
station-id number 100
caller-id enable
!
voice-port 1/0/1
input gain 10
output attenuation 10
no comfort-noise
caller-id enable
!
voice-port 1/1/0
!
voice-port 1/1/1
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 100 pots
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 8
port 1/0/0
!
dial-peer voice 2 voip
destination-pattern 1...
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx:5060
session transport udp
dtmf-relay h245-alphanumeric
clid strip
no vad
!
dial-peer voice 1 pots
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:xxx.xxx.xxx.xxx
!
!
!
telephony-service
transfer-pattern ....
transfer-system full-blind
!
!
line con 0
exec-timeout 0 0
line aux 0
line vty 0 4
password
login
!
!
end

Thank you

Sang-Kil (Sam) Suh
System administrator

--
Ticoon Technology Inc.


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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 --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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



Thank you

Sang-Kil (Sam) Suh
System administrator

--
Ticoon Technology Inc.
56 The Esplanade, Suite 404
Toronto, Ontario
M5E 1A7

Tel: (416) 513-9524 (ext. 299)
Cell: (416) 902-2890
Fax: (416) 513-9525

No virus found in this incoming message.
Checked by AVG.
Version: 8.0.100 / Virus Database: 270.4.1/1510 - Release Date: 6/19/2008
3:21 PM

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[samtam888 at gmail.com]

Well to be honest, our experience with asterisk never works with under NAT.
if you got DMZ then it will otherwise don't hold your breath for it.

If you want to use it as a production server

Your option is 1. Get a Real IP

2. there is no 2 really just get an ReaL Public IP
Sam



_____

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Fidel Garcia
Sent: Saturday, June 21, 2008 6:19 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Voice only works from one way.



I was never able to get it to work that way. When I had Asterisk in NAT I
was able to make calls, but most of the times they were one way voice.



I was able to get two-way voice when I configured the remote phone using
STUN and Symetrical RTP. However, the calls dropped every 19-20 seconds. I
read several threads online, but nobody explained the requirements in
details. Everything works fine if you have a public IP address or DMZ on
Asterisk.



Good luck and please let me know if you get it up and running.



Fidel Garcia

System Engineer



sysTeam.

7205 NW 19th Street, Suite 302
Miami, Florida 33126

Email: fgarcia at systeamusa.com

Tel: (305)-477-7303 Fax: (305)-477-0013

http://www.systeamusa.com



From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Friday, June 20, 2008 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice only works from one way.



Yes, both Asterisk and Cisco are behind Nat.
On 6/20/08 3:26 PM, "Sam Tam" <samtam888 at gmail.com> wrote:


Are you using NAT?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]
<mailto:asterisk-users-bounces at lists.digium.com%5d> On Behalf Of Sang-Kil
(Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Voice only works from one way.

Hello, everyone.

Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.

For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to call F, and it will ring. Voice from softphone to F carries
over and I can hear it; however, no voice from F to softphone will carry. I
have been experimenting with different codec and other cisco/asterisk config
tips from the web. None had worked so far.

If anyone have experienced such problem and knows how to solve this, I will
be eternally grateful.

< sip.conf >
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
disallow = all
nat=yes
canreinvite=yes
allowguest=no
allow=ulaw
allow=alaw
allow=g711
allow=g729
allow=gsm
allow=ilbc


[2000]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[2001]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[2002]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[2003]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic

[xxx.xxx.xxx.yyy]
context=pstn-incoming
type=friend
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very

[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001
insecure=very

< extensions.conf >
[my-phones]
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)
exten => 2002,1,Dial(SIP/2002)
exten => 2003,1,Dial(SIP/2003)
exten => 6000,1,MeetMe(600,i,54321)
;include => lan-phones

[bogon-calls]
exten => _.,1,Congestion

[pstn-incoming]
include => lan-phones

[local-phones]
include => lan-phones
include => pstn-outbound

[pstn-outbound]
; Calls starting with 9 have the 9 stripped & are then routed out to the
PSTN
exten => _9.,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx.yyy) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
;exten => _9XXXX,1,Dial,SIP/${EXTEN}@xxx.xxx.xxx.yyy ; IP address of Cisco
gateway
;exten => _9XXXX,2,Congestion
exten => _9.,2,Congestion

[lan-phones]
exten => 1001,1,Dial(SIP/1001,20)
exten => 1001,2,Voicemail(u1001)
exten => 1001,3,Answer(SIP/1001)
exten => 1001,102,Voicemail(b1001)
exten => 1001,103,Hangup

< Cisco 2611 config >

Building configuration...

Current configuration : 2030 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname fxroute
!
logging queue-limit 100
enable secret
enable password
!
clock timezone GMT 0
ip subnet-zero
no ip routing
!
!
!
ip audit notify log
ip audit po max-events 100
!
!
!
!
!
voice rtp send-recv
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 gsmefr
codec preference 4 gsmfr
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
mta receive maximum-recipients 0
!
!
!
!
interface Ethernet0/0
ip address xxx.xxx.xxx.yyy 255.255.255.0
no ip route-cache
no ip mroute-cache
full-duplex
no cdp enable
!
interface Ethernet0/1
no ip address
no ip route-cache
no ip mroute-cache
shutdown
half-duplex
no cdp enable
!
ip http server
no ip http secure-server
ip classless
!
!
!
!
call rsvp-sync
!
voice-port 1/0/0
input gain 10
output attenuation 10
no comfort-noise
connection plar opx 1001
station-id number 100
caller-id enable
!
voice-port 1/0/1
input gain 10
output attenuation 10
no comfort-noise
caller-id enable
!
voice-port 1/1/0
!
voice-port 1/1/1
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 100 pots
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 8
port 1/0/0
!
dial-peer voice 2 voip
destination-pattern 1...
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx:5060
session transport udp
dtmf-relay h245-alphanumeric
clid strip
no vad
!
dial-peer voice 1 pots
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:xxx.xxx.xxx.xxx
!
!
!
telephony-service
transfer-pattern ....
transfer-system full-blind
!
!
line con 0
exec-timeout 0 0
line aux 0
line vty 0 4
password
login
!
!
end

Thank you

Sang-Kil (Sam) Suh
System administrator

--
Ticoon Technology Inc.


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_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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Register Now: http://www.astricon.net

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



Thank you

Sang-Kil (Sam) Suh
System administrator

--
Ticoon Technology Inc.
56 The Esplanade, Suite 404
Toronto, Ontario
M5E 1A7

Tel: (416) 513-9524 (ext. 299)
Cell: (416) 902-2890
Fax: (416) 513-9525

No virus found in this incoming message.
Checked by AVG.
Version: 8.0.100 / Virus Database: 270.4.1/1510 - Release Date: 6/19/2008
3:21 PM

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[fred at teamforrest.com]

Have you tried keeping asterisk in on the call with a /n connection in
the dial-plan?

Is there any firewall that is blocking udp ports to any of your clients?
Fred Posner
fred at teamforrest.com



On Jun 21, 2008, at 12:36 AM, Sam Tam wrote:

[david at linuxcrazy.com]