20 Apr
2005
20 Apr
'05
5:25 a.m.
Hi!
I'am new in to this VoIP technolgie.
I have a SER 0.9.0 wit Cisco IOS Gatewy.
I only want to configure a registrar Server with a
Cisco gateway, which is connected to the PSTN.
Can someone give me an ser.cfg example ?
How can I tell the ser server that every call,
which starts with 0, route directly to the
IOS Gateway ?
I added two user to the ser db.
The users got a login, password and an email.
Additional to this I added to both user a userloc
with serctl ul add <user> sip:1223@domain.de.
Both users can not phone each other. I got an busy tone
and my Cisco 7960 Phone shows "reorder".
Did I forgot something ?
Here an output of serctl ul show:
eagle[admin] # ./serctl ul show
Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y
===Domain list===
---Domain---
name : 'aliases'
size : 512
table: fcda0dd8
d_ll {
n : 0
first: 0
last : 0
}
---/Domain---
---Domain---
name : 'location'
size : 512
table: fcd9e8b8
d_ll {
n : 2
first: fcda08e8
last : fcda0af8
}
...Record(fcda08e8)...
domain: 'location'
aor : 'ahmad02'
~~~Contact(fcda0978)~~~
domain : 'location'
aor : 'ahmad02'
Contact : 'sip:460@netuse.de'
Expires : -60589
q : 1
Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything'
CSeq : 42
replic : 0
User-Agent: 'SIP Express Router FIFO'
received : ''
State : CS_SYNC
Flags : 128
next : 0
prev : 0
~~~/Contact~~~~
.../Record...
...Record(fcda0af8)...
domain: 'location'
aor : 'ahmad'
~~~Contact(fcda0b88)~~~
domain : 'location'
aor : 'ahmad'
Contact : 'sip:459@netuse.de'
Expires : -60655
q : 1
Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything'
CSeq : 42
replic : 0
User-Agent: 'SIP Express Router FIFO'
received : ''
State : CS_SYNC
Flags : 128
next : 0
prev : 0
~~~/Contact~~~~
.../Record...
---/Domain---
===/Domain list===
Thanks in advance,
Ahmad
--
Ahmad Cheikh-Moussa
NetUSE AG
Dr.-Hell-Straße, 24107 Kiel, Germany
Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499
Service: Service(a)NetUSE.DE -- http://NetUSE.DE/
20 Apr
20 Apr
6:40 a.m.
Ahmad:
There are a number of examples of this in the archives. Search for PSTN
gateway and I'm sure you will find what you need. Basically the gateway
needs
dial-peer statements to "handle" the dialed digits. You'll also want to
define
a sip-ua in the Cisco box where the server points to the address of your
SER proxy. The server address can contain different values including an IP
address, hostname or SRV name.
In your SER proxy you'll want to add conditional statements to test if
the dialed number begins with whatever string indicates the call should go
to the Cisco box, then relay the call to the gateway.
Finally I'd get IP to IP calling through the proxy working first. If
that isn't
working you may have additional issues.
Ahmad Cheikh-Moussa wrote:
Hi!
I'am new in to this VoIP technolgie.
I have a SER 0.9.0 wit Cisco IOS Gatewy.
I only want to configure a registrar Server with a
Cisco gateway, which is connected to the PSTN.
Can someone give me an ser.cfg example ?
How can I tell the ser server that every call,
which starts with 0, route directly to the
IOS Gateway ?
I added two user to the ser db.
The users got a login, password and an email.
Additional to this I added to both user a userloc
with serctl ul add <user> sip:1223@domain.de.
Both users can not phone each other. I got an busy tone
and my Cisco 7960 Phone shows "reorder".
Did I forgot something ?
Here an output of serctl ul show:
eagle[admin] # ./serctl ul show
Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y
===Domain list===
---Domain---
name : 'aliases'
size : 512
table: fcda0dd8
d_ll {
n : 0
first: 0
last : 0
}
---/Domain---
---Domain---
name : 'location'
size : 512
table: fcd9e8b8
d_ll {
n : 2
first: fcda08e8
last : fcda0af8
}
...Record(fcda08e8)...
domain: 'location'
aor : 'ahmad02'
~~~Contact(fcda0978)~~~
domain : 'location'
aor : 'ahmad02'
Contact : 'sip:460@netuse.de'
Expires : -60589
q : 1
Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything'
CSeq : 42
replic : 0
User-Agent: 'SIP Express Router FIFO'
received : ''
State : CS_SYNC
Flags : 128
next : 0
prev : 0
~~~/Contact~~~~
.../Record...
...Record(fcda0af8)...
domain: 'location'
aor : 'ahmad'
~~~Contact(fcda0b88)~~~
domain : 'location'
aor : 'ahmad'
Contact : 'sip:459@netuse.de'
Expires : -60655
q : 1
Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything'
CSeq : 42
replic : 0
User-Agent: 'SIP Express Router FIFO'
received : ''
State : CS_SYNC
Flags : 128
next : 0
prev : 0
~~~/Contact~~~~
.../Record...
---/Domain---
===/Domain list===
Thanks in advance,
Ahmad
--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:blairs@upenn.edu
7:26 a.m.
Ahmad,
I'd suggest surfing to http://onsip.org/ and getting the Getting Started
docs.
Regards,
Paul
On 4/20/05, Ahmad Cheikh-Moussa <acm(a)netuse.de> wrote:
Hi!
I'am new in to this VoIP technolgie.
I have a SER 0.9.0 wit Cisco IOS Gatewy.
I only want to configure a registrar Server with a
Cisco gateway, which is connected to the PSTN.
Can someone give me an ser.cfg example ?
How can I tell the ser server that every call,
which starts with 0, route directly to the
IOS Gateway ?
I added two user to the ser db.
The users got a login, password and an email.
Additional to this I added to both user a userloc
with serctl ul add <user> sip:1223@domain.de.
Both users can not phone each other. I got an busy tone
and my Cisco 7960 Phone shows "reorder".
Did I forgot something ?
Here an output of serctl ul show:
eagle[admin] # ./serctl ul show
Dumping all contacts may take long: are you sure you want to proceed?
[Y|N] y
===Domain list===
---Domain---
name : 'aliases'
size : 512
table: fcda0dd8
d_ll {
n : 0
first: 0
last : 0
}
---/Domain---
---Domain---
name : 'location'
size : 512
table: fcd9e8b8
d_ll {
n : 2
first: fcda08e8
last : fcda0af8
}
...Record(fcda08e8)...
domain: 'location'
aor : 'ahmad02'
~~~Contact(fcda0978)~~~
domain : 'location'
aor : 'ahmad02'
Contact : 'sip:460@netuse.de'
Expires : -60589
q : 1
Call-ID :
'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything'
CSeq : 42
replic : 0
User-Agent: 'SIP Express Router FIFO'
received : ''
State : CS_SYNC
Flags : 128
next : 0
prev : 0
~~~/Contact~~~~
.../Record...
...Record(fcda0af8)...
domain: 'location'
aor : 'ahmad'
~~~Contact(fcda0b88)~~~
domain : 'location'
aor : 'ahmad'
Contact : 'sip:459@netuse.de'
Expires : -60655
q : 1
Call-ID :
'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything'
CSeq : 42
replic : 0
User-Agent: 'SIP Express Router FIFO'
received : ''
State : CS_SYNC
Flags : 128
next : 0
prev : 0
~~~/Contact~~~~
.../Record...
---/Domain---
===/Domain list===
Thanks in advance,
Ahmad
--
Ahmad Cheikh-Moussa
NetUSE AG
Dr.-Hell-Straße, 24107 Kiel, Germany
Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499
Service: Service(a)NetUSE.DE -- http://NetUSE.DE/
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
9:15 a.m.
Hi!
Phone to Phone call functions now properly.
But I still got problems to make an externall
call.
Is this configuration right for stateless forwarding ?
The ip of the gateway is 192.168.254.30.
Here a part of my ser.cfg:
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (msg:len >= max_len ) {
sl_send_reply("513", "Message too big");
break;
};
if (src_ip==193.175.135.0/24){
#force_send_socket(smaug:5080);
forward(193.175.135.179);
break;
}
#if (uri=~"^sip:0[0-9]*@netuse.de") {
# forward(192.168.254.203);
# break;
#}
# Default route zu Cisco Gateway
if (method == "INVITE" && uri=~"^sip:0") {
rewritehostport("192.168.254.203:5060");
t_relay_to_udp("192.168.254.203", "5060");
break;
}
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("iptel.org", "subscriber"))
{
# www_challenge("iptel.org", "0");
# break;
# };
save("location");
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
router configuration:
voice service voip
sip
!
!
voice class codec 2
codec preference 1 g711alaw
!
dial-peer voice 1 pots
description Default-Dial-peer fuer ausgehende Anrufe
preference 3
service session
max-conn 25
destination-pattern 0T
progress_ind alert enable 8
direct-inward-dial
!
dial-peer voice 10 voip
preference 2
destination-pattern 4..
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711alaw
!
sip-ua
set sip-status 401 pstn-cause 127
set sip-status 407 pstn-cause 127
set sip-status 410 pstn-cause 22
set sip-status 415 pstn-cause 127
set sip-status 480 pstn-cause 19
set sip-status 503 pstn-cause 127
set sip-status 580 pstn-cause 127
retry invite 3
retry register 3
timers register 150
registrar ipv4:192.168.254.30 expires 3600
sip-server ipv4:192.168.254.30
!
Thanks,
Ahmad
--
Ahmad Cheikh-Moussa
NetUSE AG
Dr.-Hell-Straße, 24107 Kiel, Germany
Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499
Service: Service(a)NetUSE.DE -- http://NetUSE.DE/
9:21 a.m.
The first thing that caught my attention is the lack of a port in your
pots dial-peer. If this peer is matched where should the call go? Do you
have a VWIC interface in your router?
Ahmad Cheikh-Moussa wrote:
Hi!
Phone to Phone call functions now properly.
But I still got problems to make an externall
call.
Is this configuration right for stateless forwarding ?
The ip of the gateway is 192.168.254.30.
Here a part of my ser.cfg:
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (msg:len >= max_len ) {
sl_send_reply("513", "Message too big");
break;
};
if (src_ip==193.175.135.0/24){
#force_send_socket(smaug:5080);
forward(193.175.135.179);
break;
}
#if (uri=~"^sip:0[0-9]*@netuse.de") {
# forward(192.168.254.203);
# break;
#}
# Default route zu Cisco Gateway
if (method == "INVITE" && uri=~"^sip:0") {
rewritehostport("192.168.254.203:5060");
t_relay_to_udp("192.168.254.203", "5060");
break;
}
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("iptel.org", "subscriber"))
{
# www_challenge("iptel.org", "0");
# break;
# };
save("location");
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
router configuration:
voice service voip
sip
!
!
voice class codec 2
codec preference 1 g711alaw
!
dial-peer voice 1 pots
description Default-Dial-peer fuer ausgehende Anrufe
preference 3
service session
max-conn 25
destination-pattern 0T
progress_ind alert enable 8
direct-inward-dial
!
dial-peer voice 10 voip
preference 2
destination-pattern 4..
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711alaw
!
sip-ua
set sip-status 401 pstn-cause 127
set sip-status 407 pstn-cause 127
set sip-status 410 pstn-cause 22
set sip-status 415 pstn-cause 127
set sip-status 480 pstn-cause 19
set sip-status 503 pstn-cause 127
set sip-status 580 pstn-cause 127
retry invite 3
retry register 3
timers register 150
registrar ipv4:192.168.254.30 expires 3600
sip-server ipv4:192.168.254.30
!
Thanks,
Ahmad
--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:blairs@upenn.edu
7093
days inactive
7093
days old
4 comments
3 participants
participants (3)
-
Ahmad Cheikh-Moussa -
Java Rockx -
Steve Blair