Call in and out of same SIP Trunk

[rickuk]

HI,

I have a situation where some users are using the full DDI to call and internal extension, because obviously its easier to remember the DDI number that a 3 digit extension, but forgetting that argument, the call gets dropped,

So user calls DDI number this goes out on the SIP trunk, the SIP proxy then sends it back in, will be the same SIP Credintials for the outbound then inbound call. The ISP has said the call disconnect comes from the IP office, here is a treace.


15:36:26 1987147044mS SIP Rx: UDP 91.151.2.130:5060 -> 192.192.200.185:5060
INVITE sip:01274207220@195.11.160.138:5060;transport=udp SIP/2.0
Record-Route: <sip:91.151.2.130;lr=on;ftag=756f374e208b35b3>
Via: SIP/2.0/UDP 91.151.2.130;branch=z9hG4bKd8bc.74a9531.0
Via: SIP/2.0/UDP 195.11.160.138:5060;received=195.11.160.138;rport=5060;branch=z9hG4bK7c30a789e9842e9cae223795e7622087
From: "Brd Remote 281" <sip:xxxxxxxx@sip.voip-unlimited.net>;tag=756f374e208b35b3
To: <sip:01274207220@sip.voip-unlimited.net>
Call-ID: 0659d49cdcd6e577c47fe1fca8b503bf
CSeq: 1761580069 INVITE
P-Asserted-Identity: "Brd Remote 281" <sip:xxxxxxxxxx@195.11.xx.xx:5060>
Contact: "Brd Remote 281" <sip:281@195.11.xx.xx:5060;transport=udp>
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 9.1.2.0 build 91
Content-Type: application/sdp
Content-Length: 303

v=0
o=UserA 2200844316 1887893035 IN IP4 195.11.xx.xx s=Session SDP
c=IN IP4 195.11.xx.xx t=0 0
m=audio 46750 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
15:36:26 1987147051mS SIP Tx: UDP 192.192.xx.xx:5060 -> 91.151.xx.xx:5060
CANCEL sip:01274207xxxx@sip.voip-unlimited.net SIP/2.0
Via: SIP/2.0/UDP 195.11.xx.xx:5060;rport;branch=z9hG4bK7c30a789e9842e9cae223795e7622087
From: "Brd Remote 281" <sip:01274772xxx@sip.voip-unlimited.net>;tag=756f374e208b35b3
To: <sip:01274207xxx@sip.voip-unlimited.net>
Call-ID: 0659d49cdcd6e577c47fe1fca8b503bf
CSeq: 1761580069 CANCEL
Max-Forwards: 70
Proxy-Authorization: Digest username="01274772xx",realm="sip.voip-unlimited.net",nonce="56c492bb00003b1ba9c1a5be4c3ca05edec130c373dacdc0",response="fd3a311a7b5470b3d84616c25305afdc",uri="sip:01274207220@sip.voip-unlimited.net"
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Reason: Q.850;cause=16;text="Normal call clearing"
User-Agent: IP Office 9.1.2.0 build 91
Content-Length: 0

15:36:26 1987147066mS SIP Rx: UDP 91.151.xx.xx-> 192.192.xx.xx:5060
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 195.11.xx.xx:5060;rport=5060;branch=z9hG4bK7c30a789e9842e9cae223795e7622087
From: "Brd Remote 281" <sip:012747727xx@sip.voip-unlimited.net>;tag=756f374e208b35b3
To: <sip:012742072xx@sip.voip-unlimited.net>;tag=4863a7036cb78b731105060715005165-8a7f
Call-ID: 0659d49cdcd6e577c47fe1fca8b503bf
CSeq: 1761580069 CANCEL
Server: VoIP-Unlimited SIP Proxy
Content-Length: 0

15:36:26 1987147069mS SIP Rx: UDP 91.151.x.xx:5060 -> 192.192.x.xx:5060
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 195.11.xx.xx:5060;rport=5060;branch=z9hG4bK7c30a789e9842e9cae223795e7622087
From: "Brd Remote 281" <sip:01274772xxx@sip.voip-unlimited.net>;tag=756f374e208b35b3
To: <sip:01274207xxx@sip.voip-unlimited.net>;tag=4863a7036cb78b731105060715005165-8a7f
Call-ID: 0659d49cdcd6e577c47fe1fca8b503bf
CSeq: 1761580069 INVITE
Server: VoIP-Unlimited SIP Proxy
Content-Length: 0

 

[joe2938]

so just to be clear here, instead of just dialing the 3 digit extension which is the best way, they instead dial the did number which routes out of the pbx and then back in to the extension. This also ties up 2 trunks for 1 call which is totally crazy if this is the case

 

[Pepp77]

Crazy, but at least with it being SIP to SIP there should be no charges

| ACSS SME |

 

[joe2938]

so they can't remember a 3 digit extension but can remember a 10 digit did number?

 

[derfloh]

Why not create a short code 1234567xxx/dial extn/N ?

 

[rickuk]

I accept I can create short codes etc, 200 DDI's and extensions going to be fun, I was just hoping to find a technical answer why it doesn't work in the first place.

 

[amriddle01]

It works normally, so I guess it's down to your programming or the way the provider handles it. From that trace it looks like the system sends the Cancel but there isn't enough in the trace to show what triggered the cancel as you've excluded a lot of data from it

 

[Pepp77]

I don't understand why there are 3 IP addresses in the trace, all SIP I have ever worked on just shows the two (local and remote)

| ACSS SME |

 

[amriddle01]

I saw that, assuming it's an SBC/proxy

 

[rickuk]

The IP Office is behind a firewall, the third IP is the internal IP 192.192.x.x is the internal IP, could NAT be casuing the issue?

 

[Pepp77]

Nust be a proxy setup of some description as with an SBC you usually still get the two IPs one the phone system and one the SBC clean side address.

| ACSS SME |

 

[rickuk]

Ill run some STUn tests and see what I get, could be the SonicWALL but Ive disabled the SIP Transformation settings

 

[amriddle01]

I think your firewall has an ALG of some kind, it's IP shouldn't show in SIP packets

 

[rickuk]

out of interest how do you set up stun and sip

I have several installs and have always set up the same way, going back and testing all site have the same issue. My set up is usually a Dell SonicWALL as firewall.

I create a NAT rule with 1 static external IP address mapped to the internal IP office IP address with firewall rules also locking down the ports mapped through.
- If the firewall rule is set to only allow from the STUN server IP and SIP proxy IP running stun on the system/LAN/Network topology will detect "port restricted con nat" if I set to allow any IP it sets to Full cone nat" it always populates the correct public IP address and port. I tend to end up setting this manually to Full cone nat. Under my line settings under transport I set protocol to UDP and use network topology from the LAN interface.

IS this correct?

 

[Pepp77]

If it detects Port Restricted Cone NAT then you do not want to set it to Full cone NAT, you want to set it to Port Restricted Cone NAT.

I use Sonicwalls as well (generally)

I will have the following setup

Address Objects

Internal IP address of IPO
Public IP address of IPO
SIP Server IP Address(es)

Service Objects - all UDP

5060
3478-3479
49152-53246

Service Group
SIP Services - this has the 3 above objects in it

Then under firewall I will have

One incoming rule

Source will be the SIP Server IP Address(es)
Destination will the the Public IP address of the IPO
Service will be the SIP Services Group

This allows traffic from the SIP providers on the SIP ports through the firewall.

Then under NAT I will have two entries

1. Outgoing
Source Original - Internal IP Address of IPO
Source Translated - Public IP address of IPO
Destination Original - Any
Destination Translated - Original
Service Original - Any
Service Translated - Original
Interface Inbound - Any
Interface Outbound - Any


2. Incoming
Source Original - Any
Source Translated - Original
Destination Original - Public IP Address of IPO
Destination Translated - Private IP Address of IPO
Service Original - SIP Services Group
Service Translated - Original
Interface Inbound - Any
Interface Outbound - Any

I do not lock the Source down on the NAT as the firewall has already done that.

Then on the VOiP tab

Consistent NAT ticked
SIP Transformation - Unticked
H323 Transformation - Unticked


With the above in place my STUN tab would then be

| ACSS SME |