SIP Configuration 2

[johnromani]

hi Guys,

Its driving me crazy... i have benn configured almost the same config like that, but now I can get any errors.

To resume the scene, I got this error:

SIP/2.0 480 No Routes Found
Via: SIP/2.0/UDP 192.168.1.32:5060;received=200.158.234.xx;branch=z9hG4bK55c2b144e81d7af270f3230ed09532db;rport=5060
From: "551149494949" <sip:551149494949@189.125.123.xx>;tag=faf504c3be374e55
To: <sip:5555002272@189.125.123.xx>;tag=aprqngfrt-aaf4s7g5jito8

The from an to fields using the same IP Address (189....)

What am I doing wrong?

I put on the SIP URI the complete Number (551149494949) (these are not the real numbers)

Regards,


Cesar Romani
Avaya ACSS | APSS (SME/Data)| ACIS (Data)
Comptia Convergence +

 

[johnromani]

Anyone??

I need to know how a change the ip address in "from" field.

Cesar Romani
Avaya ACSS | APSS (SME/Data)| ACIS (Data)
Comptia Convergence +

 

[Bas1234]

On lan 1 or 2 fill in the stun part. Use stunserver.org or a stun server from you provider. On the trunk voip tab set the lan 1or2.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

 

[johnromani]

Heçlo Bas1234,

i tried to change from LAN1 to LAN2 and any others configurations but nothing works.
I tried another sip trunk and the characteristics are the same but on this one it works well, see:

SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.1.32:5060;rport=5060;branch=z9hG4bK110e06e15d01fdd942d0696fca27242a
From: "1123911089" <sip:1123911089@177.38.216.32>;tag=25fce1577d1d9051
To: <sip:55052171@177.38.216.32>
Call-ID: 1de2ead3b94b5cc706b04a34af93a942@192.168.1.32
CSeq: 1382131788 INVITE
Server: TSP-2.2
Content-Length: 0
Warning: 392 177.38.216.32:5060 "Noisy feedback tells: pid=52973 req_src_ip=200.158.234.55 req_src_port=5060 in_uri=sip:55052171@177.38.216.32 out_uri=sip:551155052171@177.38.220.16:5060;transport=udp via_cnt==1"

172483mS CMARS: LINE ep Received: CMProceeding - child->state = CMCSOffering - ARS Call State = CMCSOverlapRecv
172981mS SIP Call Rx: 18
SIP/2.0 180 Ringing
From: <sip:1123911089@177.38.216.32>;tag=25fce1577d1d9051
To: <sip:55052171@177.38.216.32>;tag=9331612361070580132
Call-ID: 1de2ead3b94b5cc706b04a34af93a942@192.168.1.32
CSeq: 1382131788 INVITE
Via: SIP/2.0/UDP 192.168.1.32:5060;rport=5060;branch=z9hG4bK110e06e15d01fdd942d0696fca27242a
Contact: <sip:551155052171@177.38.220.16:5060;user=phone>
Supported: timer,100rel
Record-Route: <sip:177.38.220.16:5060;lr>
Record-Route: <sip:177.38.216.32;lr=on;ftag=25fce1577d1d9051>
Content-Type: application/sdp
Content-Length: 304

v=0
o=MG4000|2.0 7126 7445 IN IP4 177.38.220.16
s=-
c=IN IP4 177.38.220.16
t=0 0
m=audio 32282 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=ptime:20
a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0
a=X-vrzcap:identification bin=DSR2879 Prot=mgcp App=MG

But I cannot understand why the packet is sent with this information on the header...the ip address of the destination instead the source address.



Cesar Romani
Avaya ACSS | APSS (SME/Data)| ACIS (Data)
Comptia Convergence +

 

[redphone]

The From header does not have to contain the IP address of the source address. It should be the "domain" of the SIP message; one the UAS has authorized. The Contact header is the one that should contain the source address. If the SIP trunk provider requires the From header to contain the source address, then change the ITPS Domain Name to the local IP address.

 

[johnromani]

Man, you are the best!! lol

I never thought that we have to change the ITSP Domain... i tried many others configs, including the ITSP Proxy but not the ITSP Domain.

it works!!!

Thank you!!

Cesar Romani
Avaya ACSS | APSS (SME/Data)| ACIS (Data)
Comptia Convergence +

 

[johnromani]

Hello Bas1234,

We got another problem: The provider says that the Contact field is outbounding with our internal (LAN) IP address instead of our Public IP address.

Do you know how can I modify this? Because on SIP URI tab, all the configs (by credential, internal data,..) is going by 192.168.x.x;

Regards,

Cesar Romani
Avaya ACSS | APSS (SME/Data)| ACIS (Data)
Comptia Convergence +

 

[Bas1234]

in the ARS create a SC

Code: N;
Feat: Dial
TelN: N"@<your-IP-or-FQDN-you-like>"

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

 

[johnromani]

Hello guys, specially Bas and redphone,

the problem now is that the Contact header is goign out with IPOffice ip address, instead of the IP address inside FROM header, look:

96971mS SIP Call Tx: 17
INVITE sip:996674148@189.125.XXX.XX SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bK2cbffff10673c92c6dcb1d0f24561a33
From: "1139379412" <sip:1139379412@[highlight #FCE94F]200.158.XXX.XXX[/highlight]>;tag=81c73e963786a3f0
To: <sip:996674148@189.125.XXX.XX>
Call-ID: 1b9b7dcf347df84b7b21f32f0f0ba95f@192.168.1.100
CSeq: 1234080173 INVITE
Contact: "1139379412" <sip:1139379412@[highlight #FCE94F]192.168.1.100:5060[/highlight];transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
P-Asserted-Identity: "1139379412" <sip:1139379412@192.168.1.100:5060>
Content-Length: 228

v=0
o=UserA 1488869877 567641887 IN IP4 192.168.1.100
s=Session SDP
c=IN IP4 192.168.1.100
t=0 0
m=audio 49152 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

The provider said that they cannot send RTP data because they trying to send to this private address, I don't know why they do it, but they said that their system works like this.

Cesar Romani
Avaya ACSS | APSS (SME/Data)| ACIS (Data)
Comptia Convergence +

 

[amriddle01]

I think you need a better and less demanding provider, I have never needed to piss about with settings as much as this, if they were any good they would have a SBC that would do all this for you

 

[Westi]

these settings are changed with different firewall settings in your LAN 1 or LAN 2 settings depending which one you are using for your SIP trunk.
STUN will determine the settings to use like bas1234 suggested before.
If STUN doesn't do well try manually to change it which is a pain but there are only 4 or 5 settings there and it is not open Internet usually so one down.

Joe W.

TeleTechs.ca
FHandw, ACSS (SME), ACIS (SME)


“This is the end of the world, make sure to buy your T-shirt before it is too late"
Original expression of my daughter

 

[redphone]

If you have a static public IP address, populate the "Public IP Address" field on the Network Topology tab for the LAN you are using to route SIP traffic (System>LANx>Network Topology). Or, if this IP address changes from time to time, then a STUN server can be used to determine the public IP address. There are many public STUN servers listed on the Internet; pick one and click Run STUN. This will set the Public IP Address and Firewall/NAT Type for you.

Then on the SIP Line>Transport tab, populate which LAN to use for Network Topology Info.

Also, the provider is not looking for the IP address of the "Contact:" header to send RTP, it is looking at the "Connection Information" field for the IP address (c=IN IP4 192.168.1.100) and the Media description for the port (m=audio 49152 RTP/AVP 18 101). Setting the network topology correctly, will change both the Contact and the connection info to the external IP address. The RTP ports are set under System>Lanx>VoIP RTP. The RTP ports listed there should be mapped in your firewall to go to IP Office (192.168.1.100), along with port 5060.

 

[johnromani]

Hello redphone,

My site is configured like this:

[IP office]--LAN1--[SW]--[FW]--[Router]--provider link----[sip domain]

This is not working well... The voice is on one-way. Incoming voice is not come.

When I change the topology to this one, its works fine:

10.x.x.x[telephony]--LAN1(10.x.x.x)--[IP office]--LAN2(192.168.x.x)--[modem]--provider link----[sip domain]

I know that the problem could be on firewall or router changes, because the provider said that the Contact or "Connection Information" are changed to my private address (10.x.x.x) and because of it, they cannot send the RTP data.

I just need to know if Can I change this field anyway, because I change all the configuration (on Line and On LAN>Network Topology) and nothing changes.

I just want to send the responsability to the customer, or the network engineer.

Thanks!





Cesar Romani
Avaya ACSS | APSS (SME/Data)| ACIS (Data)
Comptia Convergence +

 

[jconrad0305]

setup your stun settings on the lan1 or 2 tab. it doesn't have to run but you will need to put your public ip and port in there and that should replace the internal with the public ip.

ACSS SME
ACSS a bunch of CM
CCNA Voice