[Asterisk-Dev] Re: [Asterisk-Users] Asterisk not working with session border controller

[Venkat Venkataraju]

I finally got what the problem is about from the guys @ middle (i still 
dont have the documentation of the system). The asterisk seems to 
replace the contact address in the registration request.

What the middle sends is
Contact: <sip:2001*68.252.134.123=68.53.80.66+63437 at 68.22.23.84:7062>

and what Middle get back from Asterisk is
Contact: <sip:2001 at 68.252.134.123>

the SIP DEBUG trace is

REGISTER sip:68.252.134.123 SIP/2.0
Via: SIP/2.0/UDP 68.22.23.84:7062;branch=z9hG4bK-middle-4328
Via: SIP/2.0/UDP 192.168.0.15:5060
From: <sip:2001 at 68.252.134.123;user=phone>;tag=420802951
To: <sip:2001 at 68.252.134.123;user=phone>
Contact: <sip:2001*68.252.134.123=68.53.80.66+63437 at 68.22.23.84:7062>
Call-ID: 2145446582 at 192.168.0.15
CSeq: 3 REGISTER
Content-Length: 0
User-Agent: Cisco ATA 186  v3.0.0 atasip (031210A)
Proxy-Authorization: Digest
username="2001",realm="asterisk",nonce="588dbbaa",uri="sip:68.252.134.123",res
ponse="e7cb8c40341f343fbbdf0ca3cb6b8868"


11 headers, 0 lines
Using latest request as basis request
Sending to 68.22.23.84 : 7062 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 68.22.23.84:7062;branch=z9hG4bK-middle-4328
Via: SIP/2.0/UDP 192.168.0.15:5060
From: <sip:2001 at 68.252.134.123;user=phone>;tag=420802951
To: <sip:2001 at 68.252.134.123;user=phone>;tag=as73b51389
Call-ID: 2145446582 at 192.168.0.15
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2001 at 68.252.134.123>
Content-Length: 0


  to 68.22.23.84:7062
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.22.23.84:7062;branch=z9hG4bK-middle-4328
Via: SIP/2.0/UDP 192.168.0.15:5060
From: <sip:2001 at 68.252.134.123;user=phone>;tag=420802951
To: <sip:2001 at 68.252.134.123;user=phone>;tag=as73b51389
Call-ID: 2145446582 at 192.168.0.15
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 120
Contact: <sip:2001 at 68.252.134.123>;expires=120
Date: Wed, 07 Jan 2004 05:51:22 GMT
Content-Length: 0

So is there a way to fix it? or work around it?

Thanks
Venkat


Olle E. Johansson wrote:
> Venkat Venkataraju wrote:
> 
>>
>> Steve Totaro wrote:
>>
>>> And you say this is a commercial product that you purchased and they 
>>> asked
>>> you to fix the problem?
>>
>>
>> guys @ middle claim that the Asterisk is breaking the RFC and they are 
>> asking me to fix it on Asterisk as our company wants to use Asterisk. 
>> The "middle" does seems to work with many other SIP proxies.
>>
>>>
>>> Where did you purchase it from or are you developing it and need 
>>> help?  More
>>> docs might help.
>>
>>
>> I'm not developing it, but i'm working with a company thats trying to 
>> bring consumer VoIP by using Asteisk. I'm working on call processing 
>> system that uses astman. But i was pulled into this to get asterisk 
>> work with Middle. I dont have any other docs. i've asked for it. i'll 
>> post it  if i get one.
>>
>> I've got logs from Middle, but i'm not sure how it may help. for the 
>> request
>>
>> REGISTER sip:sip-xxxxx.homeip.net SIP/2.0
>> Via: SIP/2.0/UDP 68.#.#.84:7062;branch=z9hG4bK-middle-4178
>> Via: SIP/2.0/UDP 68.#.#.125:5060
>> From: <sip:2002 at sip-xxxxx.homeip.net;user=phone>;tag=1722079273
>> To: <sip:2002 at sip-xxxxx.homeip.net;user=phone>
>> Contact: 
>> <sip:2002*sip-xxxxx.homeip.net=68.252.134.125+5060 at 68.#.#.84:7062>
>> Call-ID: 2365955374 at 68.#.#.125
>> CSeq: 6 REGISTER
>> Content-Length: 0
>> User-Agent: Cisco ATA 186  v2.16 ata18x (030401a)
>>
>> Asteisk is supposed to respond back to the IP on port 7062 (i was 
>> told. i'vnt read the RFC), but it send the request back on the port 
>> 7060, the originating port of the request.
>>
> Venkat,
> Please add a SIP DEBUG trace of a registration, so I see how Asterisk
> responds. That's a weird Contact: header...
> 
> /O
> 
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