[asterisk-users] chan_sip bug? (Asterisk 1.4)

[Jian Gao]

Chad, You are right. tcpdump shows Asterisk sees 777 when the packet 
arrived.

It's truned out my router somehow modified the packet! I am using a Asus 
RT-N16 router with TomatoUSB firmware. There is a setting "SIP Helper". 
I disabled this "feature" on the router then everything back to normal.

There is one thing still puzzle me: It seems enable or disable this 
"feature" doesn't effect other SIP thunks. It could be the Sippy server 
use two different IP. The INVITE come from 208.65.xxx.xxx, but in its 
packet it try to use 74.205.216.77 as contact address. Is my guess 
correct? Why it does this?

*Jian *

On 11-01-27 04:31 PM, Chad Wallace wrote:
> On Thu, 27 Jan 2011 14:52:06 -0800
> Jian Gao<jian.gao at sjgeophysics.com>  wrote:
>
>> Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk
>> stop working after the upgrade. Here is the sip debug:
>> ---------------------------------------------------------------------------
>> <--- SIP read from 208.65.xxx.xxx:5060 --->
> That packet is coming from the other end (Sippy).  The problem is
> probably there.  However, it could be that the networking routines in
> Asterisk have added a 7 at the end.  You could compare a tcpdump of
> that packet to what Asterisk sees.  If the tcpdump shows .777 then the
> problem is in Sippy.  If it shows .77 then the problem is in Asterisk.
>
>
>> INVITE sip:1778xxxxxxx at 10.11.22.77:5060 SIP/2.0
>> Via: SIP/2.0/UDP
>> 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport
>> Via: SIP/2.0/UDP
>> 208.65.xxx.xxx:5061;branch=z9hG4bK-uhhmj2ir4ew6cn4p;rport=5061
>> Max-Forwards: 69
>> Record-Route:<sip:208.65.xxx.xxx;lr>
>> Contact: "Anonymous"<sip:208.65.xxx.xxx:5061>
>> To:<sip:1778xxxxxxx at 208.65.xxx.xxx:5060>
>> From:<sip:604xxxxxxx at 208.65.xxx.xxx:5060>;tag=ixpa27sbhn3inu5x.o
>> Call-ID: 550D37B3 at 208.72.xxx.xxx~o
>> CSeq: 819 INVITE
>> Expires: 300
>> Content-Disposition: session
>> Content-Type: application/sdp
>> User-Agent: Sippy
>> cisco-GUID: 2851810672-711266784-2763915291-559912524
>> h323-conf-id: 2851810672-711266784-2763915291-559912524
>> Content-Length: 109
>>
>> v=0
>> o=Sippy 223452192 0 IN IP4 74.205.216.77
>> s=-
>> t=0 0
>> m=audio 33830 RTP/AVP 0
>> c=IN IP4 74.205.216.777
>>
>> <------------->
>> --- (17 headers 6 lines) ---
>> Sending to 208.65.xxx.xxx : 5060 (NAT)
>> Using INVITE request as basis request - 550D37B3 at 208.72.xxx.xxx~o
>> Found peer 'FreePhoneLine'
>> Found RTP audio format 0
>> [2011-01-27 14:35:18] WARNING[2911]: chan_sip.c:5948 process_sdp_c:
>> Unable to lookup RTP Audio host in c= line, 'IN IP4 74.205.216.777'
>> [2011-01-27 14:35:18] WARNING[2911]: chan_sip.c:5741 process_sdp:
>> Insufficient information in SDP (c=)...
>> -----------------------------------------------------------------------------------------------------------
>>
>>
>>
>>
>>
>> It seems in the SIP INVITE, the IP 74.205.216.77 somehow changed to
>> 74.205.216.777.
>> I am not sure this is a bug of Asterisk or not.
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