[Asterisk-Users] sip.conf and SIP client host= not recognized in some cases

[Karl Brose]

No, no.  port=5060 is the port number that * binds to on the local 
interface. got nothing to do with the clients.




Andres wrote:

> the problem is B has a different source port (it is not 5060).  Notice 
> that A does have source port=5060.   sip.conf is not flexible enough 
> to recognize users with dynamic source ports (unless they are behind 
> NAT and you have nat=yes)
>
>
> Glenn Dalgliesh wrote:
>
>> I am seeing an issue with getting certain sip devices to be 
>> recognized as
>> defined SIP clients host= in the sip.conf and the only deference that 
>> I can
>> find btw sources that work and don't work is that devices that send 
>> packets
>> with an Initial Via header of themselves appears to work and pick the
>> context correctly but those that don't have the Via just get dropped 
>> in the
>> context of the [General] context in sip.conf. Anyone have any similar
>> experiences?
>>
>> Call comes from ccc.ccc.ccc.ccc to Asterisk from Invite in Example A 
>> in ends
>> up in [inbound] context but in Example B it ends up in [default]. The 
>> only
>> difference I can find btw these two examples is the fact that A has a 
>> VIA
>> record and B doesn't. Can anyone confirm this behavior or at least 
>> explain
>> it? (Used today's CVS)
>>
>> /etc/asterisk/sip.conf
>> [general]
>> port = 5060                     ; Port to bind to
>> bindaddr = aaa.aaa.aaa.aaa               ; Address to bind to
>> context = default            ; Default for incoming calls
>>
>> [carriera]
>> type=friend
>> host=ccc.ccc.ccc.ccc
>> context=inbound
>>
>> [carrierb]
>> type=friend
>> host=bbb.bbb.bbb.bbb
>> context=inbound
>>
>> /etc/asterisk/extensions.conf
>> [inbound]
>> exten => _.,1,Playback,tt-monkeysintro
>>
>> [default]
>> exten => _.,1,Congestion
>>
>>
>> Example A:
>> U ccc.ccc.ccc.ccc:5060 -> aaa.aaa.aaa.aaa:5060
>>  INVITE sip:4445552574 at aaa.aaa.aaa.aaa SIP/2.0..
>> Via: SIP/2.0/UDP ccc.ccc.ccc.ccc:5060;branch=z9hG4bK7ab24dcc..
>> From: "asterisk" <sip:asterisk at ccc.ccc.ccc.ccc>;tag=as3a541e32..
>> To: <sip:4445552574 at aaa.aaa.aaa.aaa>..Contact:
>> <sip:asterisk at ccc.ccc.ccc.ccc>..
>> Call-ID: 75adb4aa7e9ff711120b14f518b44a1b at ccc.ccc.ccc.ccc..
>> CSeq: 102 INVITE..
>> User-Agent: Asterisk PBX..Date: Wed, 05 May 2004 21:08:44 GMT..Allow:
>> INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..
>> Content-Type: application/sdp..
>> Content-Length: 211..
>> ..
>> v=0..
>> o=root 13122 13122 IN IP4 ccc.ccc.ccc.ccc..
>> s=session..
>> c=IN IP4 ccc.ccc.ccc.ccc..
>> t=0 0..m=audio 18980 RTP/AVP 0 3 8..
>> a=rtpmap:0 PCMU/8000..
>> a=rtpmap:3 GSM/8000..
>> a=rtpmap:8 PCMA/8000..
>> a=silenceSupp:off - - - -..
>> #
>>
>> Example B:
>> U bbb.bbb.bbb.bbb:44151 -> aaa.aaa.aaa.aaa:5060
>>  INVITE sip:4445552574 at aaa.aaa.aaa.aaa:5060 SIP/2.0..
>> Call-ID: 7007601020188505154-1083791562 at bbb.bbb.bbb.bbb..
>> From: sip:8889992264 at bbb.bbb.bbb.bbb:5060;tag=12436..
>> To: sip:4445552574 at aaa.aaa.aaa.aaa:5060..
>> Content-Length: 251..
>> Content-Type: application/sdp..
>> CSeq: 1 INVITE..
>> Via: SIP/2.0/UDP bbb.bbb.bbb.bbb:5060;branch=z9hG4bK-61400000000
>>  03442-414d9af3..
>> Contact: sip:8889992264 at bbb.bbb.bbb.bbb:5060..
>> Supported: 100rel..
>> Max-Forwards: 70..
>> ..
>> v=0..
>> o=MG4000|1.0 111 12345 IN IP4 65.77.154.6..
>> s=-..
>> c=IN IP4 65.77.154.6..
>> t=0 0..
>> m=audio 7824 RTP/AVP 18 0 102 103..
>> a=rtpmap:102 G.723.1a-L/8000..
>> a=rtpmap:103 telephone-event/8000..
>> a=fmtp:103 0-15..
>> a=X-sqn: 0..a=X-cap: 1
>> image udptl t38..
>> a=ptime:10..
>>
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>>  
>>
>
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