[Asterisk-Users] sip.conf and SIP client host= not recognized
in some cases
Andres
andres at telesip.net
Thu May 6 07:23:25 MST 2004
Karl Brose wrote:
>
> No, no. port=5060 is the port number that * binds to on the local
> interface. got nothing to do with the clients.
>
I am not talking about the destination port of the packet, I am talking
about the source port of the packet. We have already gone through this
problem at our company. We solved it by making sure the remote carrier
always used the same source port (for example 5090), the we put
port=5090 in their entry in sip.conf, and problem solved.
>
>
>
> 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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