[asterisk-users] Asterisk SIP with no RTP audio port (was Internode weirdness)

Sat Jan 22 18:48:40 CST 2011

On 01/23/11 10:18, Da Rock wrote:
> On 01/22/11 22:04, Da Rock wrote:
>> On 01/22/11 20:00, Da Rock wrote:
>>> On 01/21/11 20:28, Da Rock wrote:
>>>> On 01/21/11 03:19, Tom Rymes wrote:
>>>>> On 01/19/2011 10:34 PM, Da Rock wrote:
>>>>>
>>>>>> WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
>>>>>> non-existing call leg on other UA. SIP dialog
>>>>>> '481cf0543743e6bb7006991d409ed3bc at 150.101.178.33:5060'. Giving up.
>>>>>
>>>>> Have you tried disallowing re-invites?
>>>> Sorry for the delay, but I've tried both yes and no- one of the 
>>>> first things I tried, but I get your reasoning.
>>>>
>>>> Thanks
>>> Some more information has come to light- bit of luck this clue 
>>> happen to come to my attention: My provider could be using a 
>>> Broadworks system. Does that change things much?
>>>
>>> In my sip debug for the peer it flashed up realm="Broadworks" from 
>>> the peer.
>> Being very new to asterisk and SIP I'm still trying to learn the 
>> "protocol". Perhaps someone here may be able to correct my 
>> understanding if necessary (and a point in the right direction would 
>> help significantly).
>>
>> What I wasn't realising was that if I set sip debug on it output the 
>> entire sip message. So my output looks like this:
>>
>> -- Executing [0871271201 at users:1] Goto("SIP/<local ata>-00000017", 
>> "internode-outgoing,0871271201,1") in new stack
>>     -- Goto (internode-outgoing,0871271201,1)
>>     -- Executing [0871271201 at internode-outgoing:1] Dial("SIP/<local 
>> ata>-00000017", "SIP/0871271201 at sip-out") in new stack
>> Audio is at 5060
>> Video is at <asterisk ip>:5060
>> Text is at <asterisk ip>:5060
>> Adding codec 0x4 (ulaw) to SDP
>> Adding codec 0x2 (gsm) to SDP
>> Adding codec 0x8 (alaw) to SDP
>> Adding codec 0x10 (g726aal2) to SDP
>> Adding codec 0x20 (adpcm) to SDP
>> Adding codec 0x40 (slin) to SDP
>> Adding codec 0x80 (lpc10) to SDP
>> Adding codec 0x200 (speex) to SDP
>> Adding codec 0x400 (ilbc) to SDP
>> Adding codec 0x800 (g726) to SDP
>> Adding codec 0x1000 (g722) to SDP
>> Adding codec 0x8000 (slin16) to SDP
>> Adding video codec 0x100000 (h263p) to SDP
>> Adding text codec 0x4000000 (red) to SDP
>> Adding text codec 0x8000000 (t140) to SDP
>> Adding non-codec 0x1 (telephone-event) to SDP
>> Reliably Transmitting (no NAT) to 203.2.134.1:5060:
>> INVITE sip:0871271201 at sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK41104eea
>> Max-Forwards: 70
>> From: "Skinner's Home" <sip:<local ata>@<asterisk ip>>;tag=as6683ffea
>> To: <sip:0871271201 at sip.internode.on.net>
>> Contact: <sip:<local ata>@<asterisk ip>:5060>
>> Call-ID: 5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX 1.8.1.1
>> Date: Sat, 22 Jan 2011 11:23:35 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 736
>>
>> v=0
>> o=root 189870721 189870721 IN IP4 <asterisk ip>
>> s=Asterisk PBX 1.8.1.1
>> c=IN IP4 <asterisk ip>
>> b=CT:384
>> t=0 0
>> m=audio 17220 RTP/AVP 0 3 8 112 5 10 7 110 97 111 9 118 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:112 AAL2-G726-32/8000
>> a=rtpmap:5 DVI4/8000
>> a=rtpmap:10 L16/8000
>> a=rtpmap:7 LPC/8000
>> a=rtpmap:110 speex/8000
>> a=    -- Called 0871271201 at sip-out
>>
>> <--- SIP read from UDP:203.2.134.1:5060 --->
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP <asterisk ip>:5060;received=<my ext 
>> ip>;branch=z9hG4bK41104eea;rport=61533
>> From: "<local ata>" <sip:<local ata>@<asterisk ip>>;tag=as6683ffea
>> To: <sip:0871271201 at sip.internode.on.net>
>> Call-ID: 5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060
>> CSeq: 102 INVITE
>>
>> <------------->
>> --- (6 headers 0 lines) ---
>> Reliably Transmitting (no NAT) to 203.2.134.1:5060:
>> OPTIONS sip:sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK7ab229d7
>> Max-Forwards: 70
>> From: "Unknown" <sip:Unknown@<asterisk ip>>;tag=as72e93b63
>> To: <sip:sip.internode.on.net>
>> Contact: <sip:Unknown@<asterisk ip>:5060>
>> Call-ID: 068700ad12b1d48a5059a1837a04a8b3@<asterisk ip>:5060
>> CSeq: 102 OPTIONS
>> User-Agent: Asterisk PBX 1.8.1.1
>> Date: Sat, 22 Jan 2011 11:24:00 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Length: 0
>>
>>
>> ---
>>
>> <--- SIP read from UDP:203.2.134.1:5060 --->
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP <asterisk ip>:5060;received=<my ext 
>> ip>;branch=z9hG4bK7ab229d7;rport=61533
>> From: "Unknown" <sip:Unknown@<asterisk ip>>;tag=as72e93b63
>> To: <sip:sip.internode.on.net>;tag=488684762-1295695493625
>> Call-ID: 068700ad12b1d48a5059a1837a04a8b3@<asterisk ip>:5060
>> CSeq: 102 OPTIONS
>> Allow-Events: 
>> call-info,line-seize,dialog,message-summary,as-feature-event
>> Content-Length: 0
>>
>> <------------->
>> --- (8 headers 0 lines) ---
>> Really destroying SIP dialog 
>> '068700ad12b1d48a5059a1837a04a8b3@<asterisk ip>:5060' Method: OPTIONS
>> Reliably Transmitting (no NAT) to 203.2.134.1:5060:
>> OPTIONS sip:sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK53bdf76d
>> Max-Forwards: 70
>> From: "Unknown" <sip:Unknown@<asterisk ip>>;tag=as2a328631
>> To: <sip:sip.internode.on.net>
>> Contact: <sip:Unknown@<asterisk ip>:5060>
>> Call-ID: 42d9681e4b8dcad66879b54c697d7e32@<asterisk ip>:5060
>> CSeq: 102 OPTIONS
>> User-Agent: Asterisk PBX 1.8.1.1
>> Date: Sat, 22 Jan 2011 11:24:00 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Length: 0
>>
>>
>> ---
>>
>> <--- SIP read from UDP:203.2.134.1:5060 --->
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP <asterisk ip>:5060;received=<my ext 
>> ip>;branch=z9hG4bK53bdf76d;rport=61533
>> From: "Unknown" <sip:Unknown@<asterisk ip>>;tag=as2a328631
>> To: <sip:sip.internode.on.net>;tag=1907972657-1295695493739
>> Call-ID: 42d9681e4b8dcad66879b54c697d7e32@<asterisk ip>:5060
>> CSeq: 102 OPTIONS
>> Allow-Events: 
>> call-info,line-seize,dialog,message-summary,as-feature-event
>> Content-Length: 0
>>
>> <------------->
>> --- (8 headers 0 lines) ---
>> Really destroying SIP dialog 
>> '42d9681e4b8dcad66879b54c697d7e32@<asterisk ip>:5060' Method: OPTIONS
>>
>> <--- SIP read from UDP:203.2.134.1:5060 --->
>> SIP/2.0 408 Request Timeout
>> Via: SIP/2.0/UDP <asterisk ip>:5060;received=<my ext 
>> ip>;branch=z9hG4bK41104eea;rport=61533
>> From: "<local ata>" <sip:<local ata>@<asterisk ip>>;tag=as6683ffea
>> To: <sip:0871271201 at sip.internode.on.net>;tag=aprqngfrt-3s5u6r20000c6
>> Call-ID: 5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060
>> CSeq: 102 INVITE
>>
>> <------------->
>> --- (6 headers 0 lines) ---
>> [Jan 22 21:24:08] WARNING[993]: chan_sip.c:19069 
>> handle_response_invite: Re-invite to non-existing call leg on other 
>> UA. SIP dialog '5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060'. 
>> Giving up.
>> Transmitting (no NAT) to 203.2.134.1:5060:
>> ACK sip:0871271201 at sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK41104eea
>> Max-Forwards: 70
>> From: "<local ata>" <sip:<local ata>@<asterisk ip>>;tag=as6683ffea
>> To: <sip:0871271201 at sip.internode.on.net>;tag=aprqngfrt-3s5u6r20000c6
>> Contact: <sip:<local ata>@<asterisk ip>:5060>
>> Call-ID: 5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060
>> CSeq: 102 ACK
>> User-Agent: Asterisk PBX 1.8.1.1
>> Content-Length: 0
>>
>>
>> ---
>> Scheduling destruction of SIP dialog 
>> '5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060' in 6400 ms 
>> (Method: INVITE)
>>     -- SIP/sip-out-00000018 is circuit-busy
>> Scheduling destruction of SIP dialog 
>> '5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060' in 6400 ms 
>> (Method: INVITE)
>>   == Everyone is busy/congested at this time (1:0/1/0)
>>     -- Executing [0871271201 at internode-outgoing:2] Answer("SIP/<local 
>> ata>-00000017", "2") in new stack
>>     -- Executing [0871271201 at internode-outgoing:3] 
>> Playback("SIP/<local ata>-00000017", "ss-noservice") in new stack
>>     -- <SIP/<local ata>-00000017> Playing 'ss-noservice.gsm' 
>> (language 'en')
>>   == Spawn extension (internode-outgoing, 0871271201, 3) exited 
>> non-zero on 'SIP/<local ata>-00000017'
>> Really destroying SIP dialog 
>> '5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060' Method: INVITE
>> [Jan 22 21:24:36] NOTICE[993]: chan_sip.c:12142 sip_reregister:    -- 
>> Re-registration for  0731292848 at sip.internode.on.net
>> > doing dnsmgr_lookup for 'sip.internode.on.net'
>> > ast_get_srv: SRV lookup for '_sip._udp.sip.internode.on.net' mapped 
>> to host sip.internode.on.net, port 5060
>> REGISTER 11 headers, 0 lines
>> Reliably Transmitting (no NAT) to 203.2.134.1:5060:
>> REGISTER sip:sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK07b4e860
>> Max-Forwards: 70
>> From: <sip:0731292848 at sip.internode.on.net>;tag=as1f1ee1a8
>> To: <sip:0731292848 at sip.internode.on.net>
>> Call-ID: 597b79434d3c224a3b484fd718502b34@<asterisk ip>
>> CSeq: 1440 REGISTER
>> User-Agent: Asterisk PBX 1.8.1.1
>> Authorization: Digest username="0731292848", realm="BroadWorks", 
>> algorithm=MD5, uri="sip:sip.internode.on.net", 
>> nonce="BroadWorksXgj5g2j3vTbun57kBW", 
>> response="4a529cc44fa4c19924fed72fed1da2e2", qop=auth, 
>> cnonce="44b80039", nc=0000053a
>> Expires: 120
>> Contact: <sip:0731292848@<asterisk ip>:5060>
>> Content-Length: 0
>>
>>
>> ---
>>
>> <--- SIP read from UDP:203.2.134.1:5060 --->
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP <asterisk ip>:5060;received=<my ext 
>> ip>;branch=z9hG4bK07b4e860;rport=61533
>> From: <sip:0731292848 at sip.internode.on.net>;tag=as1f1ee1a8
>> To: <sip:0731292848 at sip.internode.on.net>;tag=aprqcauh8h3-7m1v3j200810a
>> Call-ID: 597b79434d3c224a3b484fd718502b34@<asterisk ip>
>> CSeq: 1440 REGISTER
>> Contact: <sip:0731292848@<asterisk ip>:5060>;expires=150
>>
>> <------------->
>> --- (7 headers 0 lines) ---
>> Scheduling destruction of SIP dialog 
>> '597b79434d3c224a3b484fd718502b34@<asterisk ip>' in 6400 ms (Method: 
>> REGISTER)
>> [Jan 22 21:24:36] NOTICE[993]: chan_sip.c:19502 
>> handle_response_register: Outbound Registration: Expiry for 
>> sip.internode.on.net is 150 sec (Scheduling reregistration in 135 s)
>>
>> From this it looks to me that the call is gong through, but asterisk 
>> is not acting on it- is that right? Or am I misreading it?
>>
>> I see the outgoing call and an sip message generated and sent the 
>> provider, the provider calls back trying, and then sends 2x 200 
>> messages saying ok. The provider then gets no response and sends back 
>> a 408. And then asterisk acts on it and tries to attach to a non 
>> existent leg.
>>
>> Internode insist on simply opening up the firewall on a dmz to allow 
>> complete access to the asterisk server with theirs, but I have 2 
>> points to contend with that:
>>
>> 1. What about spoofing? They may have HA cluster attached to that IP, 
>> but what about an attack on my server? Couldn't someone spoof the IP 
>> for their own purposes? (Or I could be paranoid too :) )
>>
>> 2. I aim to setup a hosting solution, so I can allow clients to peer 
>> with my asterisk (not using my trunk though) so I need to be security 
>> conscious, and allow more than just the nodephone server to connect.
>>
>> Any help?
>>
>> Cheers
> Ok. With no confirmation I'm actually on the right track, I've now run 
> full tcpdumps from both the asterisk server and pf. Here is my findings:
>
> I make an incoming call and I find an RTP peer audio port in the debug 
> output which I can also see in the tcpdump on both systems.
>
> I make an outgoing call and I cannot find any RTP audio port. I also 
> cannot see any ports opening in the tcpdumps from either system. WTF?!
>
> Any clues? Please?
Actually found an obscure reference: audio at 5060 (local asterisk SIP 
dialog). That doesn't actually make sense- or does it?