[SR-Users] Routing problem
Klaus Darilion
klaus.mailinglists at pernau.at
Thu May 20 00:46:56 CEST 2010
Lots of possible reasons:
- different From:
- different To
- To and RURI differ ...
It would be interesting to verify if direct calling still works if you
use a numerical username (as what happens when dialing via Openser), or
the other way round, when going via the SIP proxy but in From still
signal the name instead of the phone number.
Regarding RURI!=TO: Instead of rewritinghostport just rewrite the
destination URI, leaving the RURI untouched:
$du = "sip:pabx-alcatel.example.com:5060";
t_relay();
exit;
If still no luck, you can try setting RURI as you already did, but
changing To header too. There is no nice way to do this, you have to do
it manually, e.g:
rewritehostport........
remove_hf("To");
append_hf("To: $tU at pabx-alcatel.example.com:5060\r\n");
...
As 404 indicates that the target was not found, but RURIs are identical,
I suspect that the PBX false uses the To header and wants its domain
inside the To URI.
Regards
Klaus
Am 19.05.2010 22:58, schrieb alet at librelogiciel.com:
> Hi,
>
> On Tue, May 18, 2010 at 09:46:09AM +0200, Klaus Darilion wrote:
>>
>> As your config and the log messages seems to be correct, you should
>> debug on the network and verify which SIP messages are sent to which
>> location. You can use tcpdump/wireshark or the simple (my preferred
>> choice) ngrep tool.
>>
>
> I've followed your instructions and I've used ngrep on the openser box.
>
> sip.example.com<=> 10.125.0.1
> pabx-alcatel.example.com<=> 10.125.0.2
> my linux box with ekiga<=> 10.10.10.100
> my real analog phone's number, connected to the pabx : 290081
> my ekiga soft phone's number : 9888
>
> Below is a trace of what happens :
>
> sip:~# openserctl ul show
> database engine 'PGSQL' loaded
> Control engine 'FIFO' loaded
> entering fifo_cmd ul_dump
> Domain:: location table=512 records=2 max_slot=1
> AOR:: 9888
> Contact:: sip:9888 at 10.10.10.100 Q=1
> Expires:: 56
> Callid:: 6667a393-ef61-df11-897f-002197011099 at lafrime
> Cseq:: 25
> User-agent:: Ekiga/3.2.6
> State:: CS_SYNC
> Flags:: 0
> Cflag:: 0
> Socket:: udp:10.125.0.1:5060
> Methods:: 22335
> AOR:: alize
> Contact:: sip:alize at 10.125.0.2;transport=UDP Q=
> Expires:: 1051
> Callid:: 4b192ee9d1173056f4fbc3c317fb94ff at 10.125.0.2
> Cseq:: 1626385337
> User-agent:: OxO_GW_710/115.006
> State:: CS_SYNC
> Flags:: 0
> Cflag:: 0
> Socket:: udp:10.125.0.1:5060
> Methods:: 0
>
> Now I place a call from ekiga on my GNU/Linux box to my analog phone, by
> dialing 290081 :
>
> sip:~# ngrep -d any -W byline -t -q -P "" port 5060
>
> U 2010/05/20 07:11:08.471776 10.10.10.100:5060 -> 10.125.0.1:5060
> INVITE sip:290081 at sip.example.com SIP/2.0
> Date: Wed, 19 May 2010 20:11:11 GMT
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK66148856-f061-df11-897f-002197011099;rport
> User-Agent: Ekiga/3.2.6
> From: "Jerome Alet"<sip:9888 at sip.example.com>;tag=8c3a8756-f061-df11-897f-002197011099
> Call-ID: 5a3f8756-f061-df11-897f-002197011099 at lafrime
> To:<sip:290081 at sip.example.com>
> Contact:<sip:9888 at 10.10.10.100>
> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
> Content-Type: application/sdp
> Content-Length: 549
> Max-Forwards: 70
>
> v=0
> o=- 1274299871 1 IN IP4 10.10.10.100
> s=Opal SIP Session
> c=IN IP4 10.10.10.100
> t=0 0
> m=audio 5062 RTP/AVP 116 0 8 9 101 120
> a=sendrecv
> a=rtpmap:116 Speex/16000/1
> a=fmtp:116 sr=16000,mode=any
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:9 G722/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16,32,36
> a=rtpmap:120 NSE/8000
> a=fmtp:120 192-193
> m=video 5064 RTP/AVP 99 31
> b=AS:4096
> b=TIAS:4096000
> a=sendrecv
> a=rtpmap:99 theora/90000
> a=fmtp:99 height=576;width=704
> a=rtpmap:31 h261/90000
> a=fmtp:31 CIF=1;QCIF=1
>
>
> Here it seeems openser tells me it is trying :
>
> U 2010/05/20 07:11:08.474968 10.125.0.1:5060 -> 10.10.10.100:5060
> SIP/2.0 100 Giving a try
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK66148856-f061-df11-897f-002197011099;rport=5060
> From: "Jerome Alet"<sip:9888 at sip.example.com>;tag=8c3a8756-f061-df11-897f-002197011099
> Call-ID: 5a3f8756-f061-df11-897f-002197011099 at lafrime
> To:<sip:290081 at sip.example.com>
> Server: OpenSER (1.3.2-notls (i386/linux))
> Content-Length: 0
>
>
> Now the host and port rewriting has taken place, so openser transmits
> the invite to the pabx :
>
> U 2010/05/20 07:11:08.479225 10.125.0.1:5060 -> 10.125.0.2:5060
> INVITE sip:290081 at pabx-alcatel.example.com:5060 SIP/2.0
> Record-Route:<sip:10.125.0.1;lr=on>
> Date: Wed, 19 May 2010 20:11:11 GMT
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 10.125.0.1;branch=z9hG4bK3be6.1a17aeb2.0
> Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK66148856-f061-df11-897f-002197011099;rport=5060
> User-Agent: Ekiga/3.2.6
> From: "Jerome Alet"<sip:9888 at sip.example.com>;tag=8c3a8756-f061-df11-897f-002197011099
> Call-ID: 5a3f8756-f061-df11-897f-002197011099 at lafrime
> To:<sip:290081 at sip.example.com>
> Contact:<sip:9888 at 10.10.10.100>
> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
> Content-Type: application/sdp
> Content-Length: 549
> Max-Forwards: 69
>
> v=0
> o=- 1274299871 1 IN IP4 10.10.10.100
> s=Opal SIP Session
> c=IN IP4 10.10.10.100
> t=0 0
> m=audio 5062 RTP/AVP 116 0 8 9 101 120
> a=sendrecv
> a=rtpmap:116 Speex/16000/1
> a=fmtp:116 sr=16000,mode=any
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:9 G722/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16,32,36
> a=rtpmap:120 NSE/8000
> a=fmtp:120 192-193
> m=video 5064 RTP/AVP 99 31
> b=AS:4096
> b=TIAS:4096000
> a=sendrecv
> a=rtpmap:99 theora/90000
> a=fmtp:99 height=576;width=704
> a=rtpmap:31 h261/90000
> a=fmtp:31 CIF=1;QCIF=1
>
>
> The pabx answers it is trying :
>
> U 2010/05/20 07:11:08.509995 10.125.0.2:5060 -> 10.125.0.1:5060
> SIP/2.0 100 Trying
> To:<sip:290081 at sip.example.com>
> From: "Jerome Alet"<sip:9888 at sip.example.com>;tag=8c3a8756-f061-df11-897f-002197011099
> Call-ID: 5a3f8756-f061-df11-897f-002197011099 at lafrime
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 10.125.0.1;branch=z9hG4bK3be6.1a17aeb2.0
> Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK66148856-f061-df11-897f-002197011099;rport=5060
> Content-Length: 0
>
>
>
> But then it says it can't find the recipient (or at least that's what I
> understand) :
>
> U 2010/05/20 07:11:08.529453 10.125.0.2:5060 -> 10.125.0.1:5060
> SIP/2.0 404 Not Found
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE
> User-Agent: OxO_GW_710/115.006
> To:<sip:290081 at sip.example.com>;tag=3c9a6f069a4e18993408270150d6e3e6
> From: "Jerome Alet"<sip:9888 at sip.example.com>;tag=8c3a8756-f061-df11-897f-002197011099
> Call-ID: 5a3f8756-f061-df11-897f-002197011099 at lafrime
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 10.125.0.1;branch=z9hG4bK3be6.1a17aeb2.0
> Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK66148856-f061-df11-897f-002197011099;rport=5060
> Content-Length: 0
>
>
>
> Then openser aknowledges :
>
> U 2010/05/20 07:11:08.530454 10.125.0.1:5060 -> 10.125.0.2:5060
> ACK sip:290081 at pabx-alcatel.example.com:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.125.0.1;branch=z9hG4bK3be6.1a17aeb2.0
> From: "Jerome Alet"<sip:9888 at sip.example.com>;tag=8c3a8756-f061-df11-897f-002197011099
> Call-ID: 5a3f8756-f061-df11-897f-002197011099 at lafrime
> To:<sip:290081 at sip.example.com>;tag=3c9a6f069a4e18993408270150d6e3e6
> CSeq: 1 ACK
> Max-Forwards: 70
> User-Agent: OpenSER (1.3.2-notls (i386/linux))
> Content-Length: 0
>
>
> And then openser sends the Not Found answer back to my Ekiga :
>
> U 2010/05/20 07:11:08.530934 10.125.0.1:5060 -> 10.10.10.100:5060
> SIP/2.0 404 Not Found
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE
> User-Agent: OxO_GW_710/115.006
> To:<sip:290081 at sip.example.com>;tag=3c9a6f069a4e18993408270150d6e3e6
> From: "Jerome Alet"<sip:9888 at sip.example.com>;tag=8c3a8756-f061-df11-897f-002197011099
> Call-ID: 5a3f8756-f061-df11-897f-002197011099 at lafrime
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK66148856-f061-df11-897f-002197011099;rport=5060
> Content-Length: 0
>
>
> And my Ekiga aknowledges :
>
> U 2010/05/20 07:11:08.533180 10.10.10.100:5060 -> 10.125.0.1:5060
> ACK sip:290081 at sip.example.com SIP/2.0
> CSeq: 1 ACK
> Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK66148856-f061-df11-897f-002197011099;rport
> From: "Jerome Alet"<sip:9888 at sip.example.com>;tag=8c3a8756-f061-df11-897f-002197011099
> Call-ID: 5a3f8756-f061-df11-897f-002197011099 at lafrime
> To:<sip:290081 at sip.example.com>;tag=3c9a6f069a4e18993408270150d6e3e6
> Content-Length: 0
> Max-Forwards: 70
>
>
>
>
> What looks bizarre to me is that if from Ekiga I dial
> 290081 at pabx-alcatel.example.com, so not passing through openser not the
> openser box, it works perfectly, as can be seen below (using ngrep on my
> GNU/Linux box) :
>
>
>
> root at lafrime:~# ngrep -d any -W byline -t -q -P "" port 5060
> interface: any
> filter: (ip or ip6) and ( port 5060 )
>
> U 2010/05/20 07:34:49.070434 10.10.10.100:5060 -> 10.125.0.2:5060
> INVITE sip:290081 at pabx-alcatel.example.com SIP/2.0
> Date: Wed, 19 May 2010 20:34:49 GMT
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK044163a3-f361-df11-897f-002197011099;rport
> User-Agent: Ekiga/3.2.6
> From: "Jerome Alet"<sip:jerome at 10.10.10.100>;tag=be0f62a3-f361-df11-897f-002197011099
> Call-ID: 421362a3-f361-df11-897f-002197011099 at lafrime
> To:<sip:290081 at pabx-alcatel.example.com>
> Contact:<sip:jerome at 10.10.10.100>
> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
> Content-Type: application/sdp
> Content-Length: 549
> Max-Forwards: 70
>
> v=0
> o=- 1274301289 1 IN IP4 10.10.10.100
> s=Opal SIP Session
> c=IN IP4 10.10.10.100
> t=0 0
> m=audio 5072 RTP/AVP 116 0 8 9 101 120
> a=sendrecv
> a=rtpmap:116 Speex/16000/1
> a=fmtp:116 sr=16000,mode=any
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:9 G722/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16,32,36
> a=rtpmap:120 NSE/8000
> a=fmtp:120 192-193
> m=video 5074 RTP/AVP 99 31
> b=AS:4096
> b=TIAS:4096000
> a=sendrecv
> a=rtpmap:99 theora/90000
> a=fmtp:99 height=576;width=704
> a=rtpmap:31 h261/90000
> a=fmtp:31 CIF=1;QCIF=1
>
>
> So here the INVITE was sent to the pabx, and the pabx says it is trying
> :
>
> U 2010/05/20 07:34:49.099009 10.125.0.2:5060 -> 10.10.10.100:5060
> SIP/2.0 100 Trying
> To:<sip:290081 at pabx-alcatel.example.com>
> From: "Jerome Alet"<sip:jerome at 10.10.10.100>;tag=be0f62a3-f361-df11-897f-002197011099
> Call-ID: 421362a3-f361-df11-897f-002197011099 at lafrime
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 10.10.10.100:5060;received=10.10.10.100;branch=z9hG4bK044163a3-f361-df11-897f-002197011099;rport=5060
> Content-Length: 0
>
>
>
> And now, it seems it has found the 290081 at pabx-alcatel.example.com :
>
> U 2010/05/20 07:34:49.201710 10.125.0.2:5060 -> 10.10.10.100:5060
> SIP/2.0 180 Ringing
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE
> Contact: "ALET J 290081"<sip:290081 at 10.125.0.2;user=phone>
> Supported: from-change
> User-Agent: OxO_GW_710/115.006
> P-Asserted-Identity: "ALET J 290081"<sip:290081 at example.com;user=phone>
> To:<sip:290081 at pabx-alcatel.example.com>;tag=51a1558a4e6ac3bf5c1819f278ae2ecc
> From: "Jerome Alet"<sip:jerome at 10.10.10.100>;tag=be0f62a3-f361-df11-897f-002197011099
> Call-ID: 421362a3-f361-df11-897f-002197011099 at lafrime
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 10.10.10.100:5060;received=10.10.10.100;branch=z9hG4bK044163a3-f361-df11-897f-002197011099;rport=5060
> Content-Length: 0
>
>
> And from now on, the remaining of the "direct" call processes correctly.
>
> The major difference I see is the "To:" header : in both cases it is
> equal to what was dialed, but in the former case (through openser) the
> INVITE and To differ. Another difference is that in the latter case it
> uses my name instead of the number configured in my ekiga account for
> the openser registrar (which seems to be normal)
>
> Any idea what happens and what I should do to solve this problem ?
>
> Thanks in advance
>
> Jerome Alet
>
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