I guess the issue was a little too in-depth for assistance. Anyhow, we have resolved this issue by using maddr instead of fs_path and now have calls working between a FreeSWITCH and Nortel system which we have started to document http://wiki.freeswitch.org/wiki/ Connecting_Freeswitch_And_Nortel. Ben On May 5, 2009, at 2:51 PM, Benjamin Willis wrote:
Greetings,
We are performing interop testing to have FreeSWITCH as a gateway with a Nortel CS1000 box and would appreciate any help resolving our issues. There is a box that is a registrar and a redirect server. We can successfully register against this box. The box is picky about the invites we send; it expects something like the following :
INVITE sip:2270;phone-context=cdp.udp@newdomain.com;user=phone SIP/2.0
After great help from the freeSWITCH IRC we came up with the following dialplan entry to achieve this style header :
<action application="bridge" data="sofia/internal/sip:$1;phone- context=cdp.udp@newdomain.com;user=phone;fs_path=sip: 192.168.15.8:5060;"/>
At this point we are sending the formatted INVITE message as the redirect server expects. Our next issue where we would appreciate help is after the INVITE is sent, the redirect server sends back a 302 Moved Temporarily as expected.
SIP/2.0 302 Moved Temporarily Via: SIP/2.0/TCP 192.168.15.11;branch=z9hG4bKy0NcZvFcNr1ZB;received=192.168.15.11 From: "User 210" <sip:2300@192.168.15.11>;tag=jjmgrmHN6Qjaa To: <sip:2270;phone- context=cdp.udp@newdomain.com;user=phone>;tag=49582 Call-ID: 6998551e-b04c-122c-2d80-39a48cb53b8d CSeq: 114437328 INVITE Contact: <sip:2270;phone-context=cdp.udp@newdomain.com: 5060;maddr=192.168.15.10;transport=tcp;user=phone;x-nt- redirect=redirect-server> Content-Length: 0
The subsequent INVITE message is, again, sent back to the redirect server, not to the machine indicated by the maddr entry in the Contact field. The new INVITE looks like this :
INVITE sip:2270;phone-context=cdp.udp@newdomain.com: 5060;maddr=192.168.15.10;transport=tcp;user=phone;x-nt- redirect=redirect-server SIP/2.0 Via: SIP/2.0/UDP 192.168.15.11;rport;branch=z9hG4bK93BXjtmKD9pcN Route: <sip:192.168.15.8:5060> Max-Forwards: 69 From: "User 210" <sip:2300@192.168.15.11>;tag=ppSKZmm0rpS3e To: <sip:2270;phone-context=cdp.udp@newdomain.com;user=phone> Call-ID: ce59ac9e-b058-122c-2780-39a48cb53b8d CSeq: 114439989 INVITE Contact: <sip:mod_sofia@192.168.15.11:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session- description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 360 Remote-Party-ID: "User 210" <sip: 2300@192.168.15.11>;screen=yes;privacy=off
v=0 o=FreeSWITCH 6002387420083056305 1437766807142321498 IN IP4 192.168.15.11 s=FreeSWITCH c=IN IP4 192.168.15.11 t=0 0 m=audio 28220 RTP/AVP 10 0 8 9 3 101 13 a=rtpmap:10 L16/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
The following is a wireshark capture of the repeated INVITE messages back to the same machine, it repeats 3 times after the initial invite :
Conv.| Time | 192.168.15.11 | 192.168.15.8 | 0 |17.284 | INVITE SDP ( 16-bit audio, stereo g711U g711A ...2 GSM teleph) |SIP From: sip:2300@192.168.15.11 To:sip:2270 | |(5060) ------------------> (5060) | 0 |17.291 | 302 Moved Temporarily |SIP Status | |(5060) <------------------ (5060) | 0 |17.291 | ACK | |SIP Request | |(5060) ------------------> (5060) | --------------------------------------------------------- 1 |17.291 | INVITE SDP ( 16-bit audio, stereo g711U g711A ...2 GSM teleph) |SIP From: sip:2300@192.168.15.11 To:sip:2270 | |(5060) ------------------> (5060) | 1 |17.298 | 302 Moved Temporarily |SIP Status | |(5060) <------------------ (5060) | 1 |17.298 | ACK | |SIP Request | |(5060) ------------------> (5060) | --------------------------------------------------------- 2 |17.298 | INVITE SDP ( 16-bit audio, stereo g711U g711A ...2 GSM teleph) |SIP From: sip:2300@192.168.15.11 To:sip:2270 | |(5060) ------------------> (5060) | 2 |17.304 | 302 Moved Temporarily |SIP Status | |(5060) <------------------ (5060) | 2 |17.305 | ACK | |SIP Request | |(5060) ------------------> (5060) | --------------------------------------------------------- 3 |17.305 | INVITE SDP ( 16-bit audio, stereo g711U g711A ...2 GSM teleph) |SIP From: sip:2300@192.168.15.11 To:sip:2270 | |(5060) ------------------> (5060) | 3 |17.311 | 302 Moved Temporarily |SIP Status | |(5060) <------------------ (5060) | 3 |17.311 | ACK | |SIP Request | |(5060) ------------------> (5060) |
At this point we are a little stuck. We feel that the use of fs_path may be having unwanted effects on our redirects after looking at sofia_glue.c:sofia_glue_do_invite() and the parsing of the fs_path variable. We haven't filed this as a bug as it may be intended functionality and would like to get some insight from the more experience members before going down such a path.
Thanks in advance.
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