[Freeswitch-users] Call forwarding problem

Tue Nov 24 13:05:48 PST 2009

I was having trouble doing call forwarding from my SIP phone that is connected to FreeSWITCH. It turns out that my SIP phone is actually sending 302 Moved Temporarily responses, but my SIP gateway does not support 302 Moved Temporarily or SIP REFER messages. How do I get FreeSWITCH to forward calls without sending 302 Moved Temporarily or SIP REFER messages?

Here is the SIP debug from our gateway:
Received:
INVITE sip:+19725357722 at ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246 SIP/2.0
v: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKe19865e46222056ca70435e66fde4127.19be3eb0
Record-Route: <sip:65.243.172.245:5060;lr>
v: SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK3f49bc4eb4ac163ffa354de0e6384d30.12e7ffbd;received=63.77.76.236
record-route: <sip:63.77.76.236;lr>
f: <sip:+19729555871 at 199.173.101.208:5060;user=phone>;tag=dc7-13c4-3ec95a-3ad03068-3ec95a
t: <sip:+19725357722 at 63.77.76.236:5060;user=phone>
i: a1f37fb0d065adc713c43ec95af54289baa8ec2034c293850-0569-7989
CSeq: 1 INVITE
Max-Forwards: 18
k: 100rel, replaces
allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3ec95a-f54289ba-139ab2d1;received=199.173.101.208
m: <sip:199.173.101.208:5060;transport=UDP>
c: application/SDP
l: 210
P-Asserted-Identity: <sip:9729555871 at 63.77.76.236;user=phone>
Privacy: none

v=0
o=- 540754816 540754816 IN IP4 199.173.111.141
s=-
c=IN IP4 199.173.111.141
t=0 0
m=audio 30056 RTP/AVP 18 0 8 101
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Nov 24 15:08:00.367 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKe19865e46222056ca70435e66fde4127.19be3eb0,SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK3f49bc4eb4ac163ffa354de0e6384d30.12e7ffbd;received=63.77.76.236,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3ec95a-f54289ba-139ab2d1;received=199.173.101.208
From: <sip:+19729555871 at 199.173.101.208:5060;user=phone>;tag=dc7-13c4-3ec95a-3ad03068-3ec95a
To: <sip:+19725357722 at 63.77.76.236:5060;user=phone>
Date: Tue, 24 Nov 2009 21:08:00 GMT
Call-ID: a1f37fb0d065adc713c43ec95af54289baa8ec2034c293850-0569-7989
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

Nov 24 15:08:00.367 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:19725357722 at 168.75.202.212:5062 SIP/2.0
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870
From: <sip:19729555871 at 168.75.202.246>;tag=14E93594-2488
To: <sip:19725357722 at 168.75.202.212>
Date: Tue, 24 Nov 2009 21:08:00 GMT
Call-ID: 4802BACC-D87411DE-AC70D9DF-3419A306 at 168.75.202.246
Supported: timer,resource-priority,replaces
Min-SE:  1800
Cisco-Guid: 1208058493-3631485406-2892683743-874095366
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1259096880
Contact: <sip:19729555871 at 168.75.202.246:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 17
P-Asserted-Identity: <sip:19729555871 at 168.75.202.246>
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 314

v=0
o=CiscoSystemsSIP-GW-UserAgent 2925 1780 IN IP4 168.75.202.246
s=SIP Call
c=IN IP4 199.173.111.141
t=0 0
m=audio 30056 RTP/AVP 18 0 8 101
c=IN IP4 199.173.111.141
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Nov 24 15:08:00.367 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870
From: <sip:19729555871 at 168.75.202.246>;tag=14E93594-2488
To: <sip:19725357722 at 168.75.202.212>
Call-ID: 4802BACC-D87411DE-AC70D9DF-3419A306 at 168.75.202.246
CSeq: 101 INVITE
Timestamp: 1259096880 0.000342
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15654M
Content-Length: 0

Nov 24 15:08:00.419 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870
From: <sip:19729555871 at 168.75.202.246>;tag=14E93594-2488
To: <sip:19725357722 at 168.75.202.212>;tag=49aF8vtgHme2c
Call-ID: 4802BACC-D87411DE-AC70D9DF-3419A306 at 168.75.202.246
CSeq: 101 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15654M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
P-Asserted-Identity: "19725357722" <sip:19725357722 at 168.75.202.212>

Nov 24 15:08:00.427 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:19725357722 at 168.75.202.212:5062 SIP/2.0
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870
From: <sip:19729555871 at 168.75.202.246>;tag=14E93594-2488
To: <sip:19725357722 at 168.75.202.212>;tag=49aF8vtgHme2c
Date: Tue, 24 Nov 2009 21:08:00 GMT
Call-ID: 4802BACC-D87411DE-AC70D9DF-3419A306 at 168.75.202.246
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

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