[OpenSIPS-Users] Transfer issue
Flavio E. Goncalves
flavio at asteriskguide.com
Fri Oct 23 22:33:38 CEST 2009
Hi Peter,
You need to have support for REFER in all the SIP components, UACs
and Gateways. Your SIP provider seems to be refusing your REFERS with
the message "501 Not Implemented". The only way to workaround (as far
as I know) is to use a gateway before your SIP provider that
implements the REFER messages. You can do this using Asterisk. Handle
the REFERs in the same way you do with INVITEs, there is a parameter
called allowexternaldomains and it needs to be set to yes. The
security for REFERs is the same as the one used for INVITEs.
Regards,
Flavio E. Goncalves
At 08:39 AM 10/23/2009, you wrote:
>I moved my opensips in the network, it's now directly connected to my sip
>trunk, i can call inside, i can call outside. I can transfer inside. But
>when i try to tranfser an outside nummer i get to see this ngrep:
>
>U 90.145.5.96:5060 -> 90.145.5.83:5060
>REFER sip:SIP_5F8 at 217.112.112.114 SIP/2.0.
>Via: SIP/2.0/UDP 90.145.5.96;branch=z9hG4bK80db89a3AE4DF976.
>From:
><sip:0031851110814 at 77.73.226.254:5060;user=phone>;tag=519E7E95-45526C60.
>To: <sip:0624469780 at 217.112.112.114;user=phone>;tag=202954455.
>Route: <sip:90.145.5.83;lr=on>,
><sip:77.73.226.254;lr=on;ftag=202954455;did=4b1.a8f7e0a5>.
>CSeq: 2 REFER.
>Call-ID: 1975939792 at 217.112.112.114.
>Contact: <sip:105 at 90.145.5.96>.
>User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0133.
>Refer-To: sip:101 at 90.145.5.83:5060.
>Referred-By: <sip:105 at 90.145.5.83>.
>Max-Forwards: 70.
>Content-Length: 0.
>.
>
>
>U 90.145.5.83:5060 -> 77.73.226.254:5060
>REFER sip:SIP_5F8 at 217.112.112.114 SIP/2.0.
>Via: SIP/2.0/UDP 90.145.5.83;branch=z9hG4bK0582.ce0b1427.0.
>Via: SIP/2.0/UDP 90.145.5.96;branch=z9hG4bK80db89a3AE4DF976.
>From:
><sip:0031851110814 at 77.73.226.254:5060;user=phone>;tag=519E7E95-45526C60.
>To: <sip:0624469780 at 217.112.112.114;user=phone>;tag=202954455.
>Route: <sip:77.73.226.254;lr=on;ftag=202954455;did=4b1.a8f7e0a5>.
>CSeq: 2 REFER.
>Call-ID: 1975939792 at 217.112.112.114.
>Contact: <sip:105 at 90.145.5.96>.
>User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0133.
>Refer-To: sip:101 at 90.145.5.83:5060.
>Referred-By: <sip:105 at 90.145.5.83>.
>Max-Forwards: 69.
>Content-Length: 0.
>.
>
>
>U 77.73.226.254:5060 -> 90.145.5.83:5060
>SIP/2.0 501 Not Implemented.
>Via: SIP/2.0/UDP 90.145.5.83;branch=z9hG4bK0582.ce0b1427.0.
>Via: SIP/2.0/UDP 90.145.5.96;branch=z9hG4bK80db89a3AE4DF976.
>From:
><sip:0031851110814 at 77.73.226.254:5060;user=phone>;tag=519E7E95-45526C60.
>To: <sip:0624469780 at 217.112.112.114;user=phone>;tag=202954455.
>Call-ID: 1975939792 at 217.112.112.114.
>CSeq: 2 REFER.
>Content-Length: 0.
>.
>
>
>U 90.145.5.83:5060 -> 90.145.5.96:5060
>SIP/2.0 501 Not Implemented.
>Via: SIP/2.0/UDP 90.145.5.96;branch=z9hG4bK80db89a3AE4DF976.
>From:
><sip:0031851110814 at 77.73.226.254:5060;user=phone>;tag=519E7E95-45526C60.
>To: <sip:0624469780 at 217.112.112.114;user=phone>;tag=202954455.
>Call-ID: 1975939792 at 217.112.112.114.
>CSeq: 2 REFER.
>Content-Length: 0.
>.
>
>It makes sense to me that i forgot something in my config, a refer module or
>something? any toughts/pushes in the right direction would be greatly
>appreciated!
>
>best regards.
>--
>View this message in context:
>http://n2.nabble.com/Transfer-issue-tp3877950p3877950.html
>Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
>_______________________________________________
>Users mailing list
>Users at lists.opensips.org
>http://lists.opensips.org/cgi-bin/mailman/listinfo/users
More information about the Users
mailing list