[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

Flavio (and others),

I've created the following setup:
Sip trunk -> asterisk -> opensips

I can now call in and outside, i can transfer calls internaly on opensips, i
can blind transfer every call.. but i can't transfer the announced calls
once again. I made in the script the following changes: 

route{

        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                exit;
        }
        if (is_method("REFER")) {
        route(4);
        }

And route(4) is the drouting script, so then it should go back to the
gateway (asterisk) that knows it should do a dial to 103 right?

Everything else seem to work fine, calling outside/inside to asterisk
extentions, to opensips extentions i can even transfer a call from a outside
caller from a asterisk extention to an opensips extention, but when i try to
transfer from the opensips extention to another opensips extention i get
transfer failed.

Any input on this issue? what should i check? is the script above wrong :)?

Best regards, and thanks for all the awesome input you guys are giving me.


Flavio Goncalves wrote:
> 
> 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.
>>--
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>>http://n2.nabble.com/Transfer-issue-tp3877950p3877950.html
>>Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>
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> 
> 
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