[OpenSIPS-Users] Transfer issue

[Jeff Kronlage]

I'd like to be certain I understand from these last few posts regarding
this topic-

The suggestion is to use Asterisk 'behind' Opensips, transferring calls
to it only when a B2BUA is necessary?

I certainly understand not wanting to post a config, but can anyone
share a general idea of how this is accomplished?  I'm having a hard
time picturing how to send Asterisk an out-of-context REFER, while
Opensips 'held' the call up to that point.  Perhaps I'm over-thinking
it.

Thanks,

Jeff

-----Original Message-----
From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Peter den Hartog
Sent: Friday, October 23, 2009 3:15 PM
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] Transfer issue


Hello flavio, thank you for your response.

It makes sence to me, so i have to use asterisk as a gateway infront of
opensips, and let it handle outside domains to do stuff like
invites/refers.. 

The parameter your talking about is a Asterisk parameter right? I will
look
in to this.
Thank you for the information, it's a tricky journey into opensips, but
i'm
getting there :D

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=519E7E95-45526C6
0.
>>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=519E7E95-45526C6
0.
>>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=519E7E95-45526C6
0.
>>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=519E7E95-45526C6
0.
>>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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