[asterisk-dev] SIP Channel fails to parse refer_to_domain

[Olle E. Johansson]

On 29 Nov 2013, at 12:48, Jan Gaida <jan.gaida at grupoamper.com> wrote:

> Hello,
> 
> The scenario is an attended transfer to a remote domain.
> 
> The short description: 
> When Asterisk receives a REFER message with a refer-to header that contains "?replaces=" query, in chan_sip's function "get_refer_info" the domain is extracted with the query-part. That leads to an error (no such host) when trying to send the INVITE to this domain.
> 
> The solution is to remove the query part ('?') in the get_refer_info function (see attached diff file).
> 
> ---
> 
> The following is a description of the events like a SIP trace; with A being transferor, B transferee (Asterisk) and C the transfer-target.
> 
> Two calls are established: call1 between B and A, and call2 between A and C.
> When starting the transfer, the transferor A sends a REFER to Asterisk (B):
> A -> REFER (Refer-to: <sip:C at domain?replaces=call2>) -> B
> B -> 202 Accepted -> A
> B -> Notify sipfrag 200 ok -> A
> A -> 200 ok -> B
> 
> Now, there should be an outgoing INIVTE with Replaces header:
> B -> INVITE (Replaces: call2) > C
> But this INVITE never is send.
> 
> In Asterisk's traces you see the following:
>   chan_sip.c: Attended transfer: Will use Replace-Call-ID : call2 (No check of from/to tags)
>   chan_sip.c: SIP transfer to extension C at internal by A at domain
>   chan_sip.c: This SIP transfer is to a remote SIP extension (remote domain domain?Replaces=call2)
> 
> Here you can see the error ^^^^. The remote domain still contains the query part (?Replaces) of the uri.
> 
> The traces then continue:
> chan_sip.c: SIP attended transfer: Still not our call - generating INVITE with replaces
> 
> Then it enters the dialplan where it is redirected with a Dial command:
> logger.c:     -- Executing [s at callreferred:4] Dial("SIP/domain-09a40b60", "SIP/C at domain?Replaces=call2|60|") in new stack
> 
> And finally the error:
> WARNING[32395] chan_sip.c: No such host: domain?Replaces=call2
> DEBUG[32395] chan_sip.c: Cant create SIP call - target device not registered
> 
> ---
> 
> With the attached patch, Asterisk behaves as expected.  Should I open an issue on the tracker?
yes, thank you!

/O