[asterisk-bugs] [JIRA] (ASTERISK-22932) SIP Channel fails to parse refer_to_domain

[Jan Gaida (JIRA)]

Jan Gaida created ASTERISK-22932:
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             Summary: SIP Channel fails to parse refer_to_domain
                 Key: ASTERISK-22932
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22932
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_sip/Transfers
    Affects Versions: 11.6.0
            Reporter: Jan Gaida


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.

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