[asterisk-users] Call Transfers in SER + Asterisk architecture

[Marco Mouta]

do you have created Asterisk views to SER database? Are you using sip
realtime on asterisk?
please post your extensions.conf.

By the way, I'm Portuguese:)

Qualquer coisa manda mail pode ser q consiga ajudar.

On 11/24/06, Ricardo Carvalho <rjcarvalho at reit.up.pt> wrote:
>
> Hi Marco,
>
> Ser has IP of Asterisk server in his "trusted" table, Asterisk isn't
> registered in Ser. When Ser needs to use Asterisk, it simply rewrites
> the IP destination with Asterisk's IP, and routes them to him.
>
> For example, here's one failed attempt in transferring a call PSTN ->
> VoIP -> VoIP:
>
>
> PSTN               Asterisk                   Ser
> phone_A               phone_B
> |        INVITE        |                       |
> |                       |
> | ------------------>  |                       |                       |
>                       |
> |      100 Trying      |                       |
> |                       |
> | <------------------- |                       |
> |                       |
> |                      |         INVITE        |
> |                       |
> |                      |  ------------------>  |        INVITE
> |                       |
> |                      |                       | ------------------->
> |                       |
> |                      |                       |        100 trying
> |                       |
>                        |       100 trying      | <-------------------
> |                       |
> |      100 trying      | <-------------------  |      180 Ringing
> |                       |
> | <------------------  |      180 Ringing      | <-------------------
> |                       |
> |     180 Ringing      | <------------------   |
> |                       |
> | <------------------  |                       |
> |                       |
> |          ACK         |                       |
> |                       |
> | -------------------> |           ACK         |
> |                       |
> |                      | ------------------->  |          ACK
> |                       |
> |                      |                       | ------------------->
> |                       |
> |                      |          RTP          |
> |                       |
> | <==================================================================>
> |                       |
> |                      |                       |
> |                       |
> |                      |                       |         REFER
> |                       |
> |                      |          REFER        | <-------------------
> |                       |
> |                      |  <------------------  |
> |                       |
> |                      |     404 Not Found     |
> |                       |
> |                      |  -------------------> |     404 Not Found
> |                       |
> |                      |                       |  ------------------>
> |                       |
> |                      |                       |
> |                       |
>
> In this example, phone_A answers the PSTN originated call, and wants to
> transfer the call to phone_B. A REFER message is than routed backwards
> to Asterisk, and he replies with those 404 Not Found messages. Phone_B
> never gets called!
>
> Should Asterisk be registered in Ser so this can work properly? How can
> that be done?
>
> Thanks,
> Ricardo.
>
>
>
>
>
>
>
>
> Marco Mouta wrote:
> > Hi Ricardo,
> >
> > Could you post a specific example where your problem happens.
> >
> > That way would be easier for me to try to help you on this.
> >
> > Does asterisk is registred into SER , or you have trust based
> > relationship between them?
> >
> >
> >
> > On 11/23/06, *Ricardo Carvalho* <rjcarvalho at reit.up.pt
> > <mailto:rjcarvalho at reit.up.pt>> wrote:
> >
> >     Hi,
> >
> >     I'm deploying a SER + Asterisk architecture, where SER is used as
> SIP
> >     registrar, and Asterisk is used for voicemail and PSTN gateway.
> >
> >     This system is already able to make Call Transfers (Blind and
> >     Attended)
> >     internally between phones registered in SER, although,  I can't make
> >     Call Transfers in some scenarios involving PSTN numbers (which need
> to
> >     pass through Asterisk).
> >
> >     The problem is that when the REFER message (that carries the
> Refer-To
> >     number to whom the call should be transferred) gets to Asterisk, it
> >     replies with a 404 Not Found message, and the Call Transfer isn't
> >     established!
> >
> >     Any ideas on how can I solve this problem?
> >
> >     Thanks in advance,
> >     Ricardo.
> >
> >
> >
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> >
> >
> >
> > --
> > Best regards,
> >
> > Marco Mouta
> >
> > ------------------------------------------------------------------------
> >
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-- 
Best regards,

Marco Mouta
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