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 <[EMAIL PROTECTED]> 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* <[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>> 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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