[asterisk-users] Spiral SIP Request problem

[amit salunkhe]

Hello,

I am using OpenSIPS to register all the users and planning to use asterisk
for Auto Attendant, Queues, Voicemail and Conference Bridge.

I have a scenario where the signaling does not happen properly:

1)      A user from Opensips dials an extension 7000 which is an
auto-attendant extension. The call is routed to asterisk to play the auto
attendant messages like Welcome and Dial the extension of the party you wish
to reach etc !!

2)      Whenever a user presses any 4 digit number, asterisk routes that
call back to Openser to find if the user is available there.

3)      In Opensips the preference for the user is set such that all call
are supposed to go to voicemail which is handled by the same asterisk again;
so a 0 is prefixed to the R-URI and sent back to asterisk by opensips. 0 is
prefixed so that asterisk does not generate loop detected for the INVITE.

4)      The call gets answered by asterisk but the 200 OK message keeps
routing between asterisk and opensips and then asterisk times out with
“*Maximum
retries exceeded on transmission*” error.



Scenario:

1)      User ----7000 at sipproxy.com--------à Opensips
--------7000 at asterisk.com-----àAsterisk

2)      Asterisk ------200ok-----à Opensips---200ok---àUser

3)      Asterisk waits for extension input and user presses 7010

4)      Asterisk------INVITE
7010 at sipproxy.com--------->Opensips<https://remote.novanet.net/owa/redir.aspx?C=31661696fae74c3a94f26b78ea106eae&URL=mailto%3a7010%40sipproxy.com---------%253eOpensips>

5)      Opensips checks the preference for user 7010 and needs to send this
call to voicemail of 7010 which is handled again in asterisk.

6)      Opensips-----INVITE
07010 at asterisk.com------>Asterisk<https://remote.novanet.net/owa/redir.aspx?C=31661696fae74c3a94f26b78ea106eae&URL=mailto%3a07010%40asterisk.com------%253eAsterisk>(0
is added in the R-URI in order to avoid loop detected by asterisk)

7)      Asterisk plays minivm greet and gets ready for minivm record
although no audio is heard by the user

8)      Asterisk--------200ok-----àOpensips (Since the call to voicemail by
Opensips is answered)

9)      Opensips-------200ok-----àAsterisk (Since the call to 7010 by
asterisk is answered)

10)   Asterisk does not ACK the 200ok in step 9 and instead keeps sending
the 200ok in step 8 until maximum retransmission is reached and then the
call is hung up.





w/regards,

Amit
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