[asterisk-users] Spiral SIP Request problem

[Mark Michelson]

amit salunkhe wrote:
> 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
> 

What Asterisk version are you using? I'm guessing that since you are using 
minivm, you are likely using a 1.6 version of Asterisk. I recently (as in 
earlier this week) committed some code to the 1.6 branches to improve the spiral 
support in Asterisk. If you are able to retry with the latest subversion 
checkout of the branch you are using, you may find that things are working 
better now.

Mark Michelson