[Asterisk-Dev] MWI for VoiceMail

Sat Apr 2 13:10:10 MST 2005

Hi All,

  Have not gotten any response/pointers on the best way to implement this - Here is what I found....looking at the code...

  Everytime SIP socket reads a packet > 2 bytes , it spawns a new thread, to deal with the REGISTER/INVITE whatever is coming into the UDP socket.....

   I am thinking of creating a TYPE= friend for teh Softswitch (host = x.x.x.x). and create many mailboxes under taht for each Endpoint registered on that Softswtich.

   When some one  calls an Endpoint on the 3rd Pty Sofswitch...and it gets forwarded to the VoiceMail (Asterisk - *)...* recvs an INVITE....with the Mailbox # in the To header.

   I can check the Diversion header in the INVITE  and then based on that 
I can store INVITE details and other peer details from sip.conf in a NEW Structure called sip_peer_vm_info ( in addition to peer info, I shud store IP and Port of the Softswitch sending the INVITE in that structure).   Then after the person deposits the msg...I check for NEW MSGS and send the NOTIFY to the Softswitch based on info in the new structure....Makign sure I put voicemailbox # in the To header of the NOTIFY as per RFC.

Now, when the Endpoint on the  Softswitch wants to check VoiceMail...he calls a VoiceMailMain NUmber, I check teh Dialed Number in INVITE, coming from Softswitch...and if it matches the VoiceMailMain NUmber, then get the number from the From Field and map that to a Mailbox account (if I want to allow voicemail retieval w/o entering VM Box #)

Again, I check for new msgs, and send a NOTIFY appropriately.....

Comments are highly appreciated.......

Thx,
Jay

Jay Ray <jonty_11 at yahoo.com> wrote:
Hi All,

   I want to use Asterisk for VoiceMail for a softswitch. 

I can dial in to leave voicemail and retrieve. Now there are many SIP Endpoints registered to the Softswitch. The Asterisk is sending a NOTIFY msg to the Softswitch on <ip addr>:0

Somehow Asterisk Looses the port from where the INVITE came in, this NOTIFY msg is not going out of the Asterisk, I cannot see in Ethereal. here is the Error I get constantly on the CLI prompt...

Mar 28 13:27:38 WARNING[2044]: chan_sip.c:682 __sip_xmit: sip_xmit of 0x9204bd4 (len 457) to 12.42.87.60 returned -1: Invalid argument
Scheduling destruction of call '5f2dfa2f1f539aa603700b4c760e628d at 12.42.87.193' in 15000 ms
Retransmitting #1 (no NAT):
NOTIFY sip:12.42.87.60:0 SIP/2.0
Via: SIP/2.0/UDP 12.42.87.193:5060;branch=z9hG4bK4301f53f
From: "asterisk" <sip:asterisk at 12.42.87.193>;tag=as644a3ae7
To: <sip:12.42.87.60:0>
Contact: <sip:asterisk at 12.42.87.193>
Call-ID: 5f2dfa2f1f539aa603700b4c760e628d at 12.42.87.193
CSeq: 102 NOTIFY
User-Agent: Asterisk 193
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 44
Messages-Waiting: yes
Voice-Message: 11/0
to 12.42.87.60:0
Mar 28 13:27:39 WARNING[2044]: chan_sip.c:682 __sip_xmit: sip_xmi

Can someone direct me to the portion of the Code where Asterisk Generates reponses for Voice Mail Call Flow. I would think goes through the Normal SIP stack.

Another thing I want to do it send the MailBox Number back in the "To" header of the NOTIFY that is being sent to the Softswitch. The Softswitch will then direct the Notify to the Appropriate Endpoint.

Any help or guidance appreciated.
Jay

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