[Asterisk-Users] MWI's for Third Party Softswitch

Jay Ray jonty_11 at yahoo.com
Mon Mar 28 16:02:24 MST 2005


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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