We're running 2.0.5-174 here, and have been happy with it. We come away impressed with all the features and power available to use through vicidial.
However, our upstream provider has become more strict with the caller id passed to them from the dialer, which normally isn't a problem but some of our agents like to dial directly from their softphone instead of using the manual dial interface in vici (for customer support calls or quick calls to other offices). So, calls that used to go through when dialed directly from a softphone are now rejected.
This is what the sip trace shows when an agent tries to dial directly from their softphone (IP addresses are partially occluded):
- Code: Select all
U 24.120.183.XXX:5060 -> 69.71.55.XXX:5060
INVITE sip:1702235XXXX@69.71.55.XXX SIP/2.0.
Via: SIP/2.0/UDP 24.120.183.XXX:5060;branch=z9hG4bK430163b1;rport.
From: "5004" <sip:cc5004@24.120.183.XXX>;tag=as22e07885.
To: <sip:17022353166@69.71.55.XXX>.
Contact: <sip:cc5004@24.120.183.XXX>.
Call-ID: 199f8a2f23f06d030b3711224429ff28@24.120.183.XXX.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "5004" <sip:cc5004@24.120.183.XXX>;privacy=off;screen=no.
Date: Mon, 14 Sep 2009 17:33:01 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Content-Type: application/sdp.
Content-Length: 218.
Any ideas on how to get asterisk to pass the correct (or heck, any) CID without breaking vici?