Hi all. I'm using ser 0.8.99-dev6 with serweb. I'm not sure which version of serweb this is but I checked it out of berlios today with this command: cvs -z3 -d:pserver:anonymous@cvs.serweb.berlios.de:/cvsroot/serweb co iptel Anyhow, my click-to-dial feature is not fully functional. When I click on an entry in the phonebook it rings my extension as it should, but when I go off hook the number I'm calling never rings. Here are the click-to-dial settings from <serweb>/config/config.php $config->ctd_target="sip:699@68.90.50.100"; $config->ctd_uri="sip:699@68.90.50.100"; $config->ctd_from="sip:699@mycompany.com"; $config->ctd_outbound_proxy=""; Account 699 does not actually exist in my ser/subscriber table in MySQL. I'm very unclear on what these parameters should be set to. Also here is the ngrep output from my click-to-dial attempt. As you can see about half way down there is a REFER message but it seems to point to my Asterisk voice mail server (vm01.mycompany.com). Shouldn't this point to the person in my phone book that I'm calling? In this call sequence I called sip:1002@mycompany.com from sip:1000@mycompany.com by clicking on the 1002 phonebook entry. ### U 68.90.50.100:5060 -> 12.3.4.10:5060 INVITE sip:1001@12.3.4.10;user=phone SIP/2.0. Max-Forwards: 10. Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.95425785.0. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.85425785.0. To: <sip:1001@mycompany.com>. From: <sip:699@mycompany.com>;tag=415a15814acb0. CSeq: 1 INVITE. Call-ID: 415a15814acb0.fifouacctd. Content-Length: 131. Contact: <sip:caller@68.90.50.100:5060>. Reject-Contact: *;automata="YES". Content-Type: application/sdp. . v=0. o=click-to-dial 0 0 IN IP4 0.0.0.0. s=session. c=IN IP4 0.0.0.0. b=CT:1000. t=0 0. m=audio 9 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 12.3.4.10:5060 -> 68.90.50.100:5060 SIP/2.0 100 trying. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.95425785.0. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.85425785.0. From: <sip:699@mycompany.com>;tag=415a15814acb0. To: <sip:1001@mycompany.com>. Call-ID: 415a15814acb0.fifouacctd. CSeq: 1 INVITE. User-Agent: Grandstream BT100 1.0.5.11. Warning: 399 12.3.4.10 "detected NAT type is full cone". Content-Length: 0. . # U 12.3.4.10:5060 -> 68.90.50.100:5060 SIP/2.0 180 ringing. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.95425785.0. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.85425785.0. Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>. From: <sip:699@mycompany.com>;tag=415a15814acb0. To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba. Call-ID: 415a15814acb0.fifouacctd. CSeq: 1 INVITE. User-Agent: Grandstream BT100 1.0.5.11. Warning: 399 12.3.4.10 "detected NAT type is full cone". Content-Length: 0. . ## U 12.3.4.10:5060 -> 68.90.50.100:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.95425785.0. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.85425785.0. Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>. From: <sip:699@mycompany.com>;tag=415a15814acb0. To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba. Call-ID: 415a15814acb0.fifouacctd. CSeq: 1 INVITE. User-Agent: Grandstream BT100 1.0.5.11. Warning: 399 12.3.4.10 "detected NAT type is full cone". Contact: <sip:1001@12.3.4.10;user=phone>. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 161. . v=0. o=1001 8000 8000 IN IP4 12.3.4.10. s=SIP Call. c=IN IP4 12.3.4.10. t=0 0. m=audio 5004 RTP/AVP 0. a=recvonly. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 68.90.50.100:5060 -> 12.3.4.10:5060 ACK sip:1001@12.3.4.10;user=phone SIP/2.0. Max-Forwards: 10. Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>. Via: SIP/2.0/UDP 68.90.50.100;branch=0. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.85425785.0. From: <sip:699@mycompany.com>;tag=415a15814acb0. Call-ID: 415a15814acb0.fifouacctd. To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba. CSeq: 1 ACK. Content-Length: 0. . # U 68.90.50.100:5060 -> 68.84.242.201:5060 REFER sip:1001@vm01.mycompany.com:5060;user=phone SIP/2.0. Max-Forwards: 10. Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>. Via: SIP/2.0/UDP 68.90.50.100;branch=0. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bKf4c5.16082cd7.0. To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba. From: <sip:699@mycompany.com>;tag=415a15814acb0. CSeq: 2 REFER. Call-ID: 415a15814acb0.fifouacctd. Content-Length: 0. Contact: <sip:caller@68.90.50.100:5060>. Referred-By: <sip:699@mycompany.com>. Refer-To: sip:1002@mycompany.com. . # U 68.84.242.201:5060 -> 68.90.50.100:5060 SIP/2.0 202 Accepted. Via: SIP/2.0/UDP 68.90.50.100;branch=0;received=68.90.50.100;rport=5060. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bKf4c5.16082cd7.0. Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>. From: <sip:699@mycompany.com>;tag=415a15814acb0. To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba. Call-ID: 415a15814acb0.fifouacctd. CSeq: 2 REFER. User-Agent: VoiceMail. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: . Content-Length: 0. . ## U 68.90.50.100:5060 -> 68.84.242.201:5060 BYE sip:1001@vm01.mycompany.com:5060;user=phone SIP/2.0. Max-Forwards: 10. Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>. Via: SIP/2.0/UDP 68.90.50.100;branch=0. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK05c5.3f0aaa06.0. To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba. From: <sip:699@mycompany.com>;tag=415a15814acb0. CSeq: 3 BYE. Call-ID: 415a15814acb0.fifouacctd. Content-Length: 0. Contact: <sip:caller@68.90.50.100:5060>. . # U 68.84.242.201:5060 -> 68.90.50.100:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 68.90.50.100;branch=0;received=68.90.50.100;rport=5060. Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK05c5.3f0aaa06.0. Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>. From: <sip:699@mycompany.com>;tag=415a15814acb0. To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba. Call-ID: 415a15814acb0.fifouacctd. CSeq: 3 BYE. User-Agent: VoiceMail. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: . Content-Length: 0. . ########## U 12.3.4.10:5060 -> 68.90.50.100:5060 BYE sip:caller@68.90.50.100:5060 SIP/2.0. Via: SIP/2.0/UDP 12.3.4.10;branch=z9hG4bKbb2f53e68cca3b12. Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>. From: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba. To: <sip:699@mycompany.com>;tag=415a15814acb0. Contact: <sip:1001@12.3.4.10;user=phone>. Call-ID: 415a15814acb0.fifouacctd. CSeq: 27932 BYE. User-Agent: Grandstream BT100 1.0.5.11. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . # U 68.90.50.100:5060 -> 12.3.4.10:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 12.3.4.10;branch=z9hG4bKbb2f53e68cca3b12. From: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba. To: <sip:699@mycompany.com>;tag=415a15814acb0. Call-ID: 415a15814acb0.fifouacctd. CSeq: 27932 BYE. Content-Length: 0. Warning: 392 68.90.50.100:5060 "Noisy feedback tells: pid=26213 req_src_ip=68.90.50.100 req_src_port=5060 in_uri=sip:caller@68.90.50.100:5060 out_uri=sip:caller@68.90.50.100:5060 via_cnt==2". . Any ideas why 1002 never rings? Regards, Paul __________________________________ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail