[Serusers] serweb click to dial

Wed Sep 29 04:18:57 CEST 2004

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 at 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 at 68.90.50.100";

                                   $config->ctd_uri="sip:699 at 68.90.50.100";

                                   $config->ctd_from="sip:699 at 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 at mycompany.com from
sip:1000 at mycompany.com by clicking on the 1002 phonebook entry.

###
U 68.90.50.100:5060 -> 12.3.4.10:5060
INVITE sip:1001 at 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 at mycompany.com>.
From: <sip:699 at mycompany.com>;tag=415a15814acb0.
CSeq: 1 INVITE.
Call-ID: 415a15814acb0.fifouacctd.
Content-Length: 131.
Contact: <sip:caller at 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 at mycompany.com>;tag=415a15814acb0.
To: <sip:1001 at 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 at mycompany.com>;tag=415a15814acb0.
To: <sip:1001 at 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 at mycompany.com>;tag=415a15814acb0.
To: <sip:1001 at 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 at 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 at 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 at mycompany.com>;tag=415a15814acb0.
Call-ID: 415a15814acb0.fifouacctd.
To: <sip:1001 at mycompany.com>;tag=7a9a058a857c4aba.
CSeq: 1 ACK.
Content-Length: 0.
.

#
U 68.90.50.100:5060 -> 68.84.242.201:5060
REFER sip:1001 at 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 at mycompany.com>;tag=7a9a058a857c4aba.
From: <sip:699 at mycompany.com>;tag=415a15814acb0.
CSeq: 2 REFER.
Call-ID: 415a15814acb0.fifouacctd.
Content-Length: 0.
Contact: <sip:caller at 68.90.50.100:5060>.
Referred-By: <sip:699 at mycompany.com>.
Refer-To: sip:1002 at 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 at mycompany.com>;tag=415a15814acb0.
To: <sip:1001 at 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 at 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 at mycompany.com>;tag=7a9a058a857c4aba.
From: <sip:699 at mycompany.com>;tag=415a15814acb0.
CSeq: 3 BYE.
Call-ID: 415a15814acb0.fifouacctd.
Content-Length: 0.
Contact: <sip:caller at 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 at mycompany.com>;tag=415a15814acb0.
To: <sip:1001 at 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 at 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 at mycompany.com>;tag=7a9a058a857c4aba.
To: <sip:699 at mycompany.com>;tag=415a15814acb0.
Contact: <sip:1001 at 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 at mycompany.com>;tag=7a9a058a857c4aba.
To: <sip:699 at 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 at 68.90.50.100:5060
out_uri=sip:caller at 68.90.50.100:5060 via_cnt==2".
.

Any ideas why 1002 never rings?

Regards,
Paul

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