[Asterisk-Users] call waiting indicator do not work for me.

[nicolas]

Hi,

The call waiting indicator do not work for me.

I am using a snom200 cwi is switched on in phone-config.

Have asked snom, but there are can not help me, because it is working for
them.

When it is coming in an call while the phone is busy.
The phone returns:

-- Got SIP response 486 "Busy Here" back from 190.100.200.19

But it should not, should make a "call waiting indication".

(The same behaviour is when i am dialing the phone (in idle) from extern
without making an "exten => s,x,Answer".)

greeting
nicolas

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.100.200.18:5060;branch=z9hG4bK-kcw3axoq8oei
From: "Astrid Buero" <sip:200 at 190.100.200.1>;tag=g8uj4z79n7
To: <sip:101 at 190.100.200.1;user=phone;intercom=true>;tag=as30cdf7be
Call-ID: 3c28f81360e9-30j4vfzyh0vq at 190-100-200-18
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101 at 190.100.200.1>
Content-Length: 0


 to 190.100.200.18:5060
    -- Executing Dial("SIP/200-409e", "SIP/101|60|Ttr") in new stack
We're at 190.100.200.1 port 16492
Answering with preferred capability 1024
Answering with preferred capability 8
Answering with preferred capability 256
Answering with preferred capability 2
Answering with preferred capability 1
Answering with preferred capability 4
Answering with preferred capability 128
Answering with non-codec capability 1
12 headers, 16 lines
Reliably Transmitting:
INVITE sip:101 at 190.100.200.19 SIP/2.0
Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK5a5bb490
From: "Astrid Buero" <sip:200 at 190.100.200.1>;tag=as73047910
To: <sip:101 at 190.100.200.19>
Contact: <sip:200 at 190.100.200.1>
Call-ID: 2f6dffce31bfcd8a4b70688367c7d181 at 190.100.200.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 22 May 2004 10:08:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 364

v=0
o=root 32409 32409 IN IP4 190.100.200.1
s=session
c=IN IP4 190.100.200.1
t=0 0
m=audio 16492 RTP/AVP 97 8 18 3 4 0 7 101
a=rtpmap:97 iLBC/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:7 LPC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 190.100.200.19:5060
    -- Called 101
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 190.100.200.18:5060;branch=z9hG4bK-kcw3axoq8oei
From: "Astrid Buero" <sip:200 at 190.100.200.1>;tag=g8uj4z79n7
To: <sip:101 at 190.100.200.1;user=phone;intercom=true>;tag=as30cdf7be
Call-ID: 3c28f81360e9-30j4vfzyh0vq at 190-100-200-18
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101 at 190.100.200.1>
Content-Length: 0


 to 190.100.200.18:5060
alberspilnx8*CLI>

Sip read:
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK5a5bb490
From: "Astrid Buero" <sip:200 at 190.100.200.1>;tag=as73047910
To: <sip:101 at 190.100.200.19>;tag=7jlddlf13r
Call-ID: 2f6dffce31bfcd8a4b70688367c7d181 at 190.100.200.1
CSeq: 102 INVITE
Contact: <sip:101 at 190.100.200.19:5060;line=lhynyb3y>
Content-Length: 0


8 headers, 0 lines
    -- Got SIP response 486 "Busy Here" back from 190.100.200.19
Transmitting:CLI>
ACK sip:101 at 190.100.200.19:5060 SIP/2.0
Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK5a5bb490
From: "Astrid Buero" <sip:200 at 190.100.200.1>;tag=as73047910
To: <sip:101 at 190.100.200.19>;tag=7jlddlf13r
Contact: <sip:200 at 190.100.200.1>
Call-ID: 2f6dffce31bfcd8a4b70688367c7d181 at 190.100.200.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 190.100.200.19:5060
    -- SIP/101-8b54 is busy
  == Everyone is busy at this time
    -- Executing Wait("SIP/200-409e", "2") in new stack
    -- Executing VoiceMail("SIP/200-409e", "u200") in new stack
We're at 190.100.200.1 port 18090
Answering with preferred capability 1024
Answering with preferred capability 8
Answering with preferred capability 256
Answering with preferred capability 2
Answering with preferred capability 1
Answering with preferred capability 4
Answering with preferred capability 128
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.100.200.18:5060;branch=z9hG4bK-kcw3axoq8oei
From: "Astrid Buero" <sip:200 at 190.100.200.1>;tag=g8uj4z79n7
To: <sip:101 at 190.100.200.1;user=phone;intercom=true>;tag=as30cdf7be
Call-ID: 3c28f81360e9-30j4vfzyh0vq at 190-100-200-18
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101 at 190.100.200.1>
Content-Type: application/sdp
Content-Length: 364

v=0
o=root 32409 32409 IN IP4 190.100.200.1
s=session
c=IN IP4 190.100.200.1
t=0 0
m=audio 18090 RTP/AVP 97 8 18 3 4 0 7 101
a=rtpmap:97 iLBC/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:7 LPC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -