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@190.100.200.1>;tag=g8uj4z79n7
To: <sip:101@190.100.200.1;user=phone;intercom=true>;tag=as30cdf7be
Call-ID: 3c28f81360e9-30j4vfzyh0vq@190-100-200-18
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101@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@190.100.200.19 SIP/2.0
Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK5a5bb490
From: "Astrid Buero" <sip:200@190.100.200.1>;tag=as73047910
To: <sip:101@190.100.200.19>
Contact: <sip:200@190.100.200.1>
Call-ID: 2f6dffce31bfcd8a4b70688367c7d181@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@190.100.200.1>;tag=g8uj4z79n7
To: <sip:101@190.100.200.1;user=phone;intercom=true>;tag=as30cdf7be
Call-ID: 3c28f81360e9-30j4vfzyh0vq@190-100-200-18
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101@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@190.100.200.1>;tag=as73047910
To: <sip:101@190.100.200.19>;tag=7jlddlf13r
Call-ID: 2f6dffce31bfcd8a4b70688367c7d181@190.100.200.1
CSeq: 102 INVITE
Contact: <sip:101@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@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@190.100.200.1>;tag=as73047910
To: <sip:101@190.100.200.19>;tag=7jlddlf13r
Contact: <sip:200@190.100.200.1>
Call-ID: 2f6dffce31bfcd8a4b70688367c7d181@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@190.100.200.1>;tag=g8uj4z79n7
To: <sip:101@190.100.200.1;user=phone;intercom=true>;tag=as30cdf7be
Call-ID: 3c28f81360e9-30j4vfzyh0vq@190-100-200-18
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101@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 - - - -
Michael Swan
2004-May-24 09:29 UTC
[Asterisk-Users] call waiting indicator do not work for me.
At 02:36 PM 5/22/2004 +0200, nicolas wrote:>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".)Hi Nicolas, We experienced the same problem recently with our snom200. It happened when we were trying to upgrade to the 2.05x firmware releases. I believe what happened during one of the many restarts and reloads, a phone option got reset. Try opening up the browser interface to the phone, then clicking Settings>Redirection, then using the Event menu to set to "Never". Ours somehow got set to "Always". Once we made this change, the busy messages went away. I don't know what the default value for this setting is. As for MWI, it has been our experience that it does not work properly in the 2.04x firmware (MWI light lights but never gets cleared when all messages have been deleted.) We tried updating to all the 2.05x releases which did fix the MWI behavior but broke the flash/hold feature. We went back to 2.04 because hold was more important than a working MWI. YMMV. Michael Swan Neon Software, Inc.