Sip Profile on * with Gigaset C450 IP

Asterisk Asterisk General
1

I’m using a Gigaset C450 IP base station to try and log into my asterisk box, and I’m not quite sure what’s wrong.

The Sip conf is:

[10] type=friend username=10 secret=1234 context=internal canreinvite=no host=192.168.42.254 dtmfmode=rfc2833 qualify=200 mailbox=10 nat=0

The Gigaset has the following settings:

Auth Name: 10 Auth Pwd: 1234 Conf Auth Pwd: 1234 Username: 10 Domain: 192.168.42.200 Display Name: 10 Proxy Svr Add: blank Proxy Svr Port: 5060 Registrar Svr: 192.168.42.200 Listen ports: SIP port: 5060 RTP port: 5004 No Stun settings Codes unchanged: G711 a law, G711 u law, G726, G729

The debug info is as follows:

[quote]SIP Debugging enabled
Reliably Transmitting (no NAT) to 192.168.42.254:5060:
OPTIONS sip:192.168.42.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.42.200:5060;branch=z9hG4bK0baaeb5f;rport
From: “asterisk” sip:asterisk@192.168.42.200;tag=as08419c4b
To: sip:192.168.42.254
Contact: sip:asterisk@192.168.42.200
Call-ID: 38a7e5be7a87f47b2def59797194458e@192.168.42.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 12 Jan 2007 14:59:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

Retransmitting #1 (no NAT) to 192.168.42.254:5060:
OPTIONS sip:192.168.42.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.42.200:5060;branch=z9hG4bK0baaeb5f;rport
From: “asterisk” sip:asterisk@192.168.42.200;tag=as08419c4b
To: sip:192.168.42.254
Contact: sip:asterisk@192.168.42.200
Call-ID: 38a7e5be7a87f47b2def59797194458e@192.168.42.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 12 Jan 2007 14:59:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

linuxserver*CLI>
<— SIP read from 192.168.42.254:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.42.200:5060;branch=z9hG4bK0baaeb5f;rport=5060
From: “asterisk” sip:asterisk@192.168.42.200;tag=as08419c4b
To: sip:192.168.42.254;tag=1157064751
Call-ID: 38a7e5be7a87f47b2def59797194458e@192.168.42.200
CSeq: 102 OPTIONS
Contact: 10 sip:10@192.168.42.254:5060
Supported:
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Content-Type: application/sdp
Accept: application/sdp,application/dtmf-relay
Accept-Encoding: identity
Accept-Language: en
Content-Length: 263

v=0
o=10 5004 1 IN IP4 192.168.42.254
s=-
c=IN IP4 192.168.42.254
t=0 0
m=audio 5004 RTP/AVP 8 0 96 97 2 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000

<------------->
— (14 headers 12 lines) —
Really destroying SIP dialog ‘38a7e5be7a87f47b2def59797194458e@192.168.42.200’ Method: OPTIONS
linuxserver*CLI>
<— SIP read from 192.168.42.254:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.42.200:5060;branch=z9hG4bK0baaeb5f;rport=5060
From: “asterisk” sip:asterisk@192.168.42.200;tag=as08419c4b
To: sip:192.168.42.254;tag=1157064751
Call-ID: 38a7e5be7a87f47b2def59797194458e@192.168.42.200
CSeq: 102 OPTIONS
Contact: 10 sip:10@192.168.42.254:5060
Supported:
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Content-Type: application/sdp
Accept: application/sdp,application/dtmf-relay
Accept-Encoding: identity
Accept-Language: en
Content-Length: 263

v=0
o=10 5004 1 IN IP4 192.168.42.254
s=-
c=IN IP4 192.168.42.254
t=0 0
m=audio 5004 RTP/AVP 8 0 96 97 2 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000

<------------->
— (14 headers 12 lines) —
Reliably Transmitting (no NAT) to 192.168.42.254:5060:
OPTIONS sip:192.168.42.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.42.200:5060;branch=z9hG4bK75f3d266;rport
From: “asterisk” sip:asterisk@192.168.42.200;tag=as577c7fe5
To: sip:192.168.42.254
Contact: sip:asterisk@192.168.42.200
Call-ID: 25b418304bcfbe831b46c3481363f2f9@192.168.42.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 12 Jan 2007 15:00:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

linuxserver*CLI>
<— SIP read from 192.168.42.254:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.42.200:5060;branch=z9hG4bK75f3d266;rport=5060
From: “asterisk” sip:asterisk@192.168.42.200;tag=as577c7fe5
To: sip:192.168.42.254;tag=2865181148
Call-ID: 25b418304bcfbe831b46c3481363f2f9@192.168.42.200
CSeq: 102 OPTIONS
Contact: 10 sip:10@192.168.42.254:5060
Supported:
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Content-Type: application/sdp
Accept: application/sdp,application/dtmf-relay
Accept-Encoding: identity
Accept-Language: en
Content-Length: 263

v=0
o=10 5004 1 IN IP4 192.168.42.254
s=-
c=IN IP4 192.168.42.254
t=0 0
m=audio 5004 RTP/AVP 8 0 96 97 2 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000

<------------->
— (14 headers 12 lines) —
Really destroying SIP dialog ‘25b418304bcfbe831b46c3481363f2f9@192.168.42.200’ Method: OPTIONS
linuxserver*CLI> sip nodebug
SIP Debugging Disabled
[/quote]

2

I think it might be down to my use of domain - that is using the IP address. Is that correct, or am I missing something silly.

Cheers
Nunners

3

Yes. Try taking the IP address of the Asterisk server out of the domain: part but add it into the proxy: part.

4

on gigaset set domain=asterisk (realm in sip.conf)
ream is default asterisk, you can configure it in sip.conf
and Proxy server address = ip of asterisk

I have heard that even if gigaset has fixed ip you shall sett host=dynamic