Hi Lads.
I have a problem with the SIP on IPO 500.
I have set up a transfer to mobile on night mode instead of getting a voice mail.
The call is comming through but when I answer a call on my mobile there is no voice.
UDP xx.xx.1.228:5060 -> xxx.xxx.104.45:5060
INVITE sip:landline@xx.xx.81.188 SIP/2.0
Via: SIP/2.0/UDP xx.xx.1.228:5060;branch=z9hG4bK2106b4d7;rport
From: "number" <sip:016190200@85.91.1.228>;tag=as2789b518
To: <sip:016521170@xx.xx.81.188>
Contact: <sip:number@xx.xx.1.228>
Call-ID: 74e7251b1e65601d5b41a94a18f64c3a@xx.xx.1.228
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 05 Sep 2008 11:22:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 17892 17892 IN IP4 xx.xx.1.228
s=session
c=IN IP4 xx.xx.1.228
t=0 0
m=audio 12874 RTP/AVP 8 0 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp
ff - - - -
a=ptime:20
a=sendrecv
this is a SIP getting to the system and then system is calling my mobile:
81559mS SIP Trunk: 19:Tx
INVITE sip:mobile@xx.xx.1.228 SIP/2.0
Via: SIP/2.0/UDP xx.xx.81.188:5060;rport;branch=z9hG4bK1e9c29b7312324efbe4538833d15c551
From: "landline" <sip:016521170@xx.xx.1.228>;tag=7bfa0b2abeedcd0b
To: <sip:mobile@xx.xx.1.228>
Call-ID: b45369937405a2feecdd8ed44b1b3155@xx.xx.81.188
CSeq: 1937767882 INVITE
Contact: "landline" <sip:016521170@xx.xx.81.188:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Type: application/sdp
Content-Length: 250
v=0
o=UserA 334752519 2946980902 IN IP4 xx.xx.81.188
s=Session SDP
c=IN IP4 xx.xx.81.188
t=0 0
m=audio 49152 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=fmtp:18 annexb = no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Any ideas??
I have a problem with the SIP on IPO 500.
I have set up a transfer to mobile on night mode instead of getting a voice mail.
The call is comming through but when I answer a call on my mobile there is no voice.
UDP xx.xx.1.228:5060 -> xxx.xxx.104.45:5060
INVITE sip:landline@xx.xx.81.188 SIP/2.0
Via: SIP/2.0/UDP xx.xx.1.228:5060;branch=z9hG4bK2106b4d7;rport
From: "number" <sip:016190200@85.91.1.228>;tag=as2789b518
To: <sip:016521170@xx.xx.81.188>
Contact: <sip:number@xx.xx.1.228>
Call-ID: 74e7251b1e65601d5b41a94a18f64c3a@xx.xx.1.228
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 05 Sep 2008 11:22:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 17892 17892 IN IP4 xx.xx.1.228
s=session
c=IN IP4 xx.xx.1.228
t=0 0
m=audio 12874 RTP/AVP 8 0 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp
a=ptime:20
a=sendrecv
this is a SIP getting to the system and then system is calling my mobile:
81559mS SIP Trunk: 19:Tx
INVITE sip:mobile@xx.xx.1.228 SIP/2.0
Via: SIP/2.0/UDP xx.xx.81.188:5060;rport;branch=z9hG4bK1e9c29b7312324efbe4538833d15c551
From: "landline" <sip:016521170@xx.xx.1.228>;tag=7bfa0b2abeedcd0b
To: <sip:mobile@xx.xx.1.228>
Call-ID: b45369937405a2feecdd8ed44b1b3155@xx.xx.81.188
CSeq: 1937767882 INVITE
Contact: "landline" <sip:016521170@xx.xx.81.188:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Type: application/sdp
Content-Length: 250
v=0
o=UserA 334752519 2946980902 IN IP4 xx.xx.81.188
s=Session SDP
c=IN IP4 xx.xx.81.188
t=0 0
m=audio 49152 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=fmtp:18 annexb = no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Any ideas??