Quote:
Originally Posted by martin
Transcoding will happen.
|
Are you sure about that, or is this a special arrangement for the SIP Broker PSTN access numbers?
I've got my Atcom AT-530 SIP phone registered to us.voxalot.com. It does not support the gsm codec.
If I send a SIP call to *010 <voxalot number> @sipbroker.com using the X-Ten Lite softphone, stipulating the gsm codec only, the call fails with "480 Temporarily Unavailable":
SEND >> 64.34.162.221:5060
Code:
INVITE sip:*01066xxxx@sipbroker.com SIP/2.0
Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z9hG4bK9FD9C80863834DB7BC500FD5D6F03974
From: 8150xxxxxxxx <sip:8150xxxxxxxx@sipbroker.com>;tag=2795761264
To: <sip:*01066xxxx@sipbroker.com>
Contact: <sip:8150xxxxxxxx@221.xx.xx.yyy:5060>
Call-ID: 7101315C-4947-47C6-BF67-CE135679335F@221.xx.xx.yyy
CSeq: 55933 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 201
v=0
o=8150xxxxxxxx 1782021 1782116 IN IP4 221.xx.xx.yyy
s=X-Lite
c=IN IP4 221.xx.xx.yyy
t=0 0
m=audio 29360 RTP/AVP 3 101
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
RECEIVE << 64.34.162.221:5060
Code:
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z9hG4bK9FD9C80863834DB7BC500FD5D6F03974
From: 8150xxxxxxxx <sip:8150xxxxxxxx@sipbroker.com>;tag=2795761264
To: <sip:*01066xxxx@sipbroker.com>;tag=54333a515cb8c1bd95fa9f9db754f4d6-ffdd
Call-ID: 7101315C-4947-47C6-BF67-CE135679335F@221.xx.xx.yyy
CSeq: 55933 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 64.34.163.35:5060 "Noisy feedback tells: pid=30938 req_src_ip=64.34.162.221 req_src_port=5060 in_uri=sip:66xxxx@proxy01.us1.voxalot.com out_uri=sip:66xxxx@proxy01.us1.voxalot.com via_cnt==3"
The call is accepted if I select the ulaw codec:
Code:
SEND >> 64.34.162.221:5060
INVITE sip:*01066xxxx@sipbroker.com SIP/2.0
Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z9hG4bK3AB3665BC6344878A418C081E866D048
From: 8150xxxxxxxx <sip:8150xxxxxxxx@sipbroker.com>;tag=3531632047
To: <sip:*01066xxxx@sipbroker.com>
Contact: <sip:8150xxxxxxxx@221.xx.xx.yyy:5060>
Call-ID: 3A85AB83-874C-412F-B03C-CF0C26560A4D@221.xx.xx.yyy
CSeq: 29057 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 202
v=0
o=8150xxxxxxxx 1769675 1769766 IN IP4 221.xx.xx.yyy
s=X-Lite
c=IN IP4 221.xx.xx.yyy
t=0 0
m=audio 29360 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
RECEIVE << 64.34.162.221:5060
Code:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z9hG4bK3AB3665BC6344878A418C081E866D048
Record-Route: <sip:64.34.173.199;lr=on;ftag=3531632047>
Record-Route: <sip:64.34.163.35;lr=on;ftag=3531632047>
Record-Route: <sip:64.34.162.221;lr=on;ftag=3531632047>
From: 8150xxxxxxxx <sip:8150xxxxxxxx@sipbroker.com>;tag=3531632047
To: <sip:*01066xxxx@sipbroker.com>;tag=22037778
Call-ID: 3A85AB83-874C-412F-B03C-CF0C26560A4D@221.xx.xx.yyy
CSeq: 29057 INVITE
Contact: <sip:*01066xxxx@221.xx.xx.xx:6080>
supported: replaces
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 151
v=0
o=sdp_admin 21487122 24659200 IN IP4 221.xx.xx.xx
s=A conversation
c=IN IP4 221.xx.xx.xx
t=0 0
m=audio 10016 RTP/AVP 0
a=rtpmap:0 PCMU/8000