31 May
2007
31 May
'07
4:18 p.m.
Hi to all.
I've configured my polycom ip500 IPphone to use it with ser.
If I try a call to users registred to ser all works fine.
If I try to call PSTN Number through my gateway I've the follow sip message:
INVITE sip:0672028405@10.28.19.202:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD.
From: "0660522015" <sip:0660522015@10.28.19.202>;tag=8DDF0DB-E8BCEB84.
To: <sip:0672028405@10.28.19.202;user=phone>.
CSeq: 2 INVITE.
Call-ID: 264aa927-d2a3a7c9-64f3fe3a(a)10.28.19.143.
Contact: <sip:0660522015@10.28.19.143>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1.
Supported: 100rel,replace.
Allow-Events: talk,hold,conference.
Proxy-Authorization: Digest username="0660522015",
realm="10.28.19.202",
nonce="465ef44bbb09c0155dc8555314519a20cac896f8",
uri="sip:0672028405@10.28.19.202:5060;user=phone",
response="e6eed258e095ee6be5be6c92210f9d99", algorithm=MD5.
Max-Forwards: 70.
Content-Type: application/sdp.
Content-Length: 237.
.
v=0.
o=- 1180620549 1180620549 IN IP4 10.28.19.143.
s=Polycom IP Phone.
c=IN IP4 10.28.19.143.
t=0 0.
m=audio 2234 RTP/AVP 18 8 0 101.
a=rtpmap:18 G729/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
#
U 2007/05/31 16:09:03.527834 10.28.19.202:5060 -> 10.28.19.143:5060
SIP/2.0 479 Regretfully, we were not able to process the URI (479/SL).
Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD.
From: "0660522015" <sip:0660522015@10.28.19.202>;tag=8DDF0DB-E8BCEB84.
To:
<sip:0672028405@10.28.19.202;user=phone>;tag=979d95a734c13f6db8b9e3a72b9f44a0.14e7.
CSeq: 2 INVITE.
Call-ID: 264aa927-d2a3a7c9-64f3fe3a(a)10.28.19.143.
Server: Sip EXpress router (0.9.6 (i386/linux)).
Content-Length: 0.
Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=8314
req_src_ip=10.28.19.143 req_src_port=5060
in_uri=sip:0672028405@10.28.19.202:5060;user=phone
out_uri=sip:0672028405@10.28.52.105:5060:5060;user=phone via_cnt==1".
Have you any suggestion about?
I use my Polycom with Asterisk and BroadSoft without any problem.
Thanks for your support and great patience :D
Bye,
F.