[OpenSIPS-Users] One Way Audio Using X-Lite, OpenSIPS & Asterisk (Ross Beer)
Ross Beer
ross_beer at hotmail.com
Fri Oct 23 15:19:45 CEST 2009
Hi,
Here is the sip trace for the calls, it strange as all other phone work except X-Lite, it appears that sound is not getting from the softphone to the server. Though if the softphone talks directly to asterisk it works ok. This is the case if MediaProxy is used or not.
There is a lack of codecs in the invite which is strange as I have 4 enabled on the server and the softphone.
Thank you for your help!
Ross
--------------------------------------------------------------------------------------------------
INVITE sip:160@<SERVER IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <ROUTER IP ADDRESS>:9302;branch=z9hG4bK-d8754z-4e352701ef65e42a-1---d8754z-;rport
Max-Forwards: 69
Contact: <sip:10002*200@<ROUTER IP ADDRESS>:9302>
To: "160"<sip:160@<SERVER IP ADDRESS>>
From: "Ross"<sip:10002*200@<SERVER IP ADDRESS>>;tag=ad038800
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 236
v=0
o=- 0 2 IN IP4 192.168.1.222
s=CounterPath X-Lite 3.0
c=IN IP4 <ROUTER IP ADDRESS>
t=0 0
m=audio 10006 RTP/AVP 0 101
a=alt:1 1 : 9ImKPFaa h9RvD+sl 192.168.1.222 10006
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
---------------------------------------------------------------------------------------------------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <SERVER IP ADDRESS>;branch=z9hG4bKfe32.d0a3adc2.0;received=<SERVER IP ADDRESS>
Via: SIP/2.0/UDP <ROUTER IP ADDRESS>:9302;received=<ROUTER IP ADDRESS>;branch=z9hG4bK-d8754z-4e352701ef65e42a-1---d8754z-;rport=9302
From: "Ross"<sip:10002*200@<SERVER IP ADDRESS>>;tag=ad038800
To: "160"<sip:160@<SERVER IP ADDRESS>>;tag=as57033d07
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 1 INVITE
User-Agent: Asterisk 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="<REALM>", nonce="6d81b1ec"
Content-Length: 0
---------------------------------------------------------------------------------------------------
ACK sip:160@<SERVER IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <ROUTER IP ADDRESS>:9302;branch=z9hG4bK-d8754z-4e352701ef65e42a-1---d8754z-;rport
To: "160"<sip:160@<SERVER IP ADDRESS>>;tag=as57033d07
From: "Ross"<sip:10002*200@<SERVER IP ADDRESS>>;tag=ad038800
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 1 ACK
Content-Length: 0
---------------------------------------------------------------------------------------------------
INVITE sip:160@<SERVER IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <ROUTER IP ADDRESS>:9302;branch=z9hG4bK-d8754z-063f0a0c111bba4a-1---d8754z-;rport
Max-Forwards: 69
Contact: <sip:10002*200@<ROUTER IP ADDRESS>:9302>
To: "160"<sip:160@<SERVER IP ADDRESS>>
From: "Ross"<sip:10002*200@<SERVER IP ADDRESS>>;tag=ad038800
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="10002*200",realm="<REALM>",nonce="6d81b1ec",uri="sip:160@<SERVER IP ADDRESS>",response="cbb367ca716a5e7c62ff962824996533",algorithm=MD5
Content-Length: 236
v=0
o=- 0 2 IN IP4 192.168.1.222
s=CounterPath X-Lite 3.0
c=IN IP4 <ROUTER IP ADDRESS>
t=0 0
m=audio 10006 RTP/AVP 0 101
a=alt:1 1 : 9ImKPFaa h9RvD+sl 192.168.1.222 10006
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
-----------------------------------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <SERVER IP ADDRESS>;branch=z9hG4bKce32.ba4f096.0;received=<SERVER IP ADDRESS>
Via: SIP/2.0/UDP <ROUTER IP ADDRESS>:9302;received=<ROUTER IP ADDRESS>;branch=z9hG4bK-d8754z-063f0a0c111bba4a-1---d8754z-;rport=9302
Record-Route: <sip:<SERVER IP ADDRESS>;lr=on;ftag=ad038800>
From: "Ross"<sip:10002*200@<SERVER IP ADDRESS>>;tag=ad038800
To: "160"<sip:160@<SERVER IP ADDRESS>>
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 2 INVITE
User-Agent: Asterisk 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:160@<SERVER IP ADDRESS>:5061>
Content-Length: 0
------------------------------------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP <SERVER IP ADDRESS>;branch=z9hG4bKce32.ba4f096.0;received=<SERVER IP ADDRESS>
Via: SIP/2.0/UDP <ROUTER IP ADDRESS>:9302;received=<ROUTER IP ADDRESS>;branch=z9hG4bK-d8754z-063f0a0c111bba4a-1---d8754z-;rport=9302
Record-Route: <sip:<SERVER IP ADDRESS>;lr=on;ftag=ad038800>
From: "Ross"<sip:10002*200@<SERVER IP ADDRESS>>;tag=ad038800
To: "160"<sip:160@<SERVER IP ADDRESS>>;tag=as0f58ca6e
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 2 INVITE
User-Agent: Asterisk 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:160@<SERVER IP ADDRESS>:5061>
Content-Type: application/sdp
Content-Length: 272
v=0
o=root 1609040672 1609040672 IN IP4 <SERVER IP ADDRESS>
s=Asterisk PBX 1.6.0.16-rc2
c=IN IP4 <SERVER IP ADDRESS>
t=0 0
m=audio 49356 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
----------------------------------------------------------------------------------------------------
ACK sip:160@<SERVER IP ADDRESS>:5061 SIP/2.0
Via: SIP/2.0/UDP <ROUTER IP ADDRESS>:9302;branch=z9hG4bK-d8754z-4b605e0b6855e57e-1---d8754z-;rport
Max-Forwards: 69
Route: <sip:<SERVER IP ADDRESS>;lr;ftag=ad038800>
Contact: <sip:10002*200@<ROUTER IP ADDRESS>:9302>
To: "160"<sip:160@<SERVER IP ADDRESS>>;tag=as0f58ca6e
From: "Ross"<sip:10002*200@<SERVER IP ADDRESS>>;tag=ad038800
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 2 ACK
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="10002*200",realm="<REALM>",nonce="6d81b1ec",uri="sip:160@<SERVER IP ADDRESS>",response="cbb367ca716a5e7c62ff962824996533",algorithm=MD5
Content-Length: 0
-----------------------------------------------------------------------------------------------------
BYE sip:160@<SERVER IP ADDRESS>:5061 SIP/2.0
Via: SIP/2.0/UDP <ROUTER IP ADDRESS>:9302;branch=z9hG4bK-d8754z-7f34bf77382c646c-1---d8754z-;rport
Max-Forwards: 69
Route: <sip:<SERVER IP ADDRESS>;lr;ftag=ad038800>
Contact: <sip:10002*200@<ROUTER IP ADDRESS>:9302>
To: "160"<sip:160@<SERVER IP ADDRESS>>;tag=as0f58ca6e
From: "Ross"<sip:10002*200@<SERVER IP ADDRESS>>;tag=ad038800
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 3 BYE
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="10002*200",realm="<REALM>",nonce="6d81b1ec",uri="sip:160@<SERVER IP ADDRESS>:5061",response="da1bb28c1d5e3afa093b9e97369c5ded",algorithm=MD5
Reason: SIP;description="User Hung Up"
Content-Length: 0
------------------------------------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP <SERVER IP ADDRESS>;branch=z9hG4bKde32.1e52a2a5.0;received=<SERVER IP ADDRESS>
Via: SIP/2.0/UDP <ROUTER IP ADDRESS>:9302;received=<ROUTER IP ADDRESS>;branch=z9hG4bK-d8754z-7f34bf77382c646c-1---d8754z-;rport=9302
From: "Ross"<sip:10002*200@<SERVER IP ADDRESS>>;tag=ad038800
To: "160"<sip:160@<SERVER IP ADDRESS>>;tag=as0f58ca6e
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 3 BYE
User-Agent: Asterisk 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
_________________________________________________________________
Chat to your friends for free on selected mobiles
http://clk.atdmt.com/UKM/go/174426567/direct/01/
More information about the Users
mailing list