[Kamailio-Users] I don't have asterisk audio to openser - mediaproxy

Wed Oct 29 12:12:29 CET 2008

Hi Ricky,Where is your Kamailio config?? is this your full ngrep capture??
Rgds,
Luzango.

>
> Message: 2
> Date: Tue, 28 Oct 2008 23:10:12 -0700 (PDT)
> From: Ricky Gutierrez <xserverlinux at yahoo.com>
> Subject: [Kamailio-Users] I don't have asterisk audio to openser -
>        mediaproxy
> To: users at lists.kamailio.org
> Message-ID: <654712.75990.qm at web59903.mail.ac4.yahoo.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi list is making tests with openser 1.3.2 and mediaproxy to solve the nat,
> I have gotten myself an ip it public with my supplier, I have two network
> cards in the pc that I am using for openser and mediaproxy together with
> asterisk, making tests with mediaproxy 1.9.1 when I receive a call from the
> pstn through asterisk I don't have audio, if I call to the pstn they listen
> to me well .
>
>
> From: "Ventas" <sip:112 at 192.168.10.1 <sip%3A112 at 192.168.10.1>
> >;tag=69451218021829df
> To: <sip:2685249 at 192.168.10.1 <sip%3A2685249 at 192.168.10.1>
> >;tag=329cfeaa6ded039da25ff8cbb8668bd2.b1b2
> Contact: <sip:112 at 192.168.10.30:5060;transport=udp>
> Supported: path
> Call-ID: fb5f5dac83056f72 at 192.168.10.30
> CSeq: 7492 ACK
> User-Agent: Grandstream GXP2020 1.1.6.16
> Max-Forwards: 70
> Allow:
> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Length: 0
>
>
> #
> U +0.022110 192.168.10.30:5060 -> 192.168.10.1:5060
> INVITE sip:2685249 at 192.168.10.1 <sip%3A2685249 at 192.168.10.1> SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03
> From: "Ventas" <sip:112 at 192.168.10.1 <sip%3A112 at 192.168.10.1>
> >;tag=69451218021829df
> To: <sip:2685249 at 192.168.10.1 <sip%3A2685249 at 192.168.10.1>>
> Contact: <sip:112 at 192.168.10.30:5060;transport=udp>
> Supported: replaces, timer, path
> Proxy-Authorization: Digest username="112", realm="192.168.10.1",
> algorithm=MD5, uri="sip:2685249 at 192.168.10.1 <sip%3A2685249 at 192.168.10.1>",
> nonce="4907ac8cb6dc757eb6ba5522e0fdb9786b4c3d6e",
> response="c40a9387fdf5de29115c1edadc7f79db"
> Call-ID: fb5f5dac83056f72 at 192.168.10.30
> CSeq: 7493 INVITE
> User-Agent: Grandstream GXP2020 1.1.6.16
> Max-Forwards: 70
> Allow:
> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Type: application/sdp
> Content-Length: 358
>
> v=0
> o=112 8000 8001 IN IP4 192.168.10.30
> s=SIP Call
> c=IN IP4 192.168.10.30
> t=0 0
> m=audio 5004 RTP/AVP 0 18 3 97 2 9 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=20
> a=rtpmap:2 G726-32/8000
> a=rtpmap:9 G722/16000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
>
> #
> U +0.003938 192.168.10.1:5060 -> 192.168.10.30:5060
> SIP/2.0 100 Giving a try
> Via: SIP/2.0/UDP 192.168.10.30:5060
> ;branch=z9hG4bKf428b928c25dad03;rport=5060
> From: "Ventas" <sip:112 at 192.168.10.1 <sip%3A112 at 192.168.10.1>
> >;tag=69451218021829df
> To: <sip:2685249 at 192.168.10.1 <sip%3A2685249 at 192.168.10.1>>
> Call-ID: fb5f5dac83056f72 at 192.168.10.30
> CSeq: 7493 INVITE
> Server: OpenSER (1.3.2-notls (i386/linux))
> Content-Length: 0
>
>
> #
> U +0.000115 192.168.10.1:5060 -> 192.168.10.1:5070
> INVITE sip:2685249 at 192.168.10.1:5070 SIP/2.0
> Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes>
> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0
> Via: SIP/2.0/UDP 192.168.10.30:5060
> ;rport=5060;branch=z9hG4bKf428b928c25dad03
> From: "Ventas" <sip:112 at 192.168.10.1 <sip%3A112 at 192.168.10.1>
> >;tag=69451218021829df
> To: <sip:2685249 at 192.168.10.1 <sip%3A2685249 at 192.168.10.1>>
> Contact: <sip:112 at 192.168.10.30:5060;transport=udp>
> Supported: replaces, timer, path
> Call-ID: fb5f5dac83056f72 at 192.168.10.30
> CSeq: 7493 INVITE
> User-Agent: Grandstream GXP2020 1.1.6.16
> Max-Forwards: 69
> Allow:
> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Type: application/sdp
> Content-Length: 358
> P-hint: route(3)|setflag7,forcerport,fix_contact
> P-hint: inbound->inbound
> P-hint: Route[6]: mediaproxy
>
> v=0
> o=112 8000 8001 IN IP4 192.168.10.30
> s=SIP Call
> c=IN IP4 192.168.1.64
> t=0 0
> m=audio 35040 RTP/AVP 0 18 3 97 2 9 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=20
> a=rtpmap:2 G726-32/8000
> a=rtpmap:9 G722/16000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
>
> #
> U +0.000471 192.168.10.1:5070 -> 192.168.10.1:5060
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0;received=
> 192.168.10.1
> Via: SIP/2.0/UDP 192.168.10.30:5060
> ;rport=5060;branch=z9hG4bKf428b928c25dad03
> Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes>
> From: "Ventas" <sip:112 at 192.168.10.1 <sip%3A112 at 192.168.10.1>
> >;tag=69451218021829df
> To: <sip:2685249 at 192.168.10.1 <sip%3A2685249 at 192.168.10.1>>
> Call-ID: fb5f5dac83056f72 at 192.168.10.30
> CSeq: 7493 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:2685249 at 192.168.10.1:5070>
> Content-Length: 0
>
>
> I don't have a lot of experience with mediaproxy, and I have some doubts
> that such you see they can help me to clarify, inside the file
> mediaproxy.ini some options appear which I have configured them but I am not
> sure if it is the best way.
>
> my scenario is the following one:
>
> <->  UAC<-> NAT  <-> ADSL  <-> Internet <->
> eth0 wan (public ip x.x.x.x ) <- openser/mediaproxy/asterisk -> eth1 lan (
> 192.168.11.1) <-> UAC
>
>
> [MediaProxy]
>
> start = yes
> socket = /var/run/mediaproxy.sock
> group = openser
> listen = None
> allow = None
> proxyIP = x.x.x.x (public ip)
> ;portRange = 60000:65000
> portRange = 35000:65000
> TOS = 0xb8
> idleTimeout = 60
> holdTimeout = 3600
> forceClose = 0
>
> [Accounting]
> ; one of none, radius or database
> accounting = none
>
> [Database]
> user = dbuser
> password = dbpass
> host = dbhost
> database = radius
> table = radacct
>
> [Radius]
> secret = secret
> server = localhost
> authport = 1812
> acctport = 1813
> dictionaries = /etc/radiusclient-ng/dictionary,
> /etc/openser/radius/dictionary, /usr/share/mediaproxy/dictionary
> retries = 2
> timeout = 3
>
>
>
>
> this couple of you line inside the openser, I don't still understand them
> according to the guide of ser getting started they are for asymmetric
> clients, but I don't find an example
>
> modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients")
> modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients")
>
> somebody that can give me a good help...
>
> regards
>
> rickygm
>
>
>
>

-- 
Luzango Mfupe
TUUNE MOBILE
Tel:0128440528/0123825710
Tshwane-RSA

"...Ships are safe in harbor, but they were never meant to stay
there......."
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