[Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
luzango mfupe
luzango.mfupe at gmail.com
Thu Oct 30 07:51:42 CET 2008
Hi Ricky
I should have seen how you handle NAT in kamaiilo.conf but you can also edit
sip.conf in Asterisk and try to put Nat=yesRgds,
On Wed, Oct 29, 2008 at 6:39 PM, Ricky Gutierrez <xserverlinux at yahoo.com>wrote:
> Hi luzano thank you for your help and time, this it is my full ngrep.
> I have asterisk, mediaproxy and openser together in the
> same pc
>
> I have the doubt that when a incoming call from the PSTN by asterisk,
> sends it to an extension of openser, to which I do not request
> authentication to him within invites, I believe that there it is where I
> have problems with mediaproxy
>
> that it leaves you want to see of the openser.cfg, or everything?
>
> regards ..
>
> interface: any
> filter: (ip) and ( port 5060 )
> #
> U +0.063740 192.168.10.1:5070 -> 192.168.10.1:5060
> INVITE sip:113 at 192.168.10.1 <sip%3A113 at 192.168.10.1> SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport
> From: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as30de9085
> To: <sip:113 at 192.168.10.1 <sip%3A113 at 192.168.10.1>>
> Contact: <sip:asterisk at 192.168.10.1:5070>
> Call-ID: 373456b8787e65d9764158381c9da273 at 192.168.10.1
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Wed, 29 Oct 2008 16:26:18 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 238
>
> v=0
> o=root 9850 9850 IN IP4 192.168.10.1
> s=session
> c=IN IP4 192.168.10.1
> t=0 0
> m=audio 14750 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
>
> #
> U +0.004203 192.168.10.1:5060 -> 192.168.10.1:5070
> SIP/2.0 100 Giving a try
> Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070
> From: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as30de9085
> To: <sip:113 at 192.168.10.1 <sip%3A113 at 192.168.10.1>>
> Call-ID: 373456b8787e65d9764158381c9da273 at 192.168.10.1
> CSeq: 102 INVITE
> Server: OpenSER (1.3.2-notls (i386/linux))
> Content-Length: 0
>
>
> #
> U +0.000275 192.168.10.1:5060 -> 192.168.10.30:5062
> INVITE sip:113 at 192.168.10.30:5062;transport=udp SIP/2.0
> Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes>
> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0
> Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070
> From: "asterisk"
> <sip:asterisk at 192.168.10.1:5070>;tag=as30de9085
> To: <sip:113 at 192.168.10.1 <sip%3A113 at 192.168.10.1>>
> Contact: <sip:asterisk at 192.168.10.1:5070>
> Call-ID: 373456b8787e65d9764158381c9da273 at 192.168.10.1
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 69
> Date: Wed, 29 Oct 2008 16:26:18 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 238
> P-hint: route(3)|setflag7,forcerport,fix_contact
> P-hint: inbound->inbound
> P-hint: Route[6]: mediaproxy
>
> v=0
> o=root 9850 9850 IN IP4 192.168.10.1
> s=session
> c=IN IP4 192.168.1.64
> t=0 0
> m=audio 35058 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
>
> #
> U +0.026845 192.168.10.30:5062 -> 192.168.10.1:5060
> SIP/2.0 100 Trying
> Via:
> SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0
> Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070
> From: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as30de9085
> To: <sip:113 at 192.168.10.1 <sip%3A113 at 192.168.10.1>>
> Call-ID: 373456b8787e65d9764158381c9da273 at 192.168.10.1
> CSeq: 102 INVITE
> User-Agent: Grandstream GXP2020 1.1.6.16
> Content-Length: 0
>
>
> #
> U +0.009886 192.168.10.30:5062 -> 192.168.10.1:5060
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0
> Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070
> Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes>
> From: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as30de9085
> To: <sip:113 at 192.168.10.1 <sip%3A113 at 192.168.10.1>>;tag=21c220c2e075d838
> Call-ID: 373456b8787e65d9764158381c9da273 at 192.168.10.1
> CSeq: 102 INVITE
> User-Agent: Grandstream GXP2020 1.1.6.16
> Contact:
> <sip:113 at 192.168.10.30:5062;transport=udp>
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Length: 0
>
>
> #
> U +0.000169 192.168.10.1:5060 -> 192.168.10.1:5070
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070
> Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes>
> From: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as30de9085
> To: <sip:113 at 192.168.10.1 <sip%3A113 at 192.168.10.1>>;tag=21c220c2e075d838
> Call-ID: 373456b8787e65d9764158381c9da273 at 192.168.10.1
> CSeq: 102 INVITE
> User-Agent: Grandstream GXP2020 1.1.6.16
> Contact: <sip:113 at 192.168.10.30:5062;transport=udp>
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Length: 0
> P-hint: Onreply-route - fixcontact
>
>
> #
> U +0.000140 190.184.35.4:5060 -> 192.168.1.64:5064
> OPTIONS
> sip:130 at 192.168.1.64:5064 SIP/2.0
> Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport
> From: "asterisk" <sip:asterisk at 190.184.35.4 <sip%3Aasterisk at 190.184.35.4>>;tag=as35db6300
> To: <sip:130 at 192.168.1.64:5064>
> Contact: <sip:asterisk at 190.184.35.4 <sip%3Aasterisk at 190.184.35.4>>
> Call-ID: 16e287fc2b24afb97eb1759952eff3d3 at 190.184.35.4
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Wed, 29 Oct 2008 16:26:18 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> #
> U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064
> OPTIONS sip:130 at 192.168.1.64:5064 SIP/2.0
> Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport
> From: "asterisk" <sip:asterisk at 190.184.35.4 <sip%3Aasterisk at 190.184.35.4>>;tag=as35db6300
> To: <sip:130 at 192.168.1.64:5064>
> Contact: <sip:asterisk at 190.184.35.4 <sip%3Aasterisk at 190.184.35.4>>
> Call-ID: 16e287fc2b24afb97eb1759952eff3d3 at 190.184.35.4
> CSeq: 102
> OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Wed, 29 Oct 2008 16:26:18 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> #
> U +0.008208 192.168.10.19:5064 -> 190.184.35.4:5060
> SIP/2.0 200 OK
> To: <sip:130 at 192.168.1.64:5064>;tag=9611a90af8ab9321i1
> From: "asterisk" <sip:asterisk at 190.184.35.4 <sip%3Aasterisk at 190.184.35.4>>;tag=as35db6300
> Call-ID: 16e287fc2b24afb97eb1759952eff3d3 at 190.184.35.4
> CSeq: 102 OPTIONS
> Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282
> Server: Linksys/SPA942-5.2.8
> Content-Length: 0
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: replaces
>
>
> #
> U +0.000014 192.168.1.64:5064 -> 190.184.35.4:5060
> SIP/2.0 200 OK
> To: <sip:130 at 192.168.1.64:5064>;tag=9611a90af8ab9321i1
> From: "asterisk" <sip:asterisk at 190.184.35.4 <sip%3Aasterisk at 190.184.35.4>>;tag=as35db6300
> Call-ID:
> 16e287fc2b24afb97eb1759952eff3d3 at 190.184.35.4
> CSeq: 102 OPTIONS
> Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282
> Server: Linksys/SPA942-5.2.8
> Content-Length: 0
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: replaces
>
>
> #
> U +0.956449 192.168.10.19:5064 -> 190.184.35.4:5060
> NOTIFY sip:190.184.35.4 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219
> From: <sip:130 at 190.184.35.4 <sip%3A130 at 190.184.35.4>>;tag=7c3557f6145bd125o1
> To: <sip:190.184.35.4>
> Call-ID: ce8ea9b2-baee4a99 at 192.168.10.19
> CSeq: 12 NOTIFY
> Max-Forwards: 70
> Event: keep-alive
> User-Agent: Linksys/SPA942-5.2.8
> Content-Length: 0
>
>
> #
> U +0.000027 192.168.1.64:5064 -> 190.184.35.4:5060
> NOTIFY sip:190.184.35.4 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219
> From: <sip:130 at 190.184.35.4 <sip%3A130 at 190.184.35.4>>;tag=7c3557f6145bd125o1
> To:
> <sip:190.184.35.4>
> Call-ID: ce8ea9b2-baee4a99 at 192.168.10.19
> CSeq: 12 NOTIFY
> Max-Forwards: 70
> Event: keep-alive
> User-Agent: Linksys/SPA942-5.2.8
> Content-Length: 0
>
>
> #
> U +0.156145 190.184.35.4:5060 -> 192.168.1.64:5064
> SIP/2.0 489 Bad event
> Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64
> From: <sip:130 at 190.184.35.4 <sip%3A130 at 190.184.35.4>>;tag=7c3557f6145bd125o1
> To: <sip:190.184.35.4>;tag=as2002c003
> Call-ID: ce8ea9b2-baee4a99 at 192.168.10.19
> CSeq: 12 NOTIFY
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> #
> U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064
> SIP/2.0 489 Bad event
> Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64
> From: <sip:130 at 190.184.35.4 <sip%3A130 at 190.184.35.4>>;tag=7c3557f6145bd125o1
> To:
> <sip:190.184.35.4>;tag=as2002c003
> Call-ID: ce8ea9b2-baee4a99 at 192.168.10.19
> CSeq: 12 NOTIFY
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> #
> U +5.895003 192.168.10.30:5062 -> 192.168.10.1:5060
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0
> Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070
> Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes>
> From: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as30de9085
> To: <sip:113 at 192.168.10.1 <sip%3A113 at 192.168.10.1>>;tag=21c220c2e075d838
> Call-ID: 373456b8787e65d9764158381c9da273 at 192.168.10.1
> CSeq: 102 INVITE
> User-Agent: Grandstream GXP2020 1.1.6.16
> Contact: <sip:113 at 192.168.10.30:5062;transport=udp>
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
>
> Content-Type: application/sdp
> Supported: replaces, timer
> Content-Length: 212
>
> v=0
> o=113 8000 8000 IN IP4 192.168.10.30
> s=SIP Call
> c=IN IP4 192.168.10.30
> t=0 0
> m=audio 5004 RTP/AVP 0 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
>
> #
> U +0.001400 192.168.10.1:5060 -> 192.168.10.1:5070
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070
> Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes>
> From: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as30de9085
> To: <sip:113 at 192.168.10.1 <sip%3A113 at 192.168.10.1>>;tag=21c220c2e075d838
> Call-ID: 373456b8787e65d9764158381c9da273 at 192.168.10.1
> CSeq: 102 INVITE
> User-Agent: Grandstream GXP2020 1.1.6.16
> Contact: <sip:113 at 192.168.10.30:5062;transport=udp>
> Allow:
> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Type: application/sdp
> Supported: replaces, timer
> Content-Length: 212
> P-hint: Onreply-route - fixcontact
> P-hint: onreply_route|usemediaproxy
>
> v=0
> o=113 8000 8000 IN IP4 192.168.10.30
> s=SIP Call
> c=IN IP4 192.168.1.64
> t=0 0
> m=audio 35058 RTP/AVP 0 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
>
> #
> U +0.000557 192.168.10.1:5070 -> 192.168.10.1:5060
> ACK sip:113 at 192.168.10.30:5062;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport
> Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes>
> From: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as30de9085
> To: <sip:113 at 192.168.10.1 <sip%3A113 at 192.168.10.1>>;tag=21c220c2e075d838
> Contact: <sip:asterisk at 192.168.10.1:5070>
> Call-ID:
> 373456b8787e65d9764158381c9da273 at 192.168.10.1
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> #
> U +0.000215 192.168.10.1:5060 -> 192.168.10.30:5062
> ACK sip:113 at 192.168.10.30:5062;transport=udp SIP/2.0
> Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes>
> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.2
> Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport=5070
> From: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as30de9085
> To: <sip:113 at 192.168.10.1 <sip%3A113 at 192.168.10.1>>;tag=21c220c2e075d838
> Contact: <sip:asterisk at 192.168.10.1:5070>
> Call-ID: 373456b8787e65d9764158381c9da273 at 192.168.10.1
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 69
> Content-Length: 0
> P-hint: LR|fixcontact,setflag6
>
>
> #
> U +0.886545 192.168.10.28:5060 -> 192.168.10.1:5060
>
>
>
> ------------------------------
> *From:* luzango mfupe <luzango.mfupe at gmail.com>
> *To:* users at lists.kamailio.org
> *Sent:* Wednesday, October 29, 2008 5:12:29 AM
> *Subject:* Re: [Kamailio-Users] I don't have asterisk audio to openser -
> mediaproxy
>
>
> Hi Ricky,Where is your Kamailio config?? is this your full ngrep capture??
> Rgds,
> Luzango.
>
>
>
>
--
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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