[OpenSIPS-Users] [OPENSIPS] How to route calls out Openser to Voicemail GW with Right RURI
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Thu Apr 9 13:17:25 CEST 2009
Hi,
yes, Asterisk will send media to RTPproxy IP and not to the UAC. Looking
at the first trace you send, I see that rtpporxy was not set for the 200
OK reply (only for INVITE request) - because of this, the media is broken.
Regards,
Bogdan
oso che bol wrote:
> Dear Bogdan,
>
> My current Problem is: My Asterisk VOICEMAIL app, /which opensips
> foward to/, return voice announcement to UA OF OPENSIPS, and expected
> results will be: we could here voice of that. But, actually result is
> that we could not hear anything.
>
> I also attach trace of Asterisk communication with Opensips when
> INVITE come out opensips to Asterisk. One notice that Invite come out
> Opensips also involve RTPPROXY IP :(, maybe that is glue of my
> problem? And we have 2 ACK from opensips to Asterisk at the last of trace.
>
> Thanks and Regards,
> -LN
>
> ==========TRACE of ASTERISK=============
>
> filter: (ip) and ( port 5060 )
>
> U OPENSIPS_IP:5060 -> ASTERISK_IP:5060
> INVITE sip:3000 at ASTERISK_IP:5060 SIP/2.0.
> Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>.
> Via: SIP/2.0/UDP OPENSIPS_IP;branch=z9hG4bK8303.a0576353.1.
> Via: SIP/2.0/UDP
> 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-b676bc4d1975ea69-1---d8754z-;rport=25480.
> Max-Forwards: 69.
> Contact: <sip:6000 at 118.69.139.66:25480
> <http://sip:6000@118.69.139.66:25480>>.
> To: "3000"<sip:3000 at OPENSIPS_IP>.
> From: "6000"<sip:6000 at OPENSIPS_IP>;tag=43241208.
> Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk..
> CSeq: 1 INVITE.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO.
> Content-Type: application/sdp.
> User-Agent: X-Lite release 1100l stamp 47546.
> Content-Length: 443.
> P-hint: HTK: Fix nated contact.
> P-hint: HTK: INVITE go to on_reply_route[1].
> P-hint: INVITE||ACK + FORCE_RTP_PROXY.
> P-hint: HTK - 408 and route to VM ... .
> .
> v=0.
> o=- 8 2 IN IP4 192.168.1.150.
> s=CounterPath X-Lite 3.0.
> c=IN IP4 *RTPPROXY_IP.*
> t=0 0.
> m=audio 42762 RTP/AVP 107 119 100 106 0 105 98 8 3 101.
> a=alt:1 1 : pt1QlLaH Tih+tcui 192.168.1.150 50854.
> a=fmtp:101 0-15.
> a=rtpmap:107 BV32/16000.
> a=rtpmap:119 BV32-FEC/16000.
> a=rtpmap:100 SPEEX/16000.
> a=rtpmap:106 SPEEX-FEC/16000.
> a=rtpmap:105 SPEEX-FEC/8000.
> a=rtpmap:98 iLBC/8000.
> a=rtpmap:101 telephone-event/8000.
> a=sendrecv.
> a=nortpproxy:yes.
>
>
> U ASTERISK_IP:5060 -> OPENSIPS_IP:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP
> OPENSIPS_IP;branch=z9hG4bK8303.a0576353.1;received=OPENSIPS_IP.
> Via: SIP/2.0/UDP
> 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-b676bc4d1975ea69-1---d8754z-;rport=25480.
> Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>.
> From: "6000"<sip:6000 at OPENSIPS_IP>;tag=43241208.
> To: "3000"<sip:3000 at OPENSIPS_IP>.
> Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk..
> CSeq: 1 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces.
> Contact: <sip:3000 at ASTERISK_IP>.
> Content-Length: 0.
> .
>
>
> U ASTERISK_IP:5060 -> OPENSIPS_IP:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP
> OPENSIPS_IP;branch=z9hG4bK8303.a0576353.1;received=OPENSIPS_IP.
> Via: SIP/2.0/UDP
> 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-b676bc4d1975ea69-1---d8754z-;rport=25480.
> Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>.
> From: "6000"<sip:6000 at OPENSIPS_IP>;tag=43241208.
> To: "3000"<sip:3000 at OPENSIPS_IP>;tag=as39e194d1.
> Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk..
> CSeq: 1 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces.
> Contact: <sip:3000 at ASTERISK_IP>.
> Content-Length: 0.
> .
>
>
> U ASTERISK_IP:5060 -> OPENSIPS_IP:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> OPENSIPS_IP;branch=z9hG4bK8303.a0576353.1;received=OPENSIPS_IP.
> Via: SIP/2.0/UDP
> 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-b676bc4d1975ea69-1---d8754z-;rport=25480.
> Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>.
> From: "6000"<sip:6000 at OPENSIPS_IP>;tag=43241208.
> To: "3000"<sip:3000 at OPENSIPS_IP>;tag=as39e194d1.
> Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk..
> CSeq: 1 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces.
> Contact: <sip:3000 at ASTERISK_IP>.
> Content-Type: application/sdp.
> Content-Length: 289.
> .
> v=0.
> o=root 9042 9042 IN IP4 ASTERISK_IP.
> s=session.
> c=IN IP4 ASTERISK_IP.
> t=0 0.
> m=audio 12610 RTP/AVP 3 0 8 101.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
>
>
> U ASTERISK_IP:5060 -> OPENSIPS_IP:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> OPENSIPS_IP;branch=z9hG4bK8303.a0576353.1;received=OPENSIPS_IP.
> Via: SIP/2.0/UDP
> 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-b676bc4d1975ea69-1---d8754z-;rport=25480.
> Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>.
> From: "6000"<sip:6000 at OPENSIPS_IP>;tag=43241208.
> To: "3000"<sip:3000 at OPENSIPS_IP>;tag=as39e194d1.
> Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk..
> CSeq: 1 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces.
> Contact: <sip:3000 at ASTERISK_IP>.
> Content-Type: application/sdp.
> Content-Length: 289.
> .
> v=0.
> o=root 9042 9042 IN IP4 ASTERISK_IP.
> s=session.
> c=IN IP4 ASTERISK_IP.
> t=0 0.
> m=audio 12610 RTP/AVP 3 0 8 101.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
>
>
> U OPENSIPS_IP:5060 -> ASTERISK_IP:5060
> ACK sip:3000 at ASTERISK_IP SIP/2.0.
> Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>.
> Via: SIP/2.0/UDP OPENSIPS_IP;branch=z9hG4bK8303.a0576353.3.
> Via: SIP/2.0/UDP
> 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-ca5726034435a134-1---d8754z-;rport=25480.
> Max-Forwards: 69.
> Contact: <sip:6000 at 118.69.139.66:25480
> <http://sip:6000@118.69.139.66:25480>>.
> To: "3000"<sip:3000 at OPENSIPS_IP>;tag=as39e194d1.
> From: "6000"<sip:6000 at OPENSIPS_IP>;tag=43241208.
> Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk..
> CSeq: 1 ACK.
> User-Agent: X-Lite release 1100l stamp 47546.
> Content-Length: 0.
> P-hint: HTK: rr-enforced.
> P-hint: Route[1] Processing.
> .
>
>
> U OPENSIPS_IP:5060 -> ASTERISK_IP:5060
> ACK sip:3000 at ASTERISK_IP SIP/2.0.
> Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>.
> Via: SIP/2.0/UDP OPENSIPS_IP;branch=z9hG4bK8303.a0576353.3.
> Via: SIP/2.0/UDP
> 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-ca5726034435a134-1---d8754z-;rport=25480.
> Max-Forwards: 69.
> Contact: <sip:6000 at 118.69.139.66:25480
> <http://sip:6000@118.69.139.66:25480>>.
> To: "3000"<sip:3000 at OPENSIPS_IP>;tag=as39e194d1.
> From: "6000"<sip:6000 at OPENSIPS_IP>;tag=43241208.
> Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk..
> CSeq: 1 ACK.
> User-Agent: X-Lite release 1100l stamp 47546.
> Content-Length: 0.
> P-hint: HTK: rr-enforced.
> P-hint: Route[1] Processing.
> .
>
>
>
> On Thu, Apr 9, 2009 at 4:25 PM, Bogdan-Andrei Iancu
> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>
> Hi,
>
> The script and signalling are ok and looks good. Maybe you can
> detail a bit what you do not like and what you want to change.
>
> Regards,
> Bogdan
>
> oso che bol wrote:
>
> Dear All,
>
> Platform:
> - Opensip 1.4.5
> - RTPPROXY: 1.2.0
> - MySQL: 5.0
> - Asterisk 1.4.24
>
> Asterisk acts as a VoiceMail system when Users on Opensips Do
> Not answer or Busy, so call will route to Asterisk to leave
> Voicemail.
> Voicemail system works properly.
>
> The problem is: "When call route out of Opensips Server to
> Asterisk, Request URI use IP of Opensips to request Asterisk".
>
> So, Asterisk will return to wrong IP of UA Client, and of
> course, no voice from Asterisk to UA Client.
>
> Expected Results:1. "I want to use 1000 at IP_OF_UACLient to
> request Asterisk for leave voicemail?" or "If it uses
> 1000 at IP_OF_OPENSIPS to request Asterisk for Voicemail, UA
> Client should hear Voice".
>
> Bellow is my Configuration Files and Log File of OpenSIPs.
>
> Thanks and Regards,
> -LN
>
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