Do you have your softphone setup to use a stun server so it can send it's public IP address in the SIP packets? I see in the SIP debug output a 192.168 address for the RTP packets to go to which of course will not work. -- Jim Dickenson mailto:dicken...@cfmc.com
CfMC http://www.cfmc.com/ On Jul 28, 2010, at 9:23 AM, Nasir Javaid wrote: > hi there, > > i have posted earlier on the list but got no satisfying answer. the problem > is not big. > > I have asterisk server directly connected with internet (79.80.x.x) and > clients are behind router. clients/users are registered with asterisk and are > using sipura and xlite softphone. > > Now problem is that when a user calls other by dialing his IP:Port (sip uri), > call is connected fine and he can hear the called user but the called user > can not here the caller voice. > > If the caller calls the other user by username instead of IP:Port , the voice > is perfect both ways. > > what i have noticed is that IP:Port dial is missing a parameter "rinstance" > in "Contact" , "To" headers for adf. what is "rinstance" for? Also something > with "Contact" header seems fishy. or RTP issue. > > that is > > Dial(SIP/adf,30,r) works fine with bothway audio, but > > Dial(SIP/116.18.35.235:28614,30,r) has one way audio. > / \ > | | > this is IP:Port of of adf > > please help as it's almost 2 weeks and i have found to suitable answer from > any forum. I nead to know what can i do to modify Headers or settings in conf > files to correct this problem. > > Below is the conf of calling user > > [pepsi] > username=pepsi > type=friend > secret=123456 > qualify=yes > nat=no > insecure=port,invite > incominglimit=1 > outgoinglimit=1 > host=dynamic > dtmfmode=rfc2833 > context=out > canreinvite=yes > callerid="pepsi coke" <12345678901> > accountcode=6:0:pepsi > amaflags=default > disallow=all > allow=alaw > allow=ulaw > allow=g729 > allow=gsm > > Below is the conf of called user > > [adf] > username=adf > type=friend > secret=123456 > qualify=yes > nat=yes > insecure=port,invite > incominglimit=2 > outgoinglimit=2 > host=dynamic > dtmfmode=rfc2833 > context=user > canreinvite=yes > callerid="adf xyz" <11223344556> > accountcode=1:0:adf > amaflags=default > disallow=all > allow=g729 > allow=ulaw > allow=alaw > allow=gsm > > > > below is my sip debug after dialing > > Audio is at 79.80.x.x port 16238 > Adding codec 0x8 (alaw) to SDP > Adding codec 0x4 (ulaw) to SDP > Adding codec 0x2 (gsm) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (NAT) to 116.18.35.235:28614: > INVITE sip:a...@116.18.35.235:28614 SIP/2.0 > Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport > From: "pepsi coke" <sip:12345678...@79.80.x.x:5678>;tag=as42ec768c > To: <sip:a...@116.18.35.235:28614> > Contact: <sip:12345678...@79.80.x.x:5678> > Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Wed, 21 Jul 2010 15:10:22 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 285 > > v=0 > o=root 9626 9626 IN IP4 79.80.x.x > s=session > c=IN IP4 79.80.x.x > t=0 0 > m=audio 16238 RTP/AVP 8 0 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > --- > [Jul 21 11:10:22] WARNING[23814]: chan_sip.c:2872 sip_call: Setting > auto-congest time to 15000 ms. > -- Called a...@116.18.35.235:28614 > <------------> > ast-server*CLI> > <--- SIP read from 116.18.35.235:28614 ---> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678 > Contact: <sip:a...@116.18.35.235:28614> > To: <sip:a...@116.18.35.235:28614>;tag=d54e632c > From: "pepsi coke"<sip:12345678...@79.80.x.x:5678>;tag=as42ec768c > Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x > CSeq: 102 INVITE > User-Agent: X-Lite release 1104o stamp 56125 > Content-Length: 0 > > > <-------------> > --- (9 headers 0 lines) --- > -- SIP/116.18.35.235:28614-007f4660 is ringing > ast-server*CLI> > <--- SIP read from 116.18.35.235:28614 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678 > Contact: <sip:a...@116.18.35.235:28614> > To: <sip:a...@116.18.35.235:28614>;tag=d54e632c > From: "pepsi coke"<sip:12345678...@79.80.x.x:5678>;tag=as42ec768c > Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x > CSeq: 102 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, > INFO > Content-Type: application/sdp > User-Agent: X-Lite release 1104o stamp 56125 > Content-Length: 185 > > v=0 > o=- 6 2 IN IP4 192.168.0.12 > s=CounterPath X-Lite 3.0 > c=IN IP4 192.168.0.12 > t=0 0 > m=audio 55246 RTP/AVP 8 0 101 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=sendrecv > > <-------------> > --- (11 headers 9 lines) --- > Found RTP audio format 8 > Found RTP audio format 0 > Found RTP audio format 101 > Peer audio RTP is at port 192.168.0.12:55246 > Found description format telephone-event for ID 101 > Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xc > (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) > Peer audio RTP is at port 192.168.0.12:55246 > list_route: hop: <sip:a...@116.18.35.235:28614> > [Jul 21 11:10:27] DEBUG[9707]: chan_sip.c:5695 reqprep: Strict routing > enforced for session 0433af7878e3a8067a40f896382cc...@79.80.x.x > set_destination: Parsing <sip:a...@116.18.35.235:28614> for address/port to > send to > set_destination: set destination to 116.18.35.235, port 28614 > Transmitting (NAT) to 116.18.35.235:28614: > ACK sip:a...@116.18.35.235:28614 SIP/2.0 > Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK07eb06b5;rport > From: "pepsi coke" <sip:12345678...@79.80.x.x:5678>;tag=as42ec768c > To: <sip:a...@116.18.35.235:28614>;tag=d54e632c > Contact: <sip:12345678...@79.80.x.x:5678> > Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > > --- > -- Call on SIP/116.18.35.235:28614-007f4660 left from hold > -- SIP/116.18.35.235:28614-007f4660 answered SIP/pepsi-9fdfb9a0 > ast-server*CLI> > <--- SIP read from 116.18.35.235:28614 ---> > > > > <-------------> > --- (0 headers 1 lines) --- > ast-server*CLI> > <--- SIP read from 116.18.35.235:28614 ---> > > > > <-------------> > --- (0 headers 1 lines) --- > Scheduling destruction of SIP dialog > '0433af7878e3a8067a40f896382cc...@79.80.x.x' in 32000 ms (Method: INVITE) > [Jul 21 11:11:03] DEBUG[23814]: chan_sip.c:5695 reqprep: Strict routing > enforced for session 0433af7878e3a8067a40f896382cc...@79.80.x.x > set_destination: Parsing <sip:a...@116.18.35.235:28614> for address/port to > send to > set_destination: set destination to 116.18.35.235, port 28614 > Reliably Transmitting (NAT) to 116.18.35.235:28614: > BYE sip:a...@116.18.35.235:28614 SIP/2.0 > Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK36df65b5;rport > From: "pepsi coke" <sip:12345678...@79.80.x.x:5678>;tag=as42ec768c > To: <sip:a...@116.18.35.235:28614>;tag=d54e632c > Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x > CSeq: 103 BYE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > > > > Nasir Javaid > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
