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
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