<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">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.<br><div>
<span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Monaco; ">-- </div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Monaco; ">Jim Dickenson</div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Monaco; "><a href="mailto:dickenson@cfmc.com">mailto:dickenson@cfmc.com</a></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Monaco; min-height: 14px; "><br></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Monaco; ">CfMC</div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Monaco; "><a href="http://www.cfmc.com/">http://www.cfmc.com/</a></div><div><font class="Apple-style-span" face="Monaco" size="2"><span class="Apple-style-span" style="font-size: 10px; "><br></span></font></div></div></span><br class="Apple-interchange-newline">
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<br><div><div>On Jul 28, 2010, at 9:23 AM, Nasir Javaid wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">hi there,<br><br>i have posted earlier on the list but got no satisfying answer. the problem is not big.<br><br>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.<br>
<br>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.<br><br>If the caller calls the other user by username instead of IP:Port , the voice is perfect both ways. <br>
<br>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.<br>
<br>that is <br><br>Dial(SIP/adf,30,r) works fine with bothway audio, but<br><br>Dial(SIP/<a href="http://116.18.35.235:28614/">116.18.35.235:28614</a>,30,r) has one way audio.<br> / \<br>
| | <br> this is IP:Port of of adf<br><br>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.<br>
<br>Below is the conf of calling user<br><br>[pepsi]<br>username=pepsi<br>type=friend<br>secret=123456<br>qualify=yes<br>nat=no<br>insecure=port,invite<br>incominglimit=1<br>outgoinglimit=1<br>host=dynamic<br>dtmfmode=rfc2833<br>
context=out<br>canreinvite=yes<br>callerid="pepsi coke" <12345678901><br>accountcode=6:0:pepsi<br>amaflags=default<br>disallow=all<br>allow=alaw<br>allow=ulaw<br>allow=g729<br>allow=gsm<br><br>Below is the conf of called user<br>
<br>[adf]<br>username=adf<br>type=friend<br>secret=123456<br>qualify=yes<br>nat=yes<br>insecure=port,invite<br>incominglimit=2<br>outgoinglimit=2<br>host=dynamic<br>dtmfmode=rfc2833<br>context=user<br>canreinvite=yes<br>callerid="adf xyz" <11223344556><br>
accountcode=1:0:adf<br>amaflags=default<br>disallow=all<br>allow=g729<br>allow=ulaw<br>allow=alaw<br>allow=gsm<br><br><br><br>below is my sip debug after dialing<br><br>Audio is at 79.80.x.x port 16238<br>Adding codec 0x8 (alaw) to SDP<br>
Adding codec 0x4 (ulaw) to SDP<br>Adding codec 0x2 (gsm) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br>Reliably Transmitting (NAT) to <a href="http://116.18.35.235:28614/">116.18.35.235:28614</a>:<br>INVITE <a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a> SIP/2.0<br>
Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport<br>From: "pepsi coke" <<a href="sip:12345678901@79.80.x.x:5678">sip:12345678901@79.80.x.x:5678</a>>;tag=as42ec768c<br>To: <<a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a>><br>
Contact: <<a href="sip:12345678901@79.80.x.x:5678">sip:12345678901@79.80.x.x:5678</a>><br>Call-ID: 0433af7878e3a8067a40f896382cc3a6@79.80.x.x<br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Date: Wed, 21 Jul 2010 15:10:22 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>Content-Type: application/sdp<br>Content-Length: 285<br><br>v=0<br>o=root 9626 9626 IN IP4 79.80.x.x<br>s=session<br>c=IN IP4 79.80.x.x<br>t=0 0<br>m=audio 16238 RTP/AVP 8 0 3 101<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:0 PCMU/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>---<br>[Jul 21 11:10:22] WARNING[23814]: chan_sip.c:2872 sip_call: Setting auto-congest time to 15000 ms.<br>
-- Called <a href="http://adf@116.18.35.235:28614/">adf@116.18.35.235:28614</a><br><------------><br>ast-server*CLI> <br><--- SIP read from <a href="http://116.18.35.235:28614/">116.18.35.235:28614</a> ---><br>
SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678<br>Contact: <<a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a>><br>To: <<a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a>>;tag=d54e632c<br>
From: "pepsi coke"<<a href="sip:12345678901@79.80.x.x:5678">sip:12345678901@79.80.x.x:5678</a>>;tag=as42ec768c<br>Call-ID: 0433af7878e3a8067a40f896382cc3a6@79.80.x.x<br>CSeq: 102 INVITE<br>User-Agent: X-Lite release 1104o stamp 56125<br>Content-Length: 0<br>
<br><br><-------------><br>--- (9 headers 0 lines) ---<br> -- SIP/116.18.35.235:28614-007f4660 is ringing<br>ast-server*CLI> <br><--- SIP read from <a href="http://116.18.35.235:28614/">116.18.35.235:28614</a> ---><br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678<br>Contact: <<a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a>><br>To: <<a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a>>;tag=d54e632c<br>
From: "pepsi coke"<<a href="sip:12345678901@79.80.x.x:5678">sip:12345678901@79.80.x.x:5678</a>>;tag=as42ec768c<br>Call-ID: 0433af7878e3a8067a40f896382cc3a6@79.80.x.x<br>CSeq: 102 INVITE<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>
Content-Type: application/sdp<br>User-Agent: X-Lite release 1104o stamp 56125<br>Content-Length: 185<br><br>v=0<br>o=- 6 2 IN IP4 192.168.0.12<br>s=CounterPath X-Lite 3.0<br>c=IN IP4 192.168.0.12<br>t=0 0<br>m=audio 55246 RTP/AVP 8 0 101<br>
a=fmtp:101 0-15<br>a=rtpmap:101 telephone-event/8000<br>a=sendrecv<br><br><-------------><br>--- (11 headers 9 lines) ---<br>Found RTP audio format 8<br>Found RTP audio format 0<br>Found RTP audio format 101<br>Peer audio RTP is at port <a href="http://192.168.0.12:55246/">192.168.0.12:55246</a><br>
Found description format telephone-event for ID 101<br>Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)<br>Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>
Peer audio RTP is at port <a href="http://192.168.0.12:55246/">192.168.0.12:55246</a><br>list_route: hop: <<a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a>><br>[Jul 21 11:10:27] DEBUG[9707]: chan_sip.c:5695 reqprep: Strict routing enforced for session 0433af7878e3a8067a40f896382cc3a6@79.80.x.x<br>
set_destination: Parsing <<a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a>> for address/port to send to<br>set_destination: set destination to 116.18.35.235, port 28614<br>Transmitting (NAT) to <a href="http://116.18.35.235:28614/">116.18.35.235:28614</a>:<br>
ACK <a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a> SIP/2.0<br>Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK07eb06b5;rport<br>From: "pepsi coke" <<a href="sip:12345678901@79.80.x.x:5678">sip:12345678901@79.80.x.x:5678</a>>;tag=as42ec768c<br>
To: <<a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a>>;tag=d54e632c<br>Contact: <<a href="sip:12345678901@79.80.x.x:5678">sip:12345678901@79.80.x.x:5678</a>><br>Call-ID: 0433af7878e3a8067a40f896382cc3a6@79.80.x.x<br>CSeq: 102 ACK<br>
User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Content-Length: 0<br><br><br>---<br> -- Call on SIP/116.18.35.235:28614-007f4660 left from hold<br> -- SIP/116.18.35.235:28614-007f4660 answered SIP/pepsi-9fdfb9a0<br>
ast-server*CLI> <br><--- SIP read from <a href="http://116.18.35.235:28614/">116.18.35.235:28614</a> ---><br><br><br><br><-------------><br>--- (0 headers 1 lines) ---<br>ast-server*CLI> <br><--- SIP read from <a href="http://116.18.35.235:28614/">116.18.35.235:28614</a> ---><br>
<br><br><br><-------------><br>--- (0 headers 1 lines) ---<br>Scheduling destruction of SIP dialog '0433af7878e3a8067a40f896382cc3a6@79.80.x.x' in 32000 ms (Method: INVITE)<br>[Jul 21 11:11:03] DEBUG[23814]: chan_sip.c:5695 reqprep: Strict routing enforced for session 0433af7878e3a8067a40f896382cc3a6@79.80.x.x<br>
set_destination: Parsing <<a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a>> for address/port to send to<br>set_destination: set destination to 116.18.35.235, port 28614<br>Reliably Transmitting (NAT) to <a href="http://116.18.35.235:28614/">116.18.35.235:28614</a>:<br>
BYE <a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a> SIP/2.0<br>Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK36df65b5;rport<br>From: "pepsi coke" <<a href="sip:12345678901@79.80.x.x:5678">sip:12345678901@79.80.x.x:5678</a>>;tag=as42ec768c<br>
To: <<a href="http://sip:adf@116.18.35.235:28614/">sip:adf@116.18.35.235:28614</a>>;tag=d54e632c<br>Call-ID: 0433af7878e3a8067a40f896382cc3a6@79.80.x.x<br>CSeq: 103 BYE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>
Content-Length: 0<br><br><br><br><br>Nasir Javaid<br>
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