<div dir="ltr"><div class="gmail_default" style="color:rgb(102,0,0)">To sort out RTP problems, I would recommend:</div><div class="gmail_default" style="color:rgb(102,0,0)"><br></div><div class="gmail_default" style="color:rgb(102,0,0)">1) on all endpoints use codec of allow=!all,ulaw -- this is or should be supported by all endpoints and eliminates any issues of mismatch, translation, etc., and can be adjusted later once everything is working</div><div class="gmail_default" style="color:rgb(102,0,0)"><br></div><div class="gmail_default" style="color:rgb(102,0,0)">2) add an Echo() application to your dialplan so you can call it and check RTP to and from Asterisk</div><div class="gmail_default" style="color:rgb(102,0,0)"><br></div><div class="gmail_default" style="color:rgb(102,0,0)">3) start with direct_media=no to run all the RTP through Asterisk first</div><div class="gmail_default" style="color:rgb(102,0,0)"><br></div><div class="gmail_default" style="color:rgb(102,0,0)">4) packet capture at/on the asterisk server, as well as at endpoints if need be, to identfy if and where RTP streams are being sent and received.</div><div class="gmail_default" style="color:rgb(102,0,0)"><br></div><div class="gmail_default" style="color:rgb(102,0,0)">The goal being to get two way audio calls up through Asterisk, and then change one thing at a time towards your desired configuration and retest.</div><div class="gmail_default" style="color:rgb(102,0,0)"><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Jan 8, 2015 at 7:03 PM, Sonny Rajagopalan <span dir="ltr"><<a href="mailto:sonny.rajagopalan@gmail.com" target="_blank">sonny.rajagopalan@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Well, I thought it worked, but it actually doesn't--I am able to get the caller pick up the phone, but for some reason, I cannot hear anything on either side no matter who does the calling. Again, my two SIP phones are on the local <a href="http://192.168.1.0/24" target="_blank">192.168.1.0/24</a> network (do not go over the Internet) and the Asterisk server is located in the same network (not accessed over the Internet). Any help is appreciated.<div><br></div><div>Does the fact that Asterisk is running on a VirtualBox VM on the same machine as one of the SIP phones matter? I am able to access the ARI REST interface of the Asterisk server quite fine on the host machine.</div><div><br></div><div>I suspect it has to do with RTP not being set up, but all the codec support is there. Here's a log for the SIP request from <a href="http://192.168.1.50" target="_blank">192.168.1.50</a>:</div><div><br></div><font face="monospace, monospace" size="1" color="#ff0000"><--- Received SIP request (1229 bytes) from UDP:<a href="http://192.168.1.50:64009" target="_blank">192.168.1.50:64009</a> ---><br>INVITE <a href="mailto:sip%3A6002@192.168.1.139" target="_blank">sip:6002@192.168.1.139</a>;transport=UDP SIP/2.0<br>Via: SIP/2.0/UDP 146.115.163.234:64009;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z-<br>Max-Forwards: 70<br>Contact: <sip:demo-alice@146.115.163.234:64009;transport=UDP><br>To: <<a href="mailto:sip%3A6002@192.168.1.139" target="_blank">sip:6002@192.168.1.139</a>;transport=UDP><br>From: <<a href="mailto:sip%3Ademo-alice@192.168.1.139" target="_blank">sip:demo-alice@192.168.1.139</a>;transport=UDP>;tag=b661670b<br>Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE.<br>CSeq: 2 INVITE<br>Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE<br>Content-Type: application/sdp<br>Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri<br>User-Agent: Z 3.3.21933 r21903<br><br>Authorization: Digest username="demo-alice",realm="asterisk",nonce="[removed]",uri="<a href="mailto:sip%3A6002@192.168.1.139" target="_blank">sip:6002@192.168.1.139</a>;transport=UDP",response="[removed]",cnonce="[removed]",nc=00000001,qop=auth,algorithm=md5,opaque="[removed]"<br><br>Allow-Events: presence, kpml<br>Content-Length: 245<br><br><br>v=0<br>o=Z 0 0 IN IP4 146.115.163.234<br>s=Z<br>c=IN IP4 146.115.163.234<br>t=0 0<br>m=audio 8000 RTP/AVP 0 3 110 8 98 101<br>a=rtpmap:110 speex/8000<br>a=rtpmap:98 iLBC/8000<br>a=fmtp:98 mode=20<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=sendrecv<br><br><br><--- Transmitting SIP response (319 bytes) to UDP:<a href="http://192.168.1.50:64009" target="_blank">192.168.1.50:64009</a> ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 146.115.163.234:64009;rport=64009;received=192.168.1.50;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z-<br>Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE.<br>From: <<a href="mailto:sip%3Ademo-alice@192.168.1.139" target="_blank">sip:demo-alice@192.168.1.139</a>>;tag=b661670b<br>To: <<a href="mailto:sip%3A6002@192.168.1.139" target="_blank">sip:6002@192.168.1.139</a>><br>CSeq: 2 INVITE<br>Content-Length: 0</font><div><br></div><div>Any help is appreciated. A topology is shown below in ASCII.</div><div><font face="monospace, monospace" size="1"><br></font></div><div><font face="monospace, monospace" size="1"><br></font></div><div><font face="monospace, monospace" size="1"> < ( Big bad Internet ) ></font></div><div><font face="monospace, monospace" size="1"><br></font></div><div><font face="monospace, monospace" size="1"> GW/NAPT/Router</font></div><div><font face="monospace, monospace" size="1"> |</font></div><div><font face="monospace, monospace" size="1"> ----------------------------------------------------------</font></div><div><font face="monospace, monospace" size="1"> / \ </font></div><div><font face="monospace, monospace" size="1"> | |</font></div><div><font face="monospace, monospace" size="1"> Host A Host B</font></div><div><font face="monospace, monospace" size="1">----------------- -----------------</font></div><div><font face="monospace, monospace" size="1">| Alice | | Bob |</font></div><div><font face="monospace, monospace" size="1">| 192.168.1.50 | | 192.168.1.149 |</font></div><div><font face="monospace, monospace" size="1">|---------------| |---------------|</font></div><div><font face="monospace, monospace" size="1">| Asterisk sr |</font></div><div><font face="monospace, monospace" size="1">| (VM) |</font></div><div><font face="monospace, monospace" size="1">| 192.168.1.239 |</font></div><div><font face="monospace, monospace" size="1">|---------------|</font></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan <span dir="ltr"><<a href="mailto:sonny.rajagopalan@gmail.com" target="_blank">sonny.rajagopalan@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Thank you for your note, Scott. </div><div><br></div>I set <font face="monospace, monospace" color="#ff0000">rewrite_contact=yes</font> for both contacts, and I also had to do <font face="monospace, monospace" color="#ff0000">remove_existing=yes</font> because I had to remove the existing contact information (<font face="monospace, monospace" color="#ff0000">max_contacts = 1</font> was preventing new contact information) using <font face="monospace, monospace" color="#ff0000">pjsip qualify demo-alice</font> etc., after which the right IP addresses showed in <font face="monospace, monospace" color="#ff0000">pjsip show endpoints</font>. Anyway, it works as expected now, I think. My <font face="monospace, monospace" color="#ff0000">pjsip.conf</font> is now<div><br></div><blockquote style="margin:0 0 0 40px;border:none;padding:0px"><font color="#ff0000"><font size="1" face="monospace, monospace">[transport-udp]</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">type=transport</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">protocol=udp</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">bind=0.0.0.0</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">local_net=<a href="http://192.168.1.0/24" target="_blank">192.168.1.0/24</a></font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">;Templates for the necessary config sections</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">[endpoint_internal](!)</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">type=endpoint</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">context=from-internal</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">disallow=all</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">allow=ulaw</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">[auth_userpass](!)</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">type=auth</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">auth_type=userpass</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">[aor_dynamic](!)</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">type=aor</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">max_contacts=1</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">remove_existing=yes</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">;Definitions for our phones, using the templates above</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">[demo-alice](endpoint_internal)</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">auth=demo-alice</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">aors=demo-alice</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">mailboxes=box_a</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">rewrite_contact=yes</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">[demo-alice](auth_userpass)</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">password=demo-alice ; put a strong, unique password here instead</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">username=demo-alice</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">[demo-alice](aor_dynamic)</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">[demo-bob](endpoint_internal)</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">auth=demo-bob</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">aors=demo-bob</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">mailboxes=box_b</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">rewrite_contact=yes</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">[demo-bob](auth_userpass)</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">password=demo-bob ; put a strong, unique password here instead</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">username=demo-bob</font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace"><br></font><font size="1" face="monospace, monospace">[demo-bob](aor_dynamic)</font></font></blockquote><div class="gmail_extra"><br></div><div class="gmail_extra">Thank you for your help!</div><div><div><div class="gmail_extra"><br></div><div class="gmail_extra"><div class="gmail_quote">On Thu, Jan 8, 2015 at 11:48 AM, Scott Griepentrog <span dir="ltr"><<a href="mailto:sgriepentrog@digium.com" target="_blank">sgriepentrog@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div style="color:#660000">It would appear that you have the Asterisk server on a public IP address, your two endpoints are behind a NAT, and you have rewrite_contact enabled in pjsip.conf.</div><div style="color:#660000"><br></div><div style="color:#660000">In which case, what you are seeing is correct. In order to be able to send a call to an extension where it is behind NAT, Asterisk must update the contact to have the current IP and port that the phone registered via (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state for).</div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan <span dir="ltr"><<a href="mailto:sonny.rajagopalan@gmail.com" target="_blank">sonny.rajagopalan@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><div>I am following the instructions in <a href="https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality</a> and I am trying to make a call from extension Alice (6001) to extension for Bob (6002). When I make the call, I can hear the ringing on Alice's phone (caller), but Bob's phone (callee) doesn't ring, or show a call coming in from Alice. My setup and environment is as follows: Alice, Bob and Asterisk all in the same <a href="http://192.168.1.0/24" target="_blank">192.168.1.0/24</a> network, and they are able to register to the Asterisk server running 13.1.0/PJSIP. The rest of the configuration is the same as the aforementioned wiki page, but is shown here for clarity:</div><div><br></div><blockquote style="margin:0 0 0 40px;border:none;padding:0px"><font color="#ff0000" size="1" face="monospace, monospace">root@asterisk13FFP:/var/log/asterisk# more /etc/asterisk/extensions.conf</font><font color="#ff0000" size="1" face="monospace, monospace"><br></font><font color="#ff0000" size="1" face="monospace, monospace">[from-internal]</font><font color="#ff0000" size="1" face="monospace, monospace"><br></font><font color="#ff0000" size="1" face="monospace, monospace">exten=>6001,1,Dial(PJSIP/demo-alice)</font><font color="#ff0000" size="1" face="monospace, monospace"><br></font><font color="#ff0000" size="1" face="monospace, monospace">exten=>6002,1,Dial(PJSIP/demo-bob)</font><font color="#ff0000" size="1" face="monospace, monospace"><br></font><font color="#ff0000" size="1" face="monospace, monospace">exten=>6003,1,Answer()</font><font color="#ff0000" size="1" face="monospace, monospace"><br></font><font color="#ff0000" size="1" face="monospace, monospace">same =>6003,n,Playback(hello-world)</font><font color="#ff0000" size="1" face="monospace, monospace"><br></font><font color="#ff0000" size="1" face="monospace, monospace">same =>6003,n,Hangup() </font></blockquote><div><br></div><div>What I do observe is that I when I request the output of <font face="monospace, monospace" color="#ff0000">pjsip show endpoints</font>, I get Contact information for the two SIP peers that have registered different from their actual IP addresses. I suspect this has something to do with their calls being routed elsewhere. If my assumption is correct--how do I fix this? Alice should be at 192.168.1.50 and Bob should be at 192.168.1.149, instead, they (both) show IP address 146.115.163.234. Any help is deeply appreciated. Thanks.</div><div><br></div><blockquote style="margin:0px 0px 0px 40px;border:none;padding:0px"><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000">asterisk13FFP*CLI> pjsip show endpoints</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"><br></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> I/OAuth: <AuthId/UserName...........................................................></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Aor: <Aor............................................> <MaxContact></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Contact: <Aor/ContactUri...............................> <Status....> <RTT(ms)..></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Identify: <Identify/Endpoint.........................................................></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Match: <ip/cidr.........................></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Channel: <ChannelId......................................> <State.....> <Time(sec)></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> =========================================================================================</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"><br></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Endpoint: demo-alice Unavailable 0 of inf</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> InAuth: demo-alice/demo-alice</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Aor: demo-alice 1</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Contact: demo-alice/sip:demo-alice@<b>146.115.163.234</b>:38519 Unknown nan</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"><br></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Endpoint: demo-bob Not in use 0 of inf</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> InAuth: demo-bob/demo-bob</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Aor: demo-bob 1</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Contact: demo-bob/sip:demo-bob@<b>146.115.163.234</b>:38321;tra Unknown nan</font></div></blockquote></div>
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