<div dir="ltr"><br><div class="gmail_extra">Hi Alberto,<br><br>Here's the output that you asked for: </div><div class="gmail_extra"><br></div><div class="gmail_extra"><div class="gmail_extra">Global Settings:</div><div class="gmail_extra">
----------------</div><div class="gmail_extra"> UDP Bindaddress: <a href="http://0.0.0.0:5060">0.0.0.0:5060</a></div><div class="gmail_extra"> TCP SIP Bindaddress: Disabled</div><div class="gmail_extra"> TLS SIP Bindaddress: Disabled</div>
<div class="gmail_extra"> Videosupport: Yes</div><div class="gmail_extra"> Textsupport: No</div><div class="gmail_extra"> Ignore SDP sess. ver.: No</div><div class="gmail_extra"> AutoCreate Peer: Off</div>
<div class="gmail_extra"> Match Auth Username: No</div><div class="gmail_extra"> Allow unknown access: Yes</div><div class="gmail_extra"> Allow subscriptions: Yes</div><div class="gmail_extra"> Allow overlap dialing: No</div>
<div class="gmail_extra"> Allow promisc. redir: No</div><div class="gmail_extra"> Enable call counters: No</div><div class="gmail_extra"> SIP domain support: No</div><div class="gmail_extra"> Realm. auth: No</div>
<div class="gmail_extra"> Our auth realm asterisk</div><div class="gmail_extra"> Use domains as realms: No</div><div class="gmail_extra"> Call to non-local dom.: Yes</div><div class="gmail_extra"> URI user is phone no: No</div>
<div class="gmail_extra"> Always auth rejects: Yes</div><div class="gmail_extra"> Direct RTP setup: No</div><div class="gmail_extra"> User Agent: Asterisk PBX 11.3.0</div><div class="gmail_extra">
SDP Session Name: Asterisk PBX 11.3.0</div>
<div class="gmail_extra"> SDP Owner Name: root</div><div class="gmail_extra"> Reg. context: (not set)</div><div class="gmail_extra"> Regexten on Qualify: No</div><div class="gmail_extra"> Trust RPID: No</div>
<div class="gmail_extra"> Send RPID: No</div><div class="gmail_extra"> Legacy userfield parse: No</div><div class="gmail_extra"> Send Diversion: Yes</div><div class="gmail_extra"> Caller ID: asterisk</div>
<div class="gmail_extra"> From: Domain:</div><div class="gmail_extra"> Record SIP history: Off</div><div class="gmail_extra"> Call Events: Off</div><div class="gmail_extra"> Auth. Failure Events: Off</div>
<div class="gmail_extra"> T.38 support: No</div><div class="gmail_extra"> T.38 EC mode: Unknown</div><div class="gmail_extra"> T.38 MaxDtgrm: -1</div><div class="gmail_extra"> SIP realtime: Disabled</div>
<div class="gmail_extra"> Qualify Freq : 60000 ms</div><div class="gmail_extra"> Q.850 Reason header: No</div><div class="gmail_extra"> Store SIP_CAUSE: No</div><div class="gmail_extra"><br></div><div class="gmail_extra">
Network QoS Settings:</div><div class="gmail_extra">---------------------------</div><div class="gmail_extra"> IP ToS SIP: CS0</div><div class="gmail_extra"> IP ToS RTP audio: CS0</div><div class="gmail_extra">
IP ToS RTP video: CS0</div><div class="gmail_extra"> IP ToS RTP text: CS0</div><div class="gmail_extra"> 802.1p CoS SIP: 4</div><div class="gmail_extra"> 802.1p CoS RTP audio: 5</div><div class="gmail_extra">
802.1p CoS RTP video: 6</div><div class="gmail_extra"> 802.1p CoS RTP text: 5</div><div class="gmail_extra"> Jitterbuffer enabled: No</div><div class="gmail_extra"><br></div><div class="gmail_extra">Network Settings:</div>
<div class="gmail_extra">---------------------------</div><div class="gmail_extra"> SIP address remapping: Disabled, no localnet list</div><div class="gmail_extra"> Externhost: <none></div><div class="gmail_extra">
Externaddr: (null)</div><div class="gmail_extra"> Externrefresh: 10</div><div class="gmail_extra"><br></div><div class="gmail_extra">Global Signalling Settings:</div><div class="gmail_extra">---------------------------</div>
<div class="gmail_extra"> Codecs: (gsm|ulaw|alaw|h263|testlaw)</div><div class="gmail_extra"> Codec Order: none</div><div class="gmail_extra"> Relax DTMF: No</div><div class="gmail_extra">
RFC2833 Compensation: No</div><div class="gmail_extra"> Symmetric RTP: No</div><div class="gmail_extra"> Compact SIP headers: No</div><div class="gmail_extra"> RTP Keepalive: 0 (Disabled)</div>
<div class="gmail_extra"> RTP Timeout: 0 (Disabled)</div><div class="gmail_extra"> RTP Hold Timeout: 0 (Disabled)</div><div class="gmail_extra"> MWI NOTIFY mime type: application/simple-message-summary</div>
<div class="gmail_extra"> DNS SRV lookup: No</div><div class="gmail_extra"> Pedantic SIP support: Yes</div><div class="gmail_extra"> Reg. min duration 60 secs</div><div class="gmail_extra"> Reg. max duration: 3600 secs</div>
<div class="gmail_extra"> Reg. default duration: 120 secs</div><div class="gmail_extra"> Sub. min duration 60 secs</div><div class="gmail_extra"> Sub. max duration: 3600 secs</div><div class="gmail_extra">
Outbound reg. timeout: 20 secs</div>
<div class="gmail_extra"> Outbound reg. attempts: 0</div><div class="gmail_extra"> Notify ringing state: Yes</div><div class="gmail_extra"> Include CID: No</div><div class="gmail_extra"> Notify hold state: No</div>
<div class="gmail_extra"> SIP Transfer mode: open</div><div class="gmail_extra"> Max Call Bitrate: 384 kbps</div><div class="gmail_extra"> Auto-Framing: No</div><div class="gmail_extra"> Outb. proxy: <not set></div>
<div class="gmail_extra"> Session Timers: Accept</div><div class="gmail_extra"> Session Refresher: uas</div><div class="gmail_extra"> Session Expires: 1800 secs</div><div class="gmail_extra"> Session Min-SE: 90 secs</div>
<div class="gmail_extra"> Timer T1: 500</div><div class="gmail_extra"> Timer T1 minimum: 100</div><div class="gmail_extra"> Timer B: 32000</div><div class="gmail_extra"> No premature media: Yes</div>
<div class="gmail_extra"> Max forwards: 70</div><div class="gmail_extra"><br></div><div class="gmail_extra">Default Settings:</div><div class="gmail_extra">-----------------</div><div class="gmail_extra"> Allowed transports: UDP</div>
<div class="gmail_extra"> Outbound transport: UDP</div><div class="gmail_extra"> Context: default</div><div class="gmail_extra"> Record on feature: automon</div><div class="gmail_extra"> Record off feature: automon</div>
<div class="gmail_extra"> Force rport: Auto (No)</div><div class="gmail_extra"> DTMF: rfc2833</div><div class="gmail_extra"> Qualify: 0</div><div class="gmail_extra"> Keepalive: 0</div>
<div class="gmail_extra"> Use ClientCode: No</div><div class="gmail_extra"> Progress inband: Never</div><div class="gmail_extra"> Language:</div><div class="gmail_extra"> Tone zone: <Not set></div>
<div class="gmail_extra"> MOH Interpret: default</div><div class="gmail_extra"> MOH Suggest:</div><div class="gmail_extra"> Voice Mail Extension: asterisk</div><div class="gmail_extra"><br></div><div class="gmail_extra">
<br></div><div class="gmail_extra">----</div><div><br></div><div class="gmail_quote">On Mon, Apr 29, 2013 at 7:12 PM, Alberto Llamas <span dir="ltr"><<a href="mailto:albertollamaso@gmail.com" target="_blank">albertollamaso@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
Cual es el resultao de:<br>
<br>
<b>sip show settings</b><br>
<br></div></blockquote><div><br></div><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div bgcolor="#FFFFFF" text="#000000">
<div>El 29/04/13 9:10, Moosa Khalid
escribió:<br>
</div>
<blockquote type="cite"><div><div class="h5">
<div dir="ltr">I'm trying to make a video call between two SIP
peers registered on a single asterisk box. Following is the
config of my sip peers
<div><br>
</div>
<div>
<div>[107]</div>
<div>defaultuser=107</div>
<div>secret=107</div>
<div>type=friend</div>
<div>host=dynamic</div>
<div>context=default</div>
<div>canreinvite=yes</div>
<div>videosupport=yes</div>
<div>dtmfmode=rfc2833</div>
<div>qualify=yes</div>
<div>disallow=all</div>
<div>allow=ulaw</div>
<div>
allow=alaw</div>
<div>allow=gsm</div>
<div>allow=h263</div>
<div><br>
</div>
<div>[701]<br>
</div>
<div>defaultuser=701</div>
<div>secret=701</div>
<div>type=friend</div>
<div>host=dynamic</div>
<div>context=default</div>
<div>videosupport=yes</div>
<div>qualify=yes</div>
<div>canreinvite=yes</div>
<div>dtmfmode=rfc2833</div>
<div>disallow=all</div>
<div>allow=ulaw</div>
<div>allow=alaw</div>
<div>allow=gsm</div>
<div>allow=h263</div>
</div>
<div><br>
</div>
<div>I've also enabled videosupport in the global sip
settings.</div>
<div>I'm using the softphone Xlite 4.5 which supports
h263 codec on for both clients. <b>Asterisk version is 11.3.0
LTS</b>. Clients registered are on same network. Asterisk
shows the following output on console on a call b/w peers.
Audio is fine but of course no video thanks to the following
warning. </div>
<div><br>
</div>
<div>
<div><b> == Using SIP VIDEO CoS mark 6</b></div>
<div><b> == Using SIP RTP CoS mark 5</b></div>
<div><b> -- Executing [96107@default:1]
Dial("SIP/701-00000014", "SIP/107,20,rt") in new stack</b></div>
<div><b> == Using SIP VIDEO CoS mark 6</b></div>
<div><b> == Using SIP RTP CoS mark 5</b></div>
<div><b> -- Called SIP/107</b></div>
<div><b> -- SIP/107-00000015 is ringing</b></div>
<div><b> -- SIP/107-00000015 is ringing</b></div>
<div><b>[Apr 29 18:35:12] WARNING[11363][C-0000000d]:
chan_sip.c:10141 process_sdp: Ignoring video stream offer
because port number is zero</b></div>
<div><b><br>
</b></div>
<div>This happens regardless of which sip peer
originates the call.</div>
<div><br>
</div>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
</div></div><span class=""><font color="#888888"><pre>--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
<a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a></pre>
</font></span></blockquote><span class=""><font color="#888888">
<br>
<pre cols="72">--
Alberto Llamas
Ingeniero de Telecomunicaciones
Digium Certified Asterisk Administrator
Digium Certified Asterisk Professional
Linux Administrator</pre>
</font></span></div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-video mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br></blockquote></div><br></div></div>