Hello Grygoriy, I am forwarding UDP ports from 10000 to 10100. That only means that I am forwarding 101 ports. Please take note also that when I tried calling the GTalk ID, the Asterisk box was idle or there was no any other on-going calls.
Regards, GNUbie On Thu, Jan 29, 2009 at 4:15 PM, Grygoriy Dobrovolskyy <[email protected]> wrote: > How many ports have you forwarded for the * ? (in rtp.conf) > If a limited amount (50-100), try to forward more. > > 2009/1/29 GNUbie <[email protected]> >> >> Hello all, >> >> In addition to my previous e-mail, below is a more verbosed messages I >> got on my Asterisk shell when calling from another GTalk User ID to >> the Asterisk-1.4.21.2 box: >> >> pbx*CLI> >> JABBER: gtalk INCOMING: <iq to="[email protected]/asteriskE2D976CC" >> type="set" id="49" from="[email protected]/Talk.v1041B79926B"><session >> type="initiate" id="3756468934" >> initiator="[email protected]/Talk.v1041B79926B" >> xmlns="http://www.google.com/session";><description xml:lang="en" >> xmlns="http://www.google.com/session/phone";><payload-type id="103" >> name="ISAC" clockrate="16000"/><payload-type id="97" name="IPCMWB" >> clockrate="16000" bitrate="80000"/><payload-type id="99" name="speex" >> clockrate="16000" bitrate="22000"/><payload-type id="4" name="G723" >> clockrate="8000" bitrate="6300"/><payload-type id="98" name="speex" >> clockrate="8000" bitrate="11000"/><payload-type id="100" name="EG711U" >> clockrate="8000" bitrate="64000"/><payload-type id="101" name="EG711A" >> clockrate="8000" bitrate="64000"/><payload-type id="0" name="PCMU" >> clockrate="8000" bitrate="64000"/><payload-type id="8" name="PCMA" >> clockrate="8000" bitrate="64000"/><payload-type id="13" name="CN" >> clockrate="8000"/><payload-type id="102" name="iLBC" clockrate=" >> >> JABBER: gtalk INCOMING: 8000" bitrate="13300"/><payload-type id="106" >> name="telephone-event" clockrate="8000"/></description><transport >> xmlns="http://www.google.com/transport/p2p"/></session></iq> >> [Jan 29 11:18:24] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: >> Unexpected bind error: Cannot assign requested address >> [Jan 29 11:18:24] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of >> RTP sessions? >> [Jan 29 11:18:24] WARNING[1303]: chan_gtalk.c:1175 gtalk_newcall: >> Unable to allocate gtalk structure! >> pbx*CLI> >> JABBER: gtalk INCOMING: <iq to="[email protected]/asteriskE2D976CC" >> type="set" id="51" from="[email protected]/Talk.v1041B79926B"><session >> type="transport-info" id="3756468934" >> initiator="[email protected]/Talk.v1041B79926B" >> xmlns="http://www.google.com/session";><transport >> xmlns="http://www.google.com/transport/p2p";><candidate name="rtp" >> address="10.20.1.151" port="1587" preference="1" >> username="RrBBqm7MeJW2zTgi" protocol="udp" generation="0" >> password="OjLNI9dyFLqqBi/Y" type="local" >> network="0"/></transport></session></iq> >> pbx*CLI> >> JABBER: gtalk INCOMING: <iq to="[email protected]/asteriskE2D976CC" >> type="set" id="52" from="[email protected]/Talk.v1041B79926B"><session >> type="transport-info" id="3756468934" >> initiator="[email protected]/Talk.v1041B79926B" >> xmlns="http://www.google.com/session";><transport >> xmlns="http://www.google.com/transport/p2p";><candidate name="rtp" >> address="219.74.65.168" port="1588" preference="0.9" >> username="sHhE4y2GwRBmLQUB" protocol="udp" generation="0" >> password="BYAvdVRiU94RVOJW" type="stun" >> network="0"/></transport></session></iq> >> pbx*CLI> >> JABBER: gtalk INCOMING: <iq to="[email protected]/asteriskE2D976CC" >> type="set" id="54" from="[email protected]/Talk.v1041B79926B"><session >> type="terminate" id="3756468934" >> initiator="[email protected]/Talk.v1041B79926B" >> xmlns="http://www.google.com/session"/></iq> >> [Jan 29 11:18:40] NOTICE[1303]: chan_gtalk.c:783 gtalk_hangup_farend: >> Whoa, didn't find call! >> >> JABBER: gtalk OUTGOING: <iq type='result' >> from='[email protected]/asteriskE2D976CC' >> to='[email protected]/Talk.v1041B79926B' id='54'/> >> >> JABBER: gtalk INCOMING: >> pbx*CLI> >> JABBER: gtalk INCOMING: <presence >> from="[email protected]/Talk.v1041B79926B" >> to="[email protected]"><priority>24</priority><c >> node="http://www.google.com/xmpp/client/caps"; ver="1.0.0.104" >> ext="share-v1 voice-v1" xmlns="http://jabber.org/protocol/caps"/><x >> stamp="20090129T03:17:52" xmlns="jabber:x:delay"/><status/><x >> >> xmlns="vcard-temp:x:update"><photo>8939f8f8ed0a9cd794e9e3c7065c2cc80fa9dbf0</photo></x></presence> >> pbx*CLI> >> JABBER: gtalk INCOMING: <presence >> from="[email protected]/Talk.v1041B79926B" type="unavailable" >> to="[email protected]"/> >> pbx*CLI> >> >> Thank you in advance. >> >> Regards, >> >> Marvin >> >> >> On Thu, Jan 29, 2009 at 10:47 AM, GNUbie <[email protected]> wrote: >> > Hello all, >> > >> > It used to work on calling my GTalk ID from another GTalk user. But >> > now that I tried calling it again, the caller hears only a ringtone >> > and disconnected after a few rings. The messages on my >> > Asterisk-1.4.21.2 are the following: >> > >> > [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: >> > Unexpected bind error: Cannot assign requested address >> > [Jan 29 10:37:51] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of >> > RTP sessions? >> > [Jan 29 10:37:51] WARNING[1303]: chan_gtalk.c:1175 gtalk_newcall: >> > Unable to allocate gtalk structure! >> > [Jan 29 10:38:06] NOTICE[1303]: chan_gtalk.c:783 gtalk_hangup_farend: >> > Whoa, didn't find call! >> > >> > Any idea? >> > >> > Thank you in advance. >> > >> > Regards, >> > >> > GNUbie >> > >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> 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 -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
