And what ports have you set in rtp.conf ? i suppose they are the same ? Try to search if there are no spercial jabber ports to open.
2009/1/29 GNUbie <[email protected]> > 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 >
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