if you were on trunk that line of code would be gone. you really can't do development on 1.0.4 its 6 months old and it will cause you more trouble than you think when you eventually upgrade if you do not do it soon.
2009/10/23 Georgiewskiy Yuriy <bottle...@icf.org.ru> > On 2009-10-23 16:57 +0200, Tihomir Culjaga wrote > freeswitch-us...@lists.fre...: > > i have question to developers about one proce in fs > > src/switch_ivr_originate.c > > static switch_status_t > originate_on_consume_media_transmit(switch_core_session_t *session) > { > switch_channel_t *channel = switch_core_session_get_channel(session); > > if (!switch_channel_test_flag(channel, CF_PROXY_MODE)) { > while (switch_channel_get_state(channel) == CS_CONSUME_MEDIA > && !switch_channel_test_flag(chann > if (!switch_channel_media_ready(channel)) { > switch_yield(10000); > } else { > switch_ivr_sleep(session, 10, SWITCH_FALSE, > NULL); > } > } > } > > switch_channel_clear_state_handler(channel, > &originate_state_handlers); > > return SWITCH_STATUS_FALSE; > } > > what exacly it do? > > call scheme like this sip->fs->h323->gk->h323(on same fs)->fs(same too) and > there i have no audio issues. > if bridge connect while it sleep i have audio, if it not sleep while bridge > connect i have no audio. > > TC>a solution to H323 endpoint => FS => SIP user no audio issue > TC> > TC>is to disable a wait for tx Audio ... for case > TC>SWITCH_MESSAGE_INDICATE_ANSWER:{ > TC> > TC>//m_txAudioOpened.Wait(); > TC> > TC> > TC> case SWITCH_MESSAGE_INDICATE_ANSWER:{ > TC> > TC> switch_log_printf(SWITCH_CHANNEL_LOG, > TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event\n"); > TC> > TC> if (switch_channel_test_flag(channel, > CF_OUTBOUND)) > TC>{ > TC> > TC> switch_log_printf(SWITCH_CHANNEL_LOG, > TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event - CF_OUTBOUND > TC>\n"); > TC> return SWITCH_STATUS_FALSE; > TC> } > TC> AnsweringCall(H323Connection::AnswerCallNow); > TC> > TC> switch_log_printf(SWITCH_CHANNEL_LOG, > TC>SWITCH_LOG_CONSOLE, "ANSWER: suppose the call is Answered Now\n"); > TC> PTRACE(4, "mod_h323\tMedia started on connection > " > TC><< *this); > TC> > TC> // test > TC> //switch_channel_mark_answered(m_fsChannel); > TC> > TC> m_rxAudioOpened.Wait(); > TC> switch_log_printf(SWITCH_CHANNEL_LOG, > TC>SWITCH_LOG_CONSOLE, "ANSWER: wait for m_rxAudioOpened\n"); > TC> //m_txAudioOpened.Wait(); > TC> switch_log_printf(SWITCH_CHANNEL_LOG, > TC>SWITCH_LOG_CONSOLE, "ANSWER: we disable wait for m_txAudioOpened\n"); > TC> > TC> switch_log_printf(SWITCH_CHANNEL_LOG, > TC>SWITCH_LOG_CONSOLE, "ANSWER: were waiting for rx/tx AudioOpen\n"); > TC> > TC> if (!switch_channel_test_flag(m_fsChannel, > TC>CF_EARLY_MEDIA)) { > TC> > TC> switch_log_printf(SWITCH_CHANNEL_LOG, > TC>SWITCH_LOG_CONSOLE, "ANSWER: we have early media\n"); > TC> > TC> PTRACE(4, > TC>"mod_h323\t-------------------->switch_channel_mark_answered(m_fsChannel) > " > TC><< *this); > TC> > switch_channel_mark_answered(m_fsChannel); > TC> switch_log_printf(SWITCH_CHANNEL_LOG, > TC>SWITCH_LOG_CONSOLE, "ANSWER: answered in early Media\n"); > TC> } > TC> break; > TC> } > TC> > TC> > TC>Now, I'm able to both originate and terminate cals with 2-way audio... > TC>the signaling looks correct... > TC> > TC> > TC> > TC>outgoing: > TC> > TC>1369.425046 10.4.62.7 -> 10.4.62.89 SIP/SDP Request: INVITE > TC>sip:1...@10.4.62.89 <sip%3a1...@10.4.62.89> > <sip%3a1...@10.4.62.89<sip%253a1...@10.4.62.89>>;transport=udp, > with session > TC>description > TC>1369.426255 10.4.62.7 -> 10.4.62.31 H.225.0 CS: alerting > TC>1369.435950 10.4.62.89 -> 10.4.62.7 SIP Status: 100 Trying > TC>1369.449065 10.4.62.89 -> 10.4.62.7 SIP Status: 180 Ringing > TC>1369.605109 10.4.62.7 -> 10.4.62.31 H.225.0 CS: progress > TC>OpenLogicalChannel > TC>1369.609788 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility > TC>terminalCapabilitySet > TC>1369.610489 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility > TC>masterSlaveDetermination > TC>1369.619071 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty > TC>terminalCapabilitySet > TC>1369.620349 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty > TC>terminalCapabilitySetAck > TC>1369.623215 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility > TC>terminalCapabilitySetAck > TC>1369.625591 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty > TC>masterSlaveDeterminationAck > TC>1369.628174 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility > TC>masterSlaveDeterminationAck > TC>1370.966958 10.4.62.89 -> 10.4.62.7 SIP/SDP Status: 200 OK, with > TC>session description > TC>1370.967431 10.4.62.7 -> 10.4.62.89 SIP Request: ACK > TC>sip:1...@10.4.62.89 <sip%3a1...@10.4.62.89> > <sip%3a1...@10.4.62.89<sip%253a1...@10.4.62.89> > >;transport=udp > TC>1370.975172 10.4.62.7 -> 10.4.62.31 H.225.0 CS: connect > TC>1372.354383 10.4.62.89 -> 10.4.62.7 SIP Request: BYE > TC>sip:mod_so...@10.4.62.7:5060 > TC>1372.355147 10.4.62.7 -> 10.4.62.89 SIP Status: 200 OK > TC>1372.392904 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: > releaseComplete > TC>endSessionCommand > TC>1372.397302 10.4.62.31 -> 10.4.62.7 H.225.0 CS: releaseComplete > TC> > TC> > TC>incoming: > TC> > TC> > TC>1502.817154 10.4.62.31 -> 10.4.62.7 H.225.0 CS: setup > TC>OpenLogicalChannel > TC>1502.833732 10.4.62.7 -> 10.4.62.31 H.225.0 CS: callProceeding > TC>1502.850909 10.4.62.7 -> 10.4.62.89 SIP/SDP Request: INVITE > TC>sip:1...@10.4.62.89 <sip%3a1...@10.4.62.89> > <sip%3a1...@10.4.62.89<sip%253a1...@10.4.62.89>>;transport=udp, > with session > TC>description > TC>1502.851758 10.4.62.7 -> 10.4.62.31 H.225.0 CS: alerting > TC>1502.861828 10.4.62.89 -> 10.4.62.7 SIP Status: 100 Trying > TC>1502.875127 10.4.62.89 -> 10.4.62.7 SIP Status: 180 Ringing > TC>1503.033258 10.4.62.7 -> 10.4.62.31 H.225.0 CS: progress > TC>OpenLogicalChannel > TC>1503.037908 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility > TC>terminalCapabilitySet > TC>1503.038608 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility > TC>masterSlaveDetermination > TC>1503.050154 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty > TC>terminalCapabilitySet > TC>1503.051381 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty > TC>terminalCapabilitySetAck > TC>1503.054297 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility > TC>terminalCapabilitySetAck > TC>1503.054917 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty > TC>masterSlaveDeterminationAck > TC>1503.057933 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility > TC>masterSlaveDeterminationAck > TC>1505.485493 10.4.62.89 -> 10.4.62.7 SIP/SDP Status: 200 OK, with > TC>session description > TC>1505.486018 10.4.62.7 -> 10.4.62.89 SIP Request: ACK > TC>sip:1...@10.4.62.89 <sip%3a1...@10.4.62.89> > <sip%3a1...@10.4.62.89<sip%253a1...@10.4.62.89> > >;transport=udp > TC>1505.493611 10.4.62.7 -> 10.4.62.31 H.225.0 CS: connect > TC>1509.565959 10.4.62.89 -> 10.4.62.7 SIP Request: BYE > TC>sip:mod_so...@10.4.62.7:5060 > TC>1509.566722 10.4.62.7 -> 10.4.62.89 SIP Status: 200 OK > TC>1509.577435 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: > releaseComplete > TC>endSessionCommand > TC>1509.582066 10.4.62.31 -> 10.4.62.7 H.225.0 CS: releaseComplete > TC> > TC> > TC> > TC>... i still need to check the CDRs as well but here we are :) > TC> > > can you send a diff? in you call scheme call from h323 endpoint to fs is > not have RAS?, > because i don't have no audio issues in transit from h323 to sip, but my > calls a going > thorough GK and fs is regitered on them, my call scheme is > h323ep-RAS->GK-RAS->fs. > > C уважением With Best Regards > Георгиевский Юрий. Georgiewskiy Yuriy > +7 4872 711666 +7 4872 711666 > факс +7 4872 711143 fax +7 4872 711143 > Компания ООО "Ай Ти Сервис" IT Service Ltd > http://nkoort.ru http://nkoort.ru > JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru > YG129-RIPE YG129-RIPE > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com <msn%3aanthony_miness...@hotmail.com> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org> iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org> pstn:213-799-1400
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