On 2009-10-23 10:37 -0500, Anthony Minessale wrote freeswitch-users@lists.f...: i have no way to install trunk at this time, i will go out of hospital about one week later, after this i will can try it on trunk. AM>if you were on trunk that line of code would be gone. AM>you really can't do development on 1.0.4 its 6 months old and it will cause AM>you more trouble than you think when you eventually upgrade if you do not do AM>it soon. AM> AM> AM>2009/10/23 Georgiewskiy Yuriy <bottleman@icf.org.ru> AM> AM>> On 2009-10-23 16:57 +0200, Tihomir Culjaga wrote AM>> freeswitch-users@lists.fre...: AM>> AM>> i have question to developers about one proce in fs AM>> AM>> src/switch_ivr_originate.c AM>> AM>> static switch_status_t AM>> originate_on_consume_media_transmit(switch_core_session_t *session) AM>> { AM>> switch_channel_t *channel = switch_core_session_get_channel(session); AM>> AM>> if (!switch_channel_test_flag(channel, CF_PROXY_MODE)) { AM>> while (switch_channel_get_state(channel) == CS_CONSUME_MEDIA AM>> && !switch_channel_test_flag(chann AM>> if (!switch_channel_media_ready(channel)) { AM>> switch_yield(10000); AM>> } else { AM>> switch_ivr_sleep(session, 10, SWITCH_FALSE, AM>> NULL); AM>> } AM>> } AM>> } AM>> AM>> switch_channel_clear_state_handler(channel, AM>> &originate_state_handlers); AM>> AM>> return SWITCH_STATUS_FALSE; AM>> } AM>> AM>> what exacly it do? AM>> AM>> call scheme like this sip->fs->h323->gk->h323(on same fs)->fs(same too) and AM>> there i have no audio issues. AM>> if bridge connect while it sleep i have audio, if it not sleep while bridge AM>> connect i have no audio. AM>> AM>> TC>a solution to H323 endpoint => FS => SIP user no audio issue AM>> TC> AM>> TC>is to disable a wait for tx Audio ... for case AM>> TC>SWITCH_MESSAGE_INDICATE_ANSWER:{ AM>> TC> AM>> TC>//m_txAudioOpened.Wait(); AM>> TC> AM>> TC> AM>> TC> case SWITCH_MESSAGE_INDICATE_ANSWER:{ AM>> TC> AM>> TC> switch_log_printf(SWITCH_CHANNEL_LOG, AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event\n"); AM>> TC> AM>> TC> if (switch_channel_test_flag(channel, AM>> CF_OUTBOUND)) AM>> TC>{ AM>> TC> AM>> TC> switch_log_printf(SWITCH_CHANNEL_LOG, AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event - CF_OUTBOUND AM>> TC>\n"); AM>> TC> return SWITCH_STATUS_FALSE; AM>> TC> } AM>> TC> AnsweringCall(H323Connection::AnswerCallNow); AM>> TC> AM>> TC> switch_log_printf(SWITCH_CHANNEL_LOG, AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: suppose the call is Answered Now\n"); AM>> TC> PTRACE(4, "mod_h323\tMedia started on connection AM>> " AM>> TC><< *this); AM>> TC> AM>> TC> // test AM>> TC> //switch_channel_mark_answered(m_fsChannel); AM>> TC> AM>> TC> m_rxAudioOpened.Wait(); AM>> TC> switch_log_printf(SWITCH_CHANNEL_LOG, AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: wait for m_rxAudioOpened\n"); AM>> TC> //m_txAudioOpened.Wait(); AM>> TC> switch_log_printf(SWITCH_CHANNEL_LOG, AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: we disable wait for m_txAudioOpened\n"); AM>> TC> AM>> TC> switch_log_printf(SWITCH_CHANNEL_LOG, AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: were waiting for rx/tx AudioOpen\n"); AM>> TC> AM>> TC> if (!switch_channel_test_flag(m_fsChannel, AM>> TC>CF_EARLY_MEDIA)) { AM>> TC> AM>> TC> switch_log_printf(SWITCH_CHANNEL_LOG, AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: we have early media\n"); AM>> TC> AM>> TC> PTRACE(4, AM>> TC>"mod_h323\t-------------------->switch_channel_mark_answered(m_fsChannel) AM>> " AM>> TC><< *this); AM>> TC> AM>> switch_channel_mark_answered(m_fsChannel); AM>> TC> switch_log_printf(SWITCH_CHANNEL_LOG, AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: answered in early Media\n"); AM>> TC> } AM>> TC> break; AM>> TC> } AM>> TC> AM>> TC> AM>> TC>Now, I'm able to both originate and terminate cals with 2-way audio... AM>> TC>the signaling looks correct... AM>> TC> AM>> TC> AM>> TC> AM>> TC>outgoing: AM>> TC> AM>> TC>1369.425046 10.4.62.7 -> 10.4.62.89 SIP/SDP Request: INVITE AM>> TC>sip:1001@10.4.62.89 <sip%3A1001@10.4.62.89> <sip%3A1001@10.4.62.89<sip%253A1001@10.4.62.89>>;transport=udp, AM>> with session AM>> TC>description AM>> TC>1369.426255 10.4.62.7 -> 10.4.62.31 H.225.0 CS: alerting AM>> TC>1369.435950 10.4.62.89 -> 10.4.62.7 SIP Status: 100 Trying AM>> TC>1369.449065 10.4.62.89 -> 10.4.62.7 SIP Status: 180 Ringing AM>> TC>1369.605109 10.4.62.7 -> 10.4.62.31 H.225.0 CS: progress AM>> TC>OpenLogicalChannel AM>> TC>1369.609788 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility AM>> TC>terminalCapabilitySet AM>> TC>1369.610489 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility AM>> TC>masterSlaveDetermination AM>> TC>1369.619071 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty AM>> TC>terminalCapabilitySet AM>> TC>1369.620349 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty AM>> TC>terminalCapabilitySetAck AM>> TC>1369.623215 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility AM>> TC>terminalCapabilitySetAck AM>> TC>1369.625591 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty AM>> TC>masterSlaveDeterminationAck AM>> TC>1369.628174 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility AM>> TC>masterSlaveDeterminationAck AM>> TC>1370.966958 10.4.62.89 -> 10.4.62.7 SIP/SDP Status: 200 OK, with AM>> TC>session description AM>> TC>1370.967431 10.4.62.7 -> 10.4.62.89 SIP Request: ACK AM>> TC>sip:1001@10.4.62.89 <sip%3A1001@10.4.62.89> <sip%3A1001@10.4.62.89<sip%253A1001@10.4.62.89> AM>> >;transport=udp AM>> TC>1370.975172 10.4.62.7 -> 10.4.62.31 H.225.0 CS: connect AM>> TC>1372.354383 10.4.62.89 -> 10.4.62.7 SIP Request: BYE AM>> TC>sip:mod_sofia@10.4.62.7:5060 AM>> TC>1372.355147 10.4.62.7 -> 10.4.62.89 SIP Status: 200 OK AM>> TC>1372.392904 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: AM>> releaseComplete AM>> TC>endSessionCommand AM>> TC>1372.397302 10.4.62.31 -> 10.4.62.7 H.225.0 CS: releaseComplete AM>> TC> AM>> TC> AM>> TC>incoming: AM>> TC> AM>> TC> AM>> TC>1502.817154 10.4.62.31 -> 10.4.62.7 H.225.0 CS: setup AM>> TC>OpenLogicalChannel AM>> TC>1502.833732 10.4.62.7 -> 10.4.62.31 H.225.0 CS: callProceeding AM>> TC>1502.850909 10.4.62.7 -> 10.4.62.89 SIP/SDP Request: INVITE AM>> TC>sip:1001@10.4.62.89 <sip%3A1001@10.4.62.89> <sip%3A1001@10.4.62.89<sip%253A1001@10.4.62.89>>;transport=udp, AM>> with session AM>> TC>description AM>> TC>1502.851758 10.4.62.7 -> 10.4.62.31 H.225.0 CS: alerting AM>> TC>1502.861828 10.4.62.89 -> 10.4.62.7 SIP Status: 100 Trying AM>> TC>1502.875127 10.4.62.89 -> 10.4.62.7 SIP Status: 180 Ringing AM>> TC>1503.033258 10.4.62.7 -> 10.4.62.31 H.225.0 CS: progress AM>> TC>OpenLogicalChannel AM>> TC>1503.037908 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility AM>> TC>terminalCapabilitySet AM>> TC>1503.038608 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility AM>> TC>masterSlaveDetermination AM>> TC>1503.050154 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty AM>> TC>terminalCapabilitySet AM>> TC>1503.051381 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty AM>> TC>terminalCapabilitySetAck AM>> TC>1503.054297 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility AM>> TC>terminalCapabilitySetAck AM>> TC>1503.054917 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty AM>> TC>masterSlaveDeterminationAck AM>> TC>1503.057933 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility AM>> TC>masterSlaveDeterminationAck AM>> TC>1505.485493 10.4.62.89 -> 10.4.62.7 SIP/SDP Status: 200 OK, with AM>> TC>session description AM>> TC>1505.486018 10.4.62.7 -> 10.4.62.89 SIP Request: ACK AM>> TC>sip:1001@10.4.62.89 <sip%3A1001@10.4.62.89> <sip%3A1001@10.4.62.89<sip%253A1001@10.4.62.89> AM>> >;transport=udp AM>> TC>1505.493611 10.4.62.7 -> 10.4.62.31 H.225.0 CS: connect AM>> TC>1509.565959 10.4.62.89 -> 10.4.62.7 SIP Request: BYE AM>> TC>sip:mod_sofia@10.4.62.7:5060 AM>> TC>1509.566722 10.4.62.7 -> 10.4.62.89 SIP Status: 200 OK AM>> TC>1509.577435 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: AM>> releaseComplete AM>> TC>endSessionCommand AM>> TC>1509.582066 10.4.62.31 -> 10.4.62.7 H.225.0 CS: releaseComplete AM>> TC> AM>> TC> AM>> TC> AM>> TC>... i still need to check the CDRs as well but here we are :) AM>> TC> AM>> AM>> can you send a diff? in you call scheme call from h323 endpoint to fs is AM>> not have RAS?, AM>> because i don't have no audio issues in transit from h323 to sip, but my AM>> calls a going AM>> thorough GK and fs is regitered on them, my call scheme is AM>> h323ep-RAS->GK-RAS->fs. AM>> AM>> C уважением With Best Regards AM>> Георгиевский Юрий. Georgiewskiy Yuriy AM>> +7 4872 711666 +7 4872 711666 AM>> факс +7 4872 711143 fax +7 4872 711143 AM>> Компания ООО "Ай Ти Сервис" IT Service Ltd AM>> http://nkoort.ru http://nkoort.ru AM>> JID: GHhost@jabber.tula-ix.net.ru JID: GHhost@jabber.tula-ix.net.ru AM>> YG129-RIPE YG129-RIPE AM>> AM>> _______________________________________________ AM>> FreeSWITCH-users mailing list AM>> FreeSWITCH-users@lists.freeswitch.org AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users AM>> http://www.freeswitch.org AM>> AM>> AM> AM> AM> 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: GHhost@jabber.tula-ix.net.ru JID: GHhost@jabber.tula-ix.net.ru YG129-RIPE YG129-RIPE