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 <bottleman@icf.org.ru>
On 2009-10-23 16:57 +0200, Tihomir Culjaga wrote freeswitch-users@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: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, 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: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 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_sofia@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: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, 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: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 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_sofia@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: GHhost@jabber.tula-ix.net.ru JID: GHhost@jabber.tula-ix.net.ru YG129-RIPE YG129-RIPE
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