if you were on trunk that line of code would be gone.<br>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.<br>
<br><br><div class="gmail_quote">2009/10/23 Georgiewskiy Yuriy <span dir="ltr"><<a href="mailto:bottleman@icf.org.ru">bottleman@icf.org.ru</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
On 2009-10-23 16:57 +0200, Tihomir Culjaga wrote freeswitch-users@lists.fre...:<br>
<br>
i have question to developers about one proce in fs<br>
<br>
src/switch_ivr_originate.c<br>
<br>
static switch_status_t originate_on_consume_media_transmit(switch_core_session_t *session)<br>
{<br>
switch_channel_t *channel = switch_core_session_get_channel(session);<br>
<br>
if (!switch_channel_test_flag(channel, CF_PROXY_MODE)) {<br>
while (switch_channel_get_state(channel) == CS_CONSUME_MEDIA && !switch_channel_test_flag(chann<br>
if (!switch_channel_media_ready(channel)) {<br>
switch_yield(10000);<br>
} else {<br>
switch_ivr_sleep(session, 10, SWITCH_FALSE, NULL);<br>
}<br>
}<br>
}<br>
<br>
switch_channel_clear_state_handler(channel, &originate_state_handlers);<br>
<br>
return SWITCH_STATUS_FALSE;<br>
}<br>
<br>
what exacly it do?<br>
<br>
call scheme like this sip->fs->h323->gk->h323(on same fs)->fs(same too) and there i have no audio issues.<br>
if bridge connect while it sleep i have audio, if it not sleep while bridge connect i have no audio.<br>
<br>
TC>a solution to H323 endpoint => FS => SIP user no audio issue<br>
TC><br>
TC>is to disable a wait for tx Audio ... for case<br>
TC>SWITCH_MESSAGE_INDICATE_ANSWER:{<br>
<div class="im">TC><br>
TC>//m_txAudioOpened.Wait();<br>
TC><br>
</div>TC><br>
TC> case SWITCH_MESSAGE_INDICATE_ANSWER:{<br>
TC><br>
TC> switch_log_printf(SWITCH_CHANNEL_LOG,<br>
TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event\n");<br>
TC><br>
TC> if (switch_channel_test_flag(channel, CF_OUTBOUND))<br>
TC>{<br>
TC><br>
TC> switch_log_printf(SWITCH_CHANNEL_LOG,<br>
TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event - CF_OUTBOUND<br>
TC>\n");<br>
TC> return SWITCH_STATUS_FALSE;<br>
TC> }<br>
TC> AnsweringCall(H323Connection::AnswerCallNow);<br>
TC><br>
TC> switch_log_printf(SWITCH_CHANNEL_LOG,<br>
TC>SWITCH_LOG_CONSOLE, "ANSWER: suppose the call is Answered Now\n");<br>
TC> PTRACE(4, "mod_h323\tMedia started on connection "<br>
TC><< *this);<br>
TC><br>
TC> // test<br>
TC> //switch_channel_mark_answered(m_fsChannel);<br>
TC><br>
TC> m_rxAudioOpened.Wait();<br>
TC> switch_log_printf(SWITCH_CHANNEL_LOG,<br>
TC>SWITCH_LOG_CONSOLE, "ANSWER: wait for m_rxAudioOpened\n");<br>
TC> //m_txAudioOpened.Wait();<br>
TC> switch_log_printf(SWITCH_CHANNEL_LOG,<br>
TC>SWITCH_LOG_CONSOLE, "ANSWER: we disable wait for m_txAudioOpened\n");<br>
TC><br>
TC> switch_log_printf(SWITCH_CHANNEL_LOG,<br>
TC>SWITCH_LOG_CONSOLE, "ANSWER: were waiting for rx/tx AudioOpen\n");<br>
TC><br>
TC> if (!switch_channel_test_flag(m_fsChannel,<br>
TC>CF_EARLY_MEDIA)) {<br>
TC><br>
TC> switch_log_printf(SWITCH_CHANNEL_LOG,<br>
TC>SWITCH_LOG_CONSOLE, "ANSWER: we have early media\n");<br>
TC><br>
TC> PTRACE(4,<br>
TC>"mod_h323\t-------------------->switch_channel_mark_answered(m_fsChannel) "<br>
TC><< *this);<br>
TC> switch_channel_mark_answered(m_fsChannel);<br>
TC> switch_log_printf(SWITCH_CHANNEL_LOG,<br>
TC>SWITCH_LOG_CONSOLE, "ANSWER: answered in early Media\n");<br>
TC> }<br>
TC> break;<br>
TC> }<br>
TC><br>
TC><br>
TC>Now, I'm able to both originate and terminate cals with 2-way audio...<br>
TC>the signaling looks correct...<br>
TC><br>
TC><br>
TC><br>
TC>outgoing:<br>
TC><br>
TC>1369.425046 10.4.62.7 -> 10.4.62.89 SIP/SDP Request: INVITE<br>
TC><a href="mailto:sip%3A1001@10.4.62.89">sip:1001@10.4.62.89</a> <<a href="mailto:sip%253A1001@10.4.62.89">sip%3A1001@10.4.62.89</a>>;transport=udp, with session<br>
TC>description<br>
TC>1369.426255 10.4.62.7 -> 10.4.62.31 H.225.0 CS: alerting<br>
TC>1369.435950 10.4.62.89 -> 10.4.62.7 SIP Status: 100 Trying<br>
TC>1369.449065 10.4.62.89 -> 10.4.62.7 SIP Status: 180 Ringing<br>
TC>1369.605109 10.4.62.7 -> 10.4.62.31 H.225.0 CS: progress<br>
TC>OpenLogicalChannel<br>
TC>1369.609788 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility<br>
TC>terminalCapabilitySet<br>
TC>1369.610489 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility<br>
TC>masterSlaveDetermination<br>
TC>1369.619071 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty<br>
TC>terminalCapabilitySet<br>
TC>1369.620349 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty<br>
TC>terminalCapabilitySetAck<br>
TC>1369.623215 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility<br>
TC>terminalCapabilitySetAck<br>
TC>1369.625591 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty<br>
TC>masterSlaveDeterminationAck<br>
TC>1369.628174 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility<br>
TC>masterSlaveDeterminationAck<br>
TC>1370.966958 10.4.62.89 -> 10.4.62.7 SIP/SDP Status: 200 OK, with<br>
TC>session description<br>
TC>1370.967431 10.4.62.7 -> 10.4.62.89 SIP Request: ACK<br>
TC><a href="mailto:sip%3A1001@10.4.62.89">sip:1001@10.4.62.89</a> <<a href="mailto:sip%253A1001@10.4.62.89">sip%3A1001@10.4.62.89</a>>;transport=udp<br>
TC>1370.975172 10.4.62.7 -> 10.4.62.31 H.225.0 CS: connect<br>
TC>1372.354383 10.4.62.89 -> 10.4.62.7 SIP Request: BYE<br>
TC><a href="http://sip:mod_sofia@10.4.62.7:5060" target="_blank">sip:mod_sofia@10.4.62.7:5060</a><br>
TC>1372.355147 10.4.62.7 -> 10.4.62.89 SIP Status: 200 OK<br>
TC>1372.392904 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: releaseComplete<br>
TC>endSessionCommand<br>
TC>1372.397302 10.4.62.31 -> 10.4.62.7 H.225.0 CS: releaseComplete<br>
TC><br>
TC><br>
TC>incoming:<br>
TC><br>
TC><br>
TC>1502.817154 10.4.62.31 -> 10.4.62.7 H.225.0 CS: setup<br>
TC>OpenLogicalChannel<br>
TC>1502.833732 10.4.62.7 -> 10.4.62.31 H.225.0 CS: callProceeding<br>
TC>1502.850909 10.4.62.7 -> 10.4.62.89 SIP/SDP Request: INVITE<br>
TC><a href="mailto:sip%3A1001@10.4.62.89">sip:1001@10.4.62.89</a> <<a href="mailto:sip%253A1001@10.4.62.89">sip%3A1001@10.4.62.89</a>>;transport=udp, with session<br>
TC>description<br>
TC>1502.851758 10.4.62.7 -> 10.4.62.31 H.225.0 CS: alerting<br>
TC>1502.861828 10.4.62.89 -> 10.4.62.7 SIP Status: 100 Trying<br>
TC>1502.875127 10.4.62.89 -> 10.4.62.7 SIP Status: 180 Ringing<br>
TC>1503.033258 10.4.62.7 -> 10.4.62.31 H.225.0 CS: progress<br>
TC>OpenLogicalChannel<br>
TC>1503.037908 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility<br>
TC>terminalCapabilitySet<br>
TC>1503.038608 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility<br>
TC>masterSlaveDetermination<br>
TC>1503.050154 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty<br>
TC>terminalCapabilitySet<br>
TC>1503.051381 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty<br>
TC>terminalCapabilitySetAck<br>
TC>1503.054297 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility<br>
TC>terminalCapabilitySetAck<br>
TC>1503.054917 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: empty<br>
TC>masterSlaveDeterminationAck<br>
TC>1503.057933 10.4.62.31 -> 10.4.62.7 H.225.0/H.245 CS: facility<br>
TC>masterSlaveDeterminationAck<br>
TC>1505.485493 10.4.62.89 -> 10.4.62.7 SIP/SDP Status: 200 OK, with<br>
TC>session description<br>
TC>1505.486018 10.4.62.7 -> 10.4.62.89 SIP Request: ACK<br>
TC><a href="mailto:sip%3A1001@10.4.62.89">sip:1001@10.4.62.89</a> <<a href="mailto:sip%253A1001@10.4.62.89">sip%3A1001@10.4.62.89</a>>;transport=udp<br>
TC>1505.493611 10.4.62.7 -> 10.4.62.31 H.225.0 CS: connect<br>
TC>1509.565959 10.4.62.89 -> 10.4.62.7 SIP Request: BYE<br>
TC><a href="http://sip:mod_sofia@10.4.62.7:5060" target="_blank">sip:mod_sofia@10.4.62.7:5060</a><br>
TC>1509.566722 10.4.62.7 -> 10.4.62.89 SIP Status: 200 OK<br>
TC>1509.577435 10.4.62.7 -> 10.4.62.31 H.225.0/H.245 CS: releaseComplete<br>
TC>endSessionCommand<br>
TC>1509.582066 10.4.62.31 -> 10.4.62.7 H.225.0 CS: releaseComplete<br>
TC><br>
TC><br>
TC><br>
TC>... i still need to check the CDRs as well but here we are :)<br>
TC><br>
<br>
can you send a diff? in you call scheme call from h323 endpoint to fs is not have RAS?,<br>
because i don't have no audio issues in transit from h323 to sip, but my calls a going<br>
thorough GK and fs is regitered on them, my call scheme is h323ep-RAS->GK-RAS->fs.<br>
<div><div></div><div class="h5"><br>
C ีืมึลฮษลอ With Best Regards<br>
็ลฯาวษลืำหษส เาษส. Georgiewskiy Yuriy<br>
+7 4872 711666 +7 4872 711666<br>
ฦมหำ +7 4872 711143 fax +7 4872 711143<br>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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