Hello, I tried mod_com_g720 with 2 licenses and ran into a problem: What DOES work with G729: * Calling Mailbox (Phone is Snom 360) * Calling external Numbers through Patton gatway What does NOT work * Calling another Snom Phone with G.729 enabled (both Phones use TLS/SRTP) Here an exempt from the log calling phone is 200, called phone is Snom 320: 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:3632 Audio Codec Compare [G729:18:8000:20]/[G729:18:8000:20] 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:2293 Already using G729 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:3565 Set 2833 dtmf send payload to 101 2010-04-28 16:47:57.022827 [DEBUG] sofia.c:4619 Processing updated SDP 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:2580 Audio params are unchanged for sofia/internal/sip:208@192.168.178.126:5060. 2010-04-28 16:47:57.029474 [INFO] mod_com_g729.c:146 DECODER CREATE - 0x9a09510 0x8e774c8 2010-04-28 16:47:57.128943 [ERR] mod_com_g729.c:142 DECODER CREATE FAILED - 0x8e9f9a8 (nil) 2010-04-28 16:47:57.128943 [ERR] switch_core_io.c:327 Codec G.729 decoder error! 2010-04-28 16:47:57.128943 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/200@fs00.telefaks.biz ending bridge by request from read function g729_status Permitted G.729AB channels: 2 Encoders in use: 0 Decoders in use: 0 Here is the dialplan <extension name="Local_Extension_no_Voicemail">. <condition field="destination_number" expression="^\S*$" break="never">. <action application="set" data="transfer_ringback=${us-ring}"/>. <action application="set" data="dialed_ext=208"/>. <action application="set" data="effective_caller_id_number=200"/>. <action application="set" data="effective_caller_id_name=Peter "/>. <action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/>. <action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>. <action application="set" data="dialed_ext=208"/>. <action application="export" data="dialed_ext=208"/>. <action application="ring_ready" />. <action application="bridge" data="user/208@my.domain"/> </condition>. </extension> Just tested it with an Aastra Phone without TLS: Same Behaviour. Anybody has a clue how to solve this? Best regards Peter
do you have lastest git HEAD ? can you update and try again? On Wed, Apr 28, 2010 at 11:03 AM, Peter P GMX <Prometheus001@gmx.net> wrote:
Hello,
I tried mod_com_g720 with 2 licenses and ran into a problem:
What DOES work with G729:
* Calling Mailbox (Phone is Snom 360) * Calling external Numbers through Patton gatway
What does NOT work
* Calling another Snom Phone with G.729 enabled (both Phones use TLS/SRTP)
Here an exempt from the log calling phone is 200, called phone is Snom 320: 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:3632 Audio Codec Compare [G729:18:8000:20]/[G729:18:8000:20] 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:2293 Already using G729 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:3565 Set 2833 dtmf send payload to 101 2010-04-28 16:47:57.022827 [DEBUG] sofia.c:4619 Processing updated SDP 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:2580 Audio params are unchanged for sofia/internal/sip:208@192.168.178.126:5060. 2010-04-28 16:47:57.029474 [INFO] mod_com_g729.c:146 DECODER CREATE - 0x9a09510 0x8e774c8 2010-04-28 16:47:57.128943 [ERR] mod_com_g729.c:142 DECODER CREATE FAILED - 0x8e9f9a8 (nil) 2010-04-28 16:47:57.128943 [ERR] switch_core_io.c:327 Codec G.729 decoder error! 2010-04-28 16:47:57.128943 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/200@fs00.telefaks.biz ending bridge by request from read function
g729_status Permitted G.729AB channels: 2 Encoders in use: 0 Decoders in use: 0
Here is the dialplan <extension name="Local_Extension_no_Voicemail">. <condition field="destination_number" expression="^\S*$" break="never">. <action application="set" data="transfer_ringback=${us-ring}"/>. <action application="set" data="dialed_ext=208"/>. <action application="set" data="effective_caller_id_number=200"/>. <action application="set" data="effective_caller_id_name=Peter "/>. <action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/>. <action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>. <action application="set" data="dialed_ext=208"/>. <action application="export" data="dialed_ext=208"/>. <action application="ring_ready" />. <action application="bridge" data="user/208@my.domain"/> </condition>. </extension>
Just tested it with an Aastra Phone without TLS: Same Behaviour.
Anybody has a clue how to solve this?
Best regards Peter
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
I just updated, with the same result: Anthony Minessale schrieb:
do you have lastest git HEAD ? can you update and try again?
Here's the log: 2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4185 Remote SDP: v=0 o=root 929923105 929923106 IN IP4 192.168.178.125 s=call c=IN IP4 192.168.178.125 t=0 0 m=audio 60566 RTP/AVP 18 8 0 99 3 101 a=rtpmap:18 g729/8000 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4174 Channel sofia/internal/sip:211@192.168.178.125:2048 entering state [ready][200] 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3662 Audio Codec Compare [g729:18:8000:20]/[G729:18:8000:20] 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2372 Set Codec sofia/internal/sip:211@192.168.178.125:2048 G729/8000 20 ms 160 samples 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3595 Set 2833 dtmf send payload to 101 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2612 AUDIO RTP [sofia/internal/sip:211@192.168.178.125:2048] 192.168.178.220 port 12046 -> 192.168.178.125 port 60566 codec: 18 ms: 20 2010-04-30 13:17:21.819392 [DEBUG] switch_rtp.c:1343 Starting timer [soft] 160 bytes per 20ms 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send payload to 101 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf receive payload to 101 2010-04-30 13:17:21.823983 [DEBUG] switch_channel.c:2347 Send signal sofia/internal/200@my.domain [BREAK] 2010-04-30 13:17:21.823983 [NOTICE] sofia.c:4754 Channel [sofia/internal/sip:211@192.168.178.125:2048] has been answered 2010-04-30 13:17:21.830046 [DEBUG] sofia_glue.c:2612 AUDIO RTP [sofia/internal/200@my.domain] 192.168.178.220 port 12006 -> 192.168.178.50 port 12770 codec: 18 ms: 20 2010-04-30 13:17:21.830046 [DEBUG] switch_rtp.c:1343 Starting timer [soft] 160 bytes per 20ms 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send payload to 101 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf receive payload to 101 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1173 Activating Secure RTP SEND 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type: srtp:AES_CM_128_HMAC_SHA1_32 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1153 Activating Secure RTP RECV 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type: srtp:AES_CM_128_HMAC_SHA1_32 2010-04-30 13:17:21.832815 [DEBUG] mod_sofia.c:663 Local SDP sofia/internal/200@my.domain: v=0 o=FreeSWITCH 1272614235 1272614236 IN IP4 192.168.178.220 s=FreeSWITCH c=IN IP4 192.168.178.220 t=0 0 m=audio 12006 RTP/SAVP 18 101 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:OoR/ZSNzik9jFHdcbzXyGXGSO5E3mGT6tvcpvsqK 2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:703 Send signal sofia/internal/sip:211@192.168.178.125:2048 [BREAK] 2010-04-30 13:17:21.834022 [DEBUG] sofia.c:4174 Channel sofia/internal/200@my.domain entering state [completed][200] 2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/200@my.domain [BREAK] 2010-04-30 13:17:21.835065 [NOTICE] switch_ivr_originate.c:3174 Channel [sofia/internal/200@my.domain] has been answered 2010-04-30 13:17:21.835065 [DEBUG] switch_ivr_originate.c:3219 Originate Resulted in Success: [sofia/internal/sip:211@192.168.178.125:2048] 2010-04-30 13:17:21.838531 [DEBUG] switch_ivr_originate.c:3219 Originate Resulted in Success: [sofia/internal/sip:211@192.168.178.125:2048] 2010-04-30 13:17:21.838531 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/sip:211@192.168.178.125:2048 [BREAK] 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/200@my.domain [BREAK] 2010-04-30 13:17:21.839601 [DEBUG] switch_ivr_bridge.c:1182 (sofia/internal/sip:211@192.168.178.125:2048) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/sip:211@192.168.178.125:2048 [BREAK] 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:211@192.168.178.125:2048) Running State Change CS_EXCHANGE_MEDIA 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:211@192.168.178.125:2048) State EXCHANGE_MEDIA 2010-04-30 13:17:21.839601 [DEBUG] mod_sofia.c:534 SOFIA EXCHANGE_MEDIA 2010-04-30 13:17:21.861445 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/sip:211@192.168.178.125:2048 2010-04-30 13:17:21.891110 [DEBUG] switch_rtp.c:2443 Correct ip/port confirmed. 2010-04-30 13:17:21.951915 [DEBUG] switch_rtp.c:2443 Correct ip/port confirmed. 2010-04-30 13:17:21.951915 [INFO] mod_com_g729.c:146 DECODER CREATE - 0x904e070 0x8fb39c0 2010-04-30 13:17:22.031586 [INFO] mod_com_g729.c:146 DECODER CREATE - 0x8fcd7e8 0x8e448a0 2010-04-30 13:17:22.247792 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/200@my.domain 2010-04-30 13:17:22.253759 [ERR] mod_com_g729.c:142 DECODER CREATE FAILED - 0x90990a0 (nil) 2010-04-30 13:17:22.253759 [ERR] switch_core_io.c:327 Codec G.729 decoder error! 2010-04-30 13:17:22.253759 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/200@my.domain ending bridge by request from read function
Can you refer your support with the commercial G729 to consulting@freeswitch.org. I'll try to reproduce this issue today and see if I have the same results. /b On Apr 30, 2010, at 6:23 AM, Peter P GMX wrote:
I just updated, with the same result:
Anthony Minessale schrieb:
do you have lastest git HEAD ? can you update and try again?
Here's the log:
log snippets are not useful. as always, we need full log from beginning to end of the complete call with the inline sip trace enabled. On Fri, Apr 30, 2010 at 6:23 AM, Peter P GMX <Prometheus001@gmx.net> wrote:
I just updated, with the same result:
Anthony Minessale schrieb:
do you have lastest git HEAD ? can you update and try again?
Here's the log:
2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4185 Remote SDP: v=0 o=root 929923105 929923106 IN IP4 192.168.178.125 s=call c=IN IP4 192.168.178.125 t=0 0 m=audio 60566 RTP/AVP 18 8 0 99 3 101 a=rtpmap:18 g729/8000 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4174 Channel sofia/internal/sip:211@192.168.178.125:2048 entering state [ready][200] 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3662 Audio Codec Compare [g729:18:8000:20]/[G729:18:8000:20] 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2372 Set Codec sofia/internal/sip:211@192.168.178.125:2048 G729/8000 20 ms 160 samples 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3595 Set 2833 dtmf send payload to 101 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2612 AUDIO RTP [sofia/internal/sip:211@192.168.178.125:2048] 192.168.178.220 port 12046 -> 192.168.178.125 port 60566 codec: 18 ms: 20 2010-04-30 13:17:21.819392 [DEBUG] switch_rtp.c:1343 Starting timer [soft] 160 bytes per 20ms 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send payload to 101 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf receive payload to 101 2010-04-30 13:17:21.823983 [DEBUG] switch_channel.c:2347 Send signal sofia/internal/200@my.domain [BREAK] 2010-04-30 13:17:21.823983 [NOTICE] sofia.c:4754 Channel [sofia/internal/sip:211@192.168.178.125:2048] has been answered 2010-04-30 13:17:21.830046 [DEBUG] sofia_glue.c:2612 AUDIO RTP [sofia/internal/200@my.domain] 192.168.178.220 port 12006 -> 192.168.178.50 port 12770 codec: 18 ms: 20 2010-04-30 13:17:21.830046 [DEBUG] switch_rtp.c:1343 Starting timer [soft] 160 bytes per 20ms 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send payload to 101 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf receive payload to 101 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1173 Activating Secure RTP SEND 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type: srtp:AES_CM_128_HMAC_SHA1_32 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1153 Activating Secure RTP RECV 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type: srtp:AES_CM_128_HMAC_SHA1_32 2010-04-30 13:17:21.832815 [DEBUG] mod_sofia.c:663 Local SDP sofia/internal/200@my.domain: v=0 o=FreeSWITCH 1272614235 1272614236 IN IP4 192.168.178.220 s=FreeSWITCH c=IN IP4 192.168.178.220 t=0 0 m=audio 12006 RTP/SAVP 18 101 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:OoR/ZSNzik9jFHdcbzXyGXGSO5E3mGT6tvcpvsqK
2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:703 Send signal sofia/internal/sip:211@192.168.178.125:2048 [BREAK] 2010-04-30 13:17:21.834022 [DEBUG] sofia.c:4174 Channel sofia/internal/200@my.domain entering state [completed][200] 2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/200@my.domain [BREAK] 2010-04-30 13:17:21.835065 [NOTICE] switch_ivr_originate.c:3174 Channel [sofia/internal/200@my.domain] has been answered 2010-04-30 13:17:21.835065 [DEBUG] switch_ivr_originate.c:3219 Originate Resulted in Success: [sofia/internal/sip:211@192.168.178.125:2048] 2010-04-30 13:17:21.838531 [DEBUG] switch_ivr_originate.c:3219 Originate Resulted in Success: [sofia/internal/sip:211@192.168.178.125:2048] 2010-04-30 13:17:21.838531 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/sip:211@192.168.178.125:2048 [BREAK] 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/200@my.domain [BREAK] 2010-04-30 13:17:21.839601 [DEBUG] switch_ivr_bridge.c:1182 (sofia/internal/sip:211@192.168.178.125:2048) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/sip:211@192.168.178.125:2048 [BREAK] 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:211@192.168.178.125:2048) Running State Change CS_EXCHANGE_MEDIA 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:211@192.168.178.125:2048) State EXCHANGE_MEDIA 2010-04-30 13:17:21.839601 [DEBUG] mod_sofia.c:534 SOFIA EXCHANGE_MEDIA 2010-04-30 13:17:21.861445 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/sip:211@192.168.178.125:2048 2010-04-30 13:17:21.891110 [DEBUG] switch_rtp.c:2443 Correct ip/port confirmed. 2010-04-30 13:17:21.951915 [DEBUG] switch_rtp.c:2443 Correct ip/port confirmed. 2010-04-30 13:17:21.951915 [INFO] mod_com_g729.c:146 DECODER CREATE - 0x904e070 0x8fb39c0 2010-04-30 13:17:22.031586 [INFO] mod_com_g729.c:146 DECODER CREATE - 0x8fcd7e8 0x8e448a0 2010-04-30 13:17:22.247792 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/200@my.domain 2010-04-30 13:17:22.253759 [ERR] mod_com_g729.c:142 DECODER CREATE FAILED - 0x90990a0 (nil) 2010-04-30 13:17:22.253759 [ERR] switch_core_io.c:327 Codec G.729 decoder error! 2010-04-30 13:17:22.253759 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/200@my.domain ending bridge by request from read function
_______________________________________________ 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_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Peter, every time you email me directly, when I reply it bounces, I have sent you a few emails that you do not seem to receive. Essentially if you need to record a g729 call it will cost an additional license because the recording subsystem needs to force a decoding path so it can mux the data together etc. This is what it said: Hi. This is the qmail-send program at mx0.gmx.net. I'm afraid I wasn't able to deliver your message to the following addresses. This is a permanent error; I've given up. Sorry it didn't work out. <psteinbach@de.lhsgroup.com>: Sorry,_I_couldn't_find_any_host_named_de.lhsgroup.com._(#5.1.2)/ On Fri, Apr 30, 2010 at 6:23 AM, Peter P GMX <Prometheus001@gmx.net> wrote:
I just updated, with the same result:
Anthony Minessale schrieb:
do you have lastest git HEAD ? can you update and try again?
Here's the log:
2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4185 Remote SDP: v=0 o=root 929923105 929923106 IN IP4 192.168.178.125 s=call c=IN IP4 192.168.178.125 t=0 0 m=audio 60566 RTP/AVP 18 8 0 99 3 101 a=rtpmap:18 g729/8000 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4174 Channel sofia/internal/sip:211@192.168.178.125:2048 entering state [ready][200] 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3662 Audio Codec Compare [g729:18:8000:20]/[G729:18:8000:20] 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2372 Set Codec sofia/internal/sip:211@192.168.178.125:2048 G729/8000 20 ms 160 samples 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3595 Set 2833 dtmf send payload to 101 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2612 AUDIO RTP [sofia/internal/sip:211@192.168.178.125:2048] 192.168.178.220 port 12046 -> 192.168.178.125 port 60566 codec: 18 ms: 20 2010-04-30 13:17:21.819392 [DEBUG] switch_rtp.c:1343 Starting timer [soft] 160 bytes per 20ms 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send payload to 101 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf receive payload to 101 2010-04-30 13:17:21.823983 [DEBUG] switch_channel.c:2347 Send signal sofia/internal/200@my.domain [BREAK] 2010-04-30 13:17:21.823983 [NOTICE] sofia.c:4754 Channel [sofia/internal/sip:211@192.168.178.125:2048] has been answered 2010-04-30 13:17:21.830046 [DEBUG] sofia_glue.c:2612 AUDIO RTP [sofia/internal/200@my.domain] 192.168.178.220 port 12006 -> 192.168.178.50 port 12770 codec: 18 ms: 20 2010-04-30 13:17:21.830046 [DEBUG] switch_rtp.c:1343 Starting timer [soft] 160 bytes per 20ms 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send payload to 101 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf receive payload to 101 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1173 Activating Secure RTP SEND 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type: srtp:AES_CM_128_HMAC_SHA1_32 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1153 Activating Secure RTP RECV 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type: srtp:AES_CM_128_HMAC_SHA1_32 2010-04-30 13:17:21.832815 [DEBUG] mod_sofia.c:663 Local SDP sofia/internal/200@my.domain: v=0 o=FreeSWITCH 1272614235 1272614236 IN IP4 192.168.178.220 s=FreeSWITCH c=IN IP4 192.168.178.220 t=0 0 m=audio 12006 RTP/SAVP 18 101 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:OoR/ZSNzik9jFHdcbzXyGXGSO5E3mGT6tvcpvsqK
2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:703 Send signal sofia/internal/sip:211@192.168.178.125:2048 [BREAK] 2010-04-30 13:17:21.834022 [DEBUG] sofia.c:4174 Channel sofia/internal/200@my.domain entering state [completed][200] 2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/200@my.domain [BREAK] 2010-04-30 13:17:21.835065 [NOTICE] switch_ivr_originate.c:3174 Channel [sofia/internal/200@my.domain] has been answered 2010-04-30 13:17:21.835065 [DEBUG] switch_ivr_originate.c:3219 Originate Resulted in Success: [sofia/internal/sip:211@192.168.178.125:2048] 2010-04-30 13:17:21.838531 [DEBUG] switch_ivr_originate.c:3219 Originate Resulted in Success: [sofia/internal/sip:211@192.168.178.125:2048] 2010-04-30 13:17:21.838531 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/sip:211@192.168.178.125:2048 [BREAK] 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/200@my.domain [BREAK] 2010-04-30 13:17:21.839601 [DEBUG] switch_ivr_bridge.c:1182 (sofia/internal/sip:211@192.168.178.125:2048) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/sip:211@192.168.178.125:2048 [BREAK] 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:211@192.168.178.125:2048) Running State Change CS_EXCHANGE_MEDIA 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:211@192.168.178.125:2048) State EXCHANGE_MEDIA 2010-04-30 13:17:21.839601 [DEBUG] mod_sofia.c:534 SOFIA EXCHANGE_MEDIA 2010-04-30 13:17:21.861445 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/sip:211@192.168.178.125:2048 2010-04-30 13:17:21.891110 [DEBUG] switch_rtp.c:2443 Correct ip/port confirmed. 2010-04-30 13:17:21.951915 [DEBUG] switch_rtp.c:2443 Correct ip/port confirmed. 2010-04-30 13:17:21.951915 [INFO] mod_com_g729.c:146 DECODER CREATE - 0x904e070 0x8fb39c0 2010-04-30 13:17:22.031586 [INFO] mod_com_g729.c:146 DECODER CREATE - 0x8fcd7e8 0x8e448a0 2010-04-30 13:17:22.247792 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/200@my.domain 2010-04-30 13:17:22.253759 [ERR] mod_com_g729.c:142 DECODER CREATE FAILED - 0x90990a0 (nil) 2010-04-30 13:17:22.253759 [ERR] switch_core_io.c:327 Codec G.729 decoder error! 2010-04-30 13:17:22.253759 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/200@my.domain ending bridge by request from read function
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Did you real my last email? Do not send me email directly unless its from a valid email address that I can reply to. I cannot reply to you on the direct email because it bounces back to me EVERY TIME. you need to fix your return email address. On Tue, May 4, 2010 at 12:12 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Peter, every time you email me directly, when I reply it bounces, I have sent you a few emails that you do not seem to receive.
Essentially if you need to record a g729 call it will cost an additional license because the recording subsystem needs to force a decoding path so it can mux the data together etc.
This is what it said:
Hi. This is the qmail-send program at mx0.gmx.net. I'm afraid I wasn't able to deliver your message to the following addresses. This is a permanent error; I've given up. Sorry it didn't work out.
<psteinbach@de.lhsgroup.com>: Sorry,_I_couldn't_find_any_host_named_de.lhsgroup.com._(#5.1.2)/
On Fri, Apr 30, 2010 at 6:23 AM, Peter P GMX <Prometheus001@gmx.net>wrote:
I just updated, with the same result:
Anthony Minessale schrieb:
do you have lastest git HEAD ? can you update and try again?
Here's the log:
2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4185 Remote SDP: v=0 o=root 929923105 929923106 IN IP4 192.168.178.125 s=call c=IN IP4 192.168.178.125 t=0 0 m=audio 60566 RTP/AVP 18 8 0 99 3 101 a=rtpmap:18 g729/8000 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4174 Channel sofia/internal/sip:211@192.168.178.125:2048 entering state [ready][200] 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3662 Audio Codec Compare [g729:18:8000:20]/[G729:18:8000:20] 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2372 Set Codec sofia/internal/sip:211@192.168.178.125:2048 G729/8000 20 ms 160 samples 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3595 Set 2833 dtmf send payload to 101 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2612 AUDIO RTP [sofia/internal/sip:211@192.168.178.125:2048] 192.168.178.220 port 12046 -> 192.168.178.125 port 60566 codec: 18 ms: 20 2010-04-30 13:17:21.819392 [DEBUG] switch_rtp.c:1343 Starting timer [soft] 160 bytes per 20ms 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send payload to 101 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf receive payload to 101 2010-04-30 13:17:21.823983 [DEBUG] switch_channel.c:2347 Send signal sofia/internal/200@my.domain [BREAK] 2010-04-30 13:17:21.823983 [NOTICE] sofia.c:4754 Channel [sofia/internal/sip:211@192.168.178.125:2048] has been answered 2010-04-30 13:17:21.830046 [DEBUG] sofia_glue.c:2612 AUDIO RTP [sofia/internal/200@my.domain] 192.168.178.220 port 12006 -> 192.168.178.50 port 12770 codec: 18 ms: 20 2010-04-30 13:17:21.830046 [DEBUG] switch_rtp.c:1343 Starting timer [soft] 160 bytes per 20ms 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send payload to 101 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf receive payload to 101 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1173 Activating Secure RTP SEND 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type: srtp:AES_CM_128_HMAC_SHA1_32 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1153 Activating Secure RTP RECV 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type: srtp:AES_CM_128_HMAC_SHA1_32 2010-04-30 13:17:21.832815 [DEBUG] mod_sofia.c:663 Local SDP sofia/internal/200@my.domain: v=0 o=FreeSWITCH 1272614235 1272614236 IN IP4 192.168.178.220 s=FreeSWITCH c=IN IP4 192.168.178.220 t=0 0 m=audio 12006 RTP/SAVP 18 101 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:OoR/ZSNzik9jFHdcbzXyGXGSO5E3mGT6tvcpvsqK
2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:703 Send signal sofia/internal/sip:211@192.168.178.125:2048 [BREAK] 2010-04-30 13:17:21.834022 [DEBUG] sofia.c:4174 Channel sofia/internal/200@my.domain entering state [completed][200] 2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/200@my.domain [BREAK] 2010-04-30 13:17:21.835065 [NOTICE] switch_ivr_originate.c:3174 Channel [sofia/internal/200@my.domain] has been answered 2010-04-30 13:17:21.835065 [DEBUG] switch_ivr_originate.c:3219 Originate Resulted in Success: [sofia/internal/sip:211@192.168.178.125:2048] 2010-04-30 13:17:21.838531 [DEBUG] switch_ivr_originate.c:3219 Originate Resulted in Success: [sofia/internal/sip:211@192.168.178.125:2048] 2010-04-30 13:17:21.838531 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/sip:211@192.168.178.125:2048 [BREAK] 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/200@my.domain [BREAK] 2010-04-30 13:17:21.839601 [DEBUG] switch_ivr_bridge.c:1182 (sofia/internal/sip:211@192.168.178.125:2048) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/sip:211@192.168.178.125:2048 [BREAK] 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:211@192.168.178.125:2048) Running State Change CS_EXCHANGE_MEDIA 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:211@192.168.178.125:2048) State EXCHANGE_MEDIA 2010-04-30 13:17:21.839601 [DEBUG] mod_sofia.c:534 SOFIA EXCHANGE_MEDIA 2010-04-30 13:17:21.861445 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/sip:211@192.168.178.125:2048 2010-04-30 13:17:21.891110 [DEBUG] switch_rtp.c:2443 Correct ip/port confirmed. 2010-04-30 13:17:21.951915 [DEBUG] switch_rtp.c:2443 Correct ip/port confirmed. 2010-04-30 13:17:21.951915 [INFO] mod_com_g729.c:146 DECODER CREATE - 0x904e070 0x8fb39c0 2010-04-30 13:17:22.031586 [INFO] mod_com_g729.c:146 DECODER CREATE - 0x8fcd7e8 0x8e448a0 2010-04-30 13:17:22.247792 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/200@my.domain 2010-04-30 13:17:22.253759 [ERR] mod_com_g729.c:142 DECODER CREATE FAILED - 0x90990a0 (nil) 2010-04-30 13:17:22.253759 [ERR] switch_core_io.c:327 Codec G.729 decoder error! 2010-04-30 13:17:22.253759 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/200@my.domain ending bridge by request from read function
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Hello Anthony, thanks, I received your mail. I should use a different email adreess in the future. My email provider sometimes sends these strange notifications as you have seen. eMails from this mailing lists however work. Every attempt to stop this behaviour failed, as the email provider has no clue how to solve this. Sorry for this inconvenience. Concerning the G.729 problem, this is not 100% clear to me. When I connect 2 phones with G.729 without recording, I can see that no license is used, as the call is in bypass mode: Permitted G.729AB channels: 2 Encoders in use: 0 Decoders in use: 0 So when I try to record this call, I will need 3 licenses? 1 for each call leg and one for the recording? Best regards Peter
Well if you have a g729 call up and its G729 passthru encoder/decoders will not be allocated till needed. If you record the call you'll require two decoders to record the call since it has to decode both sides and mux the call. I did test calls that were G729 end to end and no coders got allocated. The allocation happened when recording started. Also if you do fake ringback you'll allocate coders too. /b On May 5, 2010, at 6:13 PM, Peter P GMX wrote:
Hello Anthony,
thanks, I received your mail. I should use a different email adreess in the future. My email provider sometimes sends these strange notifications as you have seen. eMails from this mailing lists however work. Every attempt to stop this behaviour failed, as the email provider has no clue how to solve this. Sorry for this inconvenience.
Concerning the G.729 problem, this is not 100% clear to me. When I connect 2 phones with G.729 without recording, I can see that no license is used, as the call is in bypass mode: Permitted G.729AB channels: 2 Encoders in use: 0 Decoders in use: 0
So when I try to record this call, I will need 3 licenses? 1 for each call leg and one for the recording?
Best regards Peter
if the call is in passthru it will use 0 licenses if you begin to record it will use 2 decoder until you stop recording if the call is transcoding to g729 to ulaw and you are careful to record on the ulaw leg it will cost no extra licenses other than the one already in use otherwise it will cost 1 if you do it on the 729 side. On Wed, May 5, 2010 at 6:13 PM, Peter P GMX <Prometheus001@gmx.net> wrote:
Hello Anthony,
thanks, I received your mail. I should use a different email adreess in the future. My email provider sometimes sends these strange notifications as you have seen. eMails from this mailing lists however work. Every attempt to stop this behaviour failed, as the email provider has no clue how to solve this. Sorry for this inconvenience.
Concerning the G.729 problem, this is not 100% clear to me. When I connect 2 phones with G.729 without recording, I can see that no license is used, as the call is in bypass mode: Permitted G.729AB channels: 2 Encoders in use: 0 Decoders in use: 0
So when I try to record this call, I will need 3 licenses? 1 for each call leg and one for the recording?
Best regards Peter
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
participants (3)
-
Anthony Minessale
-
Brian West
-
Peter P GMX