G729 Decoder Error
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ZeuS
Apr 10, 2017, 3:11:39 PM4/10/17
to ASTPP
Hi Guys
As usual, the G729 giving same error, sometime its work and sometimes it start giving error
2017-04-10 21:04:48.910473 [DEBUG] mod_sofia.c:628 SOFIA EXCHANGE_MEDIA
2017-04-10 21:04:48.910473 [DEBUG] switch_ivr_async.c:1496 No silence detection configured; assuming start of speech
2017-04-10 21:04:48.910473 [DEBUG] switch_ivr_async.c:1496 No silence detection configured; assuming start of speech
2017-04-10 21:04:48.950473 [DEBUG] switch_rtp.c:6707 Correct audio ip/port confirmed.
2017-04-10 21:04:48.950473 [DEBUG] switch_core_io.c:448 Setting BUG Codec G729:18
2017-04-10 21:04:48.950473 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode!
2017-04-10 21:04:48.950473 [ERR] switch_core_io.c:633 Codec G.729 decoder error! [1]
2017-04-10 21:04:48.950473 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode!
2017-04-10 21:04:48.950473 [ERR] switch_core_io.c:1255 Codec G.729 decoder error!
i did everything Reload (mod_g729, sofia , mod_db , reloadxml, mod_xml_curl)
i restarted Freeswitch
inbound and outbound are g729,ulaw
everything is in g729 (caller , gateway) but still getting this, any other suggestion ???
Thank you
inbound and outbound are g729,ulaw
everything is in g729 (caller , gateway) but still getting this, any other suggestion ???
Thank you
Samir Doshi
Apr 10, 2017, 9:28:33 PM4/10/17
to ASTPP
If everything is in G729 then freeswitch never give you error. Please check sip log and confirm the codecs which client and provider is using and make sure G729 must get negotiated.
Best Regards
--
Samir Doshi
iNextrix Technologies Pvt. Ltd.
http://www.inextrix.com
Disclaimer:
The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it.
--
Samir Doshi
iNextrix Technologies Pvt. Ltd.
http://www.inextrix.com
Disclaimer:
The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it.
--
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ZeuS
Apr 11, 2017, 3:38:41 AM4/11/17
to ASTPP
Hi, Samir,
Gateway :
here you have logs
ASTPP:
--------------------
ASTPP:
INVITE sip:44442137...@xx.xx.xx.xx;transport=UDP SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK-524287-1---9a50c65afb5b400f;rport
Max-Forwards: 70
Contact: <sip:85313...@xx.xx.xx.xx:3199;transport=UDP>
To: <sip:44442137...@xx.xx.xx.xx;transport=UDP>
From: <sip:85313...@xx.xx.xx.xx;transport=UDP>;tag=cbb2432e
Call-ID: 4Kz_wVl_2MglXz6xoFmomw..
CSeq: 1 INVITE
Content-Type: application/sdp
User-Agent: Zoiper rv2.8.30
Allow-Events: presence, kpml, talk
Content-Length: 243
v=0
o=Zoiper 0 0 IN IP4 192.168.1.101
s=Zoiper
c=IN IP4 192.168.1.101
t=0 0
m=audio 36234 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
--------------------
send 1115 bytes to udp/[xx.xx.xx.xx]:5060 at 09:25:01.887209:
------------------------------------------------------------------------
INVITE sip:44442137...@xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx;rport;branch=z9hG4bKNN8ZBK4ZeUZ0K
Max-Forwards: 69
From: "8531320135" <sip:48103...@xx.xx.xx.xx>;tag=c97XHvpUgB4FH
To: <sip:44442137...@xx.xx.xx.xx>
Call-ID: d13162bc-992a-1235-bf82-6c0b84deb0be
CSeq: 105611686 INVITE
Contact: <sip:gw+...@xx.xx.xx.xx:5060;transport=udp;gw=ALG>
User-Agent: ASTPP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
2017-04-11 09:25:01.870443 [DEBUG] switch_core_state_machine.c:602 (sofia/default/4444213772368578) State ROUTING
Content-Type: application/sdp
Content-Disposition: session
2017-04-11 09:25:01.870443 [DEBUG] mod_sofia.c:142 sofia/default/4444213772368578 SOFIA ROUTING
Content-Length: 226
Remote-Party-ID: "8531320135" <sip:48103...@xx.xx.xx.xx>;party=calling;screen=yes;privacy=off
2017-04-11 09:25:01.870443 [DEBUG] switch_ivr_originate.c:67 (sofia/default/4444213772368578) State Change CS_ROUTING -> CS_CONSUME_MEDIA
v=0
o=FreeSWITCH 1491874313 1491874314 IN IP4 xx.xx.xx.xx
2017-04-11 09:25:01.870443 [DEBUG] switch_core_state_machine.c:602 (sofia/default/4444213772368578) State ROUTING going to sleep
s=FreeSWITCH
c=IN IP4 xxx.xx.xx.xx
t=0 0
m=audio 21188 RTP/AVP 18 101
2017-04-11 09:25:01.870443 [DEBUG] switch_core_state_machine.c:543 (sofia/default/4444213772368578) Running State Change CS_CONSUME_MEDIA
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Gateway :
o=FreeSWITCH 1491874313 1491874314 IN IP4 xx.xx.xx.xx
s=FreeSWITCH
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 21188 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (16 headers 10 lines) ---
Sending to ip.xx.xx.xx : 5060 (no NAT)
Using INVITE request as basis request - d13162bc-992a-1235-bf82-6c0b84deb0be
Found peer 'xx.xx.xx.xx'
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port xx.xx.xx.xx:21188
Looking for 4444213772368578 in termination (domain xx.xx.xx.xx)
list_route: hop: <sip:gw+...@xx.xx.xx.xx:5060;transport=udp;gw=ALG>
Really anoying, as you see the zoiper softphone in G729 , ASTPP G729 and Gateway G729
Samir Doshi
Apr 12, 2017, 3:54:07 AM4/12/17
to ASTPP
Please send full fs_cli log by enabling sip log.
Best Regards
--
Samir Doshi
iNextrix Technologies Pvt. Ltd.
http://www.inextrix.com
Disclaimer:
The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it.
--
Samir Doshi
iNextrix Technologies Pvt. Ltd.
http://www.inextrix.com
Disclaimer:
The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it.
--
=====================================================================
Documentation : https://astppdoc.atlassian.net/
Please contact at sa...@inextrix.com for commercial support.
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Gilbert Arias
Apr 12, 2017, 7:11:35 AM4/12/17
to ASTPP
the problem here is that the codec is only for passthrough as they want you to get licences to get it for full working state, FS can translate g729 but not work over it.
To view this discussion on the web visit https://groups.google.com/d/msgid/astpp/CAFABgc3Df%3DN%2Bz%2BagOkMDu2rK4QXOODu324uF8C8%2B6OmBO-WfCA%40mail.gmail.com.
Gilbert N. Arias Feliz
CEO @ Treblig Web Design & IT Solutions
CEO @ Treblig Web Design & IT Solutions
Unix/Linux Administrator
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VoIP Engineer
BackTrack Network Security Tester Team
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ZeuS
Apr 12, 2017, 10:32:16 AM4/12/17
to ASTPP
Here you have full log
---
---
send 1115 bytes to udp/[IP.IP.IP.IP]:5060 at 16:28:19.444121:
------------------------------------------------------------------------
INVITE sip:44442137...@IP.IP.IP.IP SIP/2.0
Via: SIP/2.0/UDP IP.IP.IP.IP;rport;branch=z9hG4bKBBpK6mpj8rg5S
Max-Forwards: 69
From: "8531320135" <sip:48103...@IP.IP.IP.IP>;tag=H158ty9HvUyrF
To: <sip:44442137...@IP.IP.IP.IP>
Call-ID: 1dbac104-9a2f-1235-48b1-6c0b84deb0be
CSeq: 105667585 INVITE
Contact: <sip:gw+...@IP.IP.IP.IP:5060;transport=udp;gw=ALG>
User-Agent: ASTPP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 226
Remote-Party-ID: "8531320135" <sip:48103...@IP.IP.IP.IP>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1491989539 1491989540 IN IP4 IP.IP.IP.IP
s=FreeSWITCH
c=IN IP4 IP.IP.IP.IP
t=0 0
m=audio 17760 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
------------------------------------------------------------------------
2017-04-12 16:28:19.430430 [DEBUG] sofia.c:6858 Channel sofia/default/4444213771254587 entering state [calling][0]
recv 490 bytes from udp/[IP.IP.IP.IP]:5060 at 16:28:19.448018:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP.IP.IP.IP;branch=z9hG4bKBBpK6mpj8rg5S;received=IP.IP.IP.IP;rport=5060
From: "8531320135" <sip:48103...@IP.IP.IP.IP>;tag=H158ty9HvUyrF
To: <sip:44442137...@IP.IP.IP.IP>
Call-ID: 1dbac104-9a2f-1235-48b1-6c0b84deb0be
CSeq: 105667585 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:44442137...@IP.IP.IP.IP>
Content-Length: 0
------------------------------------------------------------------------
recv 772 bytes from udp/[IP.IP.IP.IP]:5060 at 16:28:19.473815:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP.IP.IP.IP;branch=z9hG4bKBBpK6mpj8rg5S;received=IP.IP.IP.IP;rport=5060
From: "8531320135" <sip:48103...@IP.IP.IP.IP>;tag=H158ty9HvUyrF
To: <sip:44442137...@IP.IP.IP.IP>;tag=as75a0d8ed
Call-ID: 1dbac104-9a2f-1235-48b1-6c0b84deb0be
CSeq: 105667585 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:44442137...@IP.IP.IP.IP>
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 3031 3031 IN IP4 IP.IP.IP.IP
s=session
c=IN IP4 IP.IP.IP.IP
t=0 0
m=audio 25448 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
------------------------------------------------------------------------
2017-04-12 16:28:19.470446 [DEBUG] sofia.c:6858 Channel sofia/default/4444213771254587 entering state [completing][200]
2017-04-12 16:28:19.470446 [DEBUG] sofia.c:6868 Remote SDP:
v=0
o=root 3031 3031 IN IP4 IP.IP.IP.IP
s=session
c=IN IP4 IP.IP.IP.IP
t=0 0
m=audio 25448 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
send 415 bytes to udp/[IP.IP.IP.IP]:5060 at 16:28:19.474542:
------------------------------------------------------------------------
ACK sip:44442137...@IP.IP.IP.IP SIP/2.0
Via: SIP/2.0/UDP IP.IP.IP.IP;rport;branch=z9hG4bKcmFc8F7N516QN
Max-Forwards: 70
From: "8531320135" <sip:48103...@IP.IP.IP.IP>;tag=H158ty9HvUyrF
To: <sip:44442137...@IP.IP.IP.IP>;tag=as75a0d8ed
Call-ID: 1dbac104-9a2f-1235-48b1-6c0b84deb0be
CSeq: 105667585 ACK
Contact: <sip:gw+...@IP.IP.IP.IP:5060;transport=udp;gw=ALG>
Content-Length: 0
------------------------------------------------------------------------
2017-04-12 16:28:19.470446 [DEBUG] sofia.c:6858 Channel sofia/default/4444213771254587 entering state [ready][200]
2017-04-12 16:28:19.470446 [DEBUG] switch_core_media.c:4355 Audio Codec Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1]
2017-04-12 16:28:19.470446 [DEBUG] switch_core_media.c:4410 Audio Codec Compare [G729:18:8000:20:8000:1] ++++ is saved as a match
2017-04-12 16:28:19.470446 [DEBUG] switch_core_media.c:4271 Set telephone-event payload to 101@8000
2017-04-12 16:28:19.470446 [DEBUG] switch_core_media.c:3090 Set Codec sofia/default/4444213771254587 G729/8000 20 ms 160 samples 8000 bits 1 channels
2017-04-12 16:28:19.470446 [DEBUG] switch_core_codec.c:111 sofia/default/4444213771254587 Original read codec set to G729:18
2017-04-12 16:28:19.470446 [DEBUG] switch_core_media.c:4623 Set telephone-event payload to 101@8000
2017-04-12 16:28:19.470446 [DEBUG] switch_core_media.c:4681 sofia/default/4444213771254587 Set 2833 dtmf send payload to 101 recv payload to 101
2017-04-12 16:28:19.470446 [DEBUG] switch_core_media.c:6464 AUDIO RTP [sofia/default/4444213771254587] IP.IP.IP.IP port 17760 -> IP.IP.IP.IP port 25448 codec: 18 ms: 20
2017-04-12 16:28:19.470446 [DEBUG] switch_rtp.c:3832 Starting timer [soft] 160 bytes per 20ms
2017-04-12 16:28:19.490473 [DEBUG] switch_core_media.c:6763 sofia/default/4444213771254587 Set 2833 dtmf send payload to 101
2017-04-12 16:28:19.490473 [DEBUG] switch_core_media.c:6770 sofia/default/4444213771254587 Set 2833 dtmf receive payload to 101
2017-04-12 16:28:19.490473 [DEBUG] switch_core_media.c:6793 sofia/default/4444213771254587 Set rtp dtmf delay to 40
2017-04-12 16:28:19.490473 [NOTICE] sofia.c:7826 Channel [sofia/default/4444213771254587] has been answered
EXECUTE sofia/default/4444213771254587 record_session(/usr/local/freeswitch/recordings/2017-04-12-16:28:19_2457848300.wav)
2017-04-12 16:28:19.490473 [DEBUG] switch_core_media_bug.c:828 Attaching BUG to sofia/default/4444213771254587
2017-04-12 16:28:19.490473 [DEBUG] switch_channel.c:3770 (sofia/default/4444213771254587) Callstate Change DOWN -> ACTIVE
2017-04-12 16:28:19.490473 [DEBUG] switch_core_media.c:6464 AUDIO RTP [sofia/default/85313...@IP.IP.IP.IP] IP.IP.IP.IP port 30076 -> 192.168.1.101 port 38638 codec: 18 ms: 20
2017-04-12 16:28:19.490473 [DEBUG] switch_rtp.c:3832 Starting timer [soft] 160 bytes per 20ms
2017-04-12 16:28:20.550485 [DEBUG] switch_core_media.c:6763 sofia/default/85313...@IP.IP.IP.IP Set 2833 dtmf send payload to 101
2017-04-12 16:28:20.550485 [DEBUG] switch_core_media.c:6770 sofia/default/85313...@IP.IP.IP.IP Set 2833 dtmf receive payload to 101
2017-04-12 16:28:20.550485 [DEBUG] switch_core_media.c:6793 sofia/default/85313...@IP.IP.IP.IP Set rtp dtmf delay to 40
2017-04-12 16:28:20.550485 [DEBUG] mod_sofia.c:814 Local SDP sofia/default/85313...@IP.IP.IP.IP:
v=0
o=FreeSWITCH 1491977223 1491977224 IN IP4 IP.IP.IP.IP
s=FreeSWITCH
c=IN IP4 IP.IP.IP.IP
t=0 0
m=audio 30076 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
send 1042 bytes to udp/[IP.IP.IP.IP]:3326 at 16:28:20.559467:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP.IP.IP.IP:3326;branch=z9hG4bK-524287-1---6b90570a26621eac;rport=3326
From: <sip:85313...@IP.IP.IP.IP;transport=UDP>;tag=b2bc4579
To: <sip:44442137...@IP.IP.IP.IP;transport=UDP>;tag=grcgS3reZj85K
Call-ID: inK5TLVvcuKobvXtQk2h4g..
CSeq: 2 INVITE
Contact: <sip:44442137...@IP.IP.IP.IP:5060;transport=udp>
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
2017-04-12 16:28:20.550485 [NOTICE] switch_ivr_originate.c:3549 Channel [sofia/default/85313...@IP.IP.IP.IP] has been answered
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 247
v=0
o=FreeSWITCH 1491977223 1491977224 IN IP4 IP.IP.IP.IP
s=FreeSWITCH
c=IN IP4 IP.IP.IP.IP
t=0 0
m=audio 30076 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
------------------------------------------------------------------------
2017-04-12 16:28:20.550485 [DEBUG] sofia.c:6858 Channel sofia/default/85313...@IP.IP.IP.IP entering state [completed][200]
EXECUTE sofia/default/85313...@IP.IP.IP.IP record_session(/usr/local/freeswitch/recordings/2017-04-12-16:28:19_2457848300.wav)
2017-04-12 16:28:20.570457 [DEBUG] switch_core_media_bug.c:828 Attaching BUG to sofia/default/85313...@IP.IP.IP.IP
2017-04-12 16:28:20.570457 [DEBUG] switch_channel.c:3770 (sofia/default/85313...@IP.IP.IP.IP) Callstate Change RINGING -> ACTIVE
2017-04-12 16:28:20.570457 [DEBUG] switch_ivr_originate.c:3607 Originate Resulted in Success: [sofia/default/4444213771254587]
2017-04-12 16:28:20.570457 [DEBUG] switch_ivr_bridge.c:1594 (sofia/default/4444213771254587) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2017-04-12 16:28:20.570457 [DEBUG] switch_core_state_machine.c:543 (sofia/default/4444213771254587) Running State Change CS_EXCHANGE_MEDIA
2017-04-12 16:28:20.570457 [DEBUG] switch_core_state_machine.c:612 (sofia/default/4444213771254587) State EXCHANGE_MEDIA
2017-04-12 16:28:20.570457 [DEBUG] mod_sofia.c:628 SOFIA EXCHANGE_MEDIA
2017-04-12 16:28:20.570457 [DEBUG] switch_ivr_async.c:1496 No silence detection configured; assuming start of speech
2017-04-12 16:28:20.570457 [DEBUG] switch_ivr_async.c:1496 No silence detection configured; assuming start of speech
2017-04-12 16:28:20.590472 [DEBUG] switch_rtp.c:6707 Correct audio ip/port confirmed.
2017-04-12 16:28:20.590472 [DEBUG] switch_core_io.c:448 Setting BUG Codec G729:18
2017-04-12 16:28:20.590472 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode!
2017-04-12 16:28:20.590472 [ERR] switch_core_io.c:633 Codec G.729 decoder error! [1]
2017-04-12 16:28:20.590472 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode!
2017-04-12 16:28:20.590472 [ERR] switch_core_io.c:1255 Codec G.729 decoder error!
2017-04-12 16:28:20.590472 [DEBUG] switch_ivr_bridge.c:701 sofia/default/85313...@IP.IP.IP.IP ending bridge by request from write function
2017-04-12 16:28:20.590472 [DEBUG] switch_ivr_bridge.c:780 BRIDGE THREAD DONE [sofia/default/4444213771254587]
2017-04-12 16:28:20.590472 [NOTICE] switch_ivr_bridge.c:884 Hangup sofia/default/4444213771254587 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2017-04-12 16:28:20.590472 [DEBUG] switch_core_state_machine.c:612 (sofia/default/4444213771254587) State EXCHANGE_MEDIA going to sleep
2017-04-12 16:28:20.590472 [DEBUG] switch_core_state_machine.c:543 (sofia/default/4444213771254587) Running State Change CS_HANGUP
2017-04-12 16:28:20.590472 [DEBUG] switch_ivr_async.c:1312 Stop recording file /usr/local/freeswitch/recordings/2017-04-12-16:28:19_2457848300.wav
2017-04-12 16:28:20.590472 [DEBUG] switch_ivr_async.c:1376 Channel is hung up
2017-04-12 16:28:20.590472 [DEBUG] switch_core_media_bug.c:1120 Removing BUG from sofia/default/4444213771254587
2017-04-12 16:28:20.590472 [DEBUG] switch_core_state_machine.c:809 (sofia/default/4444213771254587) Callstate Change ACTIVE -> HANGUP
2017-04-12 16:28:20.590472 [DEBUG] switch_core_state_machine.c:811 (sofia/default/4444213771254587) State HANGUP
2017-04-12 16:28:20.590472 [DEBUG] mod_sofia.c:437 Channel sofia/default/4444213771254587 hanging up, cause: NORMAL_CLEARING
2017-04-12 16:28:20.590472 [DEBUG] mod_sofia.c:490 Sending BYE to sofia/default/4444213771254587
2017-04-12 16:28:20.590472 [DEBUG] switch_core_state_machine.c:60 sofia/default/4444213771254587 Standard HANGUP, cause: NORMAL_CLEARING
2017-04-12 16:28:20.590472 [DEBUG] switch_core_state_machine.c:811 (sofia/default/4444213771254587) State HANGUP going to sleep
2017-04-12 16:28:20.590472 [DEBUG] switch_core_state_machine.c:578 (sofia/default/4444213771254587) State Change CS_HANGUP -> CS_REPORTING
2017-04-12 16:28:20.590472 [DEBUG] switch_core_state_machine.c:543 (sofia/default/4444213771254587) Running State Change CS_REPORTING
send 547 bytes to udp/[IP.IP.IP.IP]:5060 at 16:28:20.597137:
------------------------------------------------------------------------
BYE sip:44442137...@IP.IP.IP.IP SIP/2.0
Via: SIP/2.0/UDP IP.IP.IP.IP;rport;branch=z9hG4bKDX849arS2aXaH
Max-Forwards: 70
From: "8531320135" <sip:48103...@IP.IP.IP.IP>;tag=H158ty9HvUyrF
To: <sip:44442137...@IP.IP.IP.IP>;tag=as75a0d8ed
Call-ID: 1dbac104-9a2f-1235-48b1-6c0b84deb0be
CSeq: 105667586 BYE
User-Agent: ASTPP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
------------------------------------------------------------------------
2017-04-12 16:28:20.590472 [DEBUG] switch_core_state_machine.c:897 (sofia/default/4444213771254587) State REPORTING
2017-04-12 16:28:20.590472 [INFO] mod_json_cdr.c:271 Process [f2ffe341-4758-4577-9388-20fee63f9f4e.cdr.json]
recv 450 bytes from udp/[IP.IP.IP.IP]:5060 at 16:28:20.598080:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP.IP.IP.IP;branch=z9hG4bKDX849arS2aXaH;received=IP.IP.IP.IP;rport=5060
From: "8531320135" <sip:48103...@IP.IP.IP.IP>;tag=H158ty9HvUyrF
To: <sip:44442137...@IP.IP.IP.IP>;tag=as75a0d8ed
Call-ID: 1dbac104-9a2f-1235-48b1-6c0b84deb0be
CSeq: 105667586 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
------------------------------------------------------------------------
2017-04-12 16:28:20.590472 [DEBUG] switch_ivr_bridge.c:780 BRIDGE THREAD DONE [sofia/default/85313...@IP.IP.IP.IP]
2017-04-12 16:28:20.590472 [DEBUG] switch_ivr_bridge.c:1692 sofia/default/4444213771254587 skip receive message [UNBRIDGE] (channel is hungup already)
2017-04-12 16:28:20.590472 [NOTICE] switch_ivr_bridge.c:1744 Hangup sofia/default/85313...@IP.IP.IP.IP [CS_EXECUTE] [NORMAL_CLEARING]
2017-04-12 16:28:20.590472 [DEBUG] switch_core_session.c:2796 sofia/default/85313...@IP.IP.IP.IP skip receive message [PHONE_EVENT] (channel is hungup already)
2017-04-12 16:28:20.590472 [DEBUG] switch_core_session.c:2796 sofia/default/85313...@IP.IP.IP.IP skip receive message [PHONE_EVENT] (channel is hungup already)
2017-04-12 16:28:20.590472 [DEBUG] switch_core_state_machine.c:609 (sofia/default/85313...@IP.IP.IP.IP) State EXECUTE going to sleep
2017-04-12 16:28:20.590472 [DEBUG] switch_core_state_machine.c:543 (sofia/default/85313...@IP.IP.IP.IP) Running State Change CS_HANGUP
2017-04-12 16:28:20.590472 [DEBUG] switch_ivr_async.c:1312 Stop recording file /usr/local/freeswitch/recordings/2017-04-12-16:28:19_2457848300.wav
2017-04-12 16:28:20.610460 [DEBUG] switch_ivr_async.c:1376 Channel is hung up
2017-04-12 16:28:20.610460 [DEBUG] switch_core_media_bug.c:1120 Removing BUG from sofia/default/85313...@IP.IP.IP.IP
2017-04-12 16:28:20.610460 [DEBUG] switch_core_state_machine.c:809 (sofia/default/85313...@IP.IP.IP.IP) Callstate Change ACTIVE -> HANGUP
2017-04-12 16:28:20.610460 [DEBUG] switch_core_state_machine.c:811 (sofia/default/85313...@IP.IP.IP.IP) State HANGUP
2017-04-12 16:28:20.610460 [DEBUG] mod_sofia.c:437 Channel sofia/default/85313...@IP.IP.IP.IP hanging up, cause: NORMAL_CLEARING
2017-04-12 16:28:20.610460 [DEBUG] mod_sofia.c:490 Sending BYE to sofia/default/85313...@IP.IP.IP.IP
2017-04-12 16:28:20.610460 [DEBUG] switch_core_state_machine.c:60 sofia/default/85313...@IP.IP.IP.IP Standard HANGUP, cause: NORMAL_CLEARING
2017-04-12 16:28:20.610460 [DEBUG] switch_core_state_machine.c:811 (sofia/default/85313...@IP.IP.IP.IP) State HANGUP going to sleep
2017-04-12 16:28:20.610460 [DEBUG] switch_core_state_machine.c:578 (sofia/default/85313...@IP.IP.IP.IP) State Change CS_HANGUP -> CS_REPORTING
2017-04-12 16:28:20.610460 [DEBUG] switch_core_state_machine.c:543 (sofia/default/85313...@IP.IP.IP.IP) Running State Change CS_REPORTING
send 560 bytes to udp/[IP.IP.IP.IP]:3326 at 16:28:20.621621:
------------------------------------------------------------------------
BYE sip:85313...@IP.IP.IP.IP:3326;transport=UDP SIP/2.0
Via: SIP/2.0/UDP IP.IP.IP.IP;rport;branch=z9hG4bKe61XB68vZKKXc
Max-Forwards: 70
From: <sip:44442137...@IP.IP.IP.IP;transport=UDP>;tag=grcgS3reZj85K
To: <sip:85313...@IP.IP.IP.IP;transport=UDP>;tag=b2bc4579
Call-ID: inK5TLVvcuKobvXtQk2h4g..
CSeq: 105667586 BYE
User-Agent: ASTPP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
------------------------------------------------------------------------
2017-04-12 16:28:20.610460 [DEBUG] switch_core_state_machine.c:897 (sofia/default/85313...@IP.IP.IP.IP) State REPORTING
2017-04-12 16:28:20.610460 [INFO] mod_json_cdr.c:271 Process [ddb13b21-45ae-4f52-941d-f29f1a0d3c5e.cdr.json]
2017-04-12 16:28:20.670471 [DEBUG] switch_core_state_machine.c:174 sofia/default/4444213771254587 Standard REPORTING, cause: NORMAL_CLEARING
2017-04-12 16:28:20.670471 [DEBUG] switch_core_state_machine.c:897 (sofia/default/4444213771254587) State REPORTING going to sleep
2017-04-12 16:28:20.670471 [DEBUG] switch_core_state_machine.c:569 (sofia/default/4444213771254587) State Change CS_REPORTING -> CS_DESTROY
2017-04-12 16:28:20.670471 [DEBUG] switch_core_session.c:1646 Session 31 (sofia/default/4444213771254587) Locked, Waiting on external entities
2017-04-12 16:28:20.670471 [NOTICE] switch_core_session.c:1664 Session 31 (sofia/default/4444213771254587) Ended
2017-04-12 16:28:20.670471 [NOTICE] switch_core_session.c:1668 Close Channel sofia/default/4444213771254587 [CS_DESTROY]
2017-04-12 16:28:20.670471 [DEBUG] switch_core_state_machine.c:700 (sofia/default/4444213771254587) Running State Change CS_DESTROY
2017-04-12 16:28:20.670471 [DEBUG] switch_core_state_machine.c:710 (sofia/default/4444213771254587) State DESTROY
2017-04-12 16:28:20.670471 [DEBUG] mod_sofia.c:342 sofia/default/4444213771254587 SOFIA DESTROY
2017-04-12 16:28:20.670471 [DEBUG] switch_core_state_machine.c:181 sofia/default/4444213771254587 Standard DESTROY
2017-04-12 16:28:20.670471 [DEBUG] switch_core_state_machine.c:710 (sofia/default/4444213771254587) State DESTROY going to sleep
recv 471 bytes from udp/[IP.IP.IP.IP]:3326 at 16:28:20.699332:
------------------------------------------------------------------------
ACK sip:44442137...@IP.IP.IP.IP:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP IP.IP.IP.IP:3326;branch=z9hG4bK-524287-1---7cbafe7f12fdf1d5;rport
Max-Forwards: 70
Contact: <sip:85313...@IP.IP.IP.IP:3326;transport=UDP>
To: <sip:44442137...@IP.IP.IP.IP;transport=UDP>;tag=grcgS3reZj85K
From: <sip:85313...@IP.IP.IP.IP;transport=UDP>;tag=b2bc4579
Call-ID: inK5TLVvcuKobvXtQk2h4g..
CSeq: 2 ACK
User-Agent: Zoiper rv2.8.30
Content-Length: 0
------------------------------------------------------------------------
recv 395 bytes from udp/[IP.IP.IP.IP]:3326 at 16:28:20.702321:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP.IP.IP.IP;rport=5060;branch=z9hG4bKe61XB68vZKKXc
Contact: <sip:85313...@IP.IP.IP.IP:3326;transport=UDP>
To: <sip:85313...@IP.IP.IP.IP;transport=UDP>;tag=b2bc4579
From: <sip:44442137...@IP.IP.IP.IP;transport=UDP>;tag=grcgS3reZj85K
Call-ID: inK5TLVvcuKobvXtQk2h4g..
CSeq: 105667586 BYE
User-Agent: Zoiper rv2.8.30
Content-Length: 0
------------------------------------------------------------------------
2017-04-12 16:28:20.690479 [DEBUG] switch_core_state_machine.c:174 sofia/default/85313...@IP.IP.IP.IP Standard REPORTING, cause: NORMAL_CLEARING
2017-04-12 16:28:20.690479 [DEBUG] switch_core_state_machine.c:897 (sofia/default/85313...@IP.IP.IP.IP) State REPORTING going to sleep
2017-04-12 16:28:20.690479 [DEBUG] switch_core_state_machine.c:569 (sofia/default/85313...@IP.IP.IP.IP) State Change CS_REPORTING -> CS_DESTROY
2017-04-12 16:28:20.690479 [DEBUG] switch_core_session.c:1646 Session 30 (sofia/default/85313...@IP.IP.IP.IP) Locked, Waiting on external entities
2017-04-12 16:28:20.690479 [NOTICE] switch_core_session.c:1664 Session 30 (sofia/default/85313...@IP.IP.IP.IP) Ended
2017-04-12 16:28:20.690479 [NOTICE] switch_core_session.c:1668 Close Channel sofia/default/85313...@IP.IP.IP.IP [CS_DESTROY]
2017-04-12 16:28:20.690479 [DEBUG] switch_core_state_machine.c:700 (sofia/default/85313...@IP.IP.IP.IP) Running State Change CS_DESTROY
2017-04-12 16:28:20.690479 [DEBUG] switch_core_state_machine.c:710 (sofia/default/85313...@IP.IP.IP.IP) State DESTROY
2017-04-12 16:28:20.690479 [DEBUG] mod_sofia.c:342 sofia/default/85313...@IP.IP.IP.IP SOFIA DESTROY
2017-04-12 16:28:20.690479 [DEBUG] switch_core_state_machine.c:181 sofia/default/85313...@IP.IP.IP.IP Standard DESTROY
2017-04-12 16:28:20.690479 [DEBUG] switch_core_state_machine.c:710 (sofia/default/85313...@IP.IP.IP.IP) State DESTROY going to sleep
2017-04-12 16:28:20.930472 [DEBUG] switch_scheduler.c:144 Deleting task 17 switch_ivr_schedule_hangup (ddb13b21-45ae-4f52-941d-f29f1a0d3c5e)
recv 785 bytes from udp/[IP.IP.IP.IP]:3326 at 16:28:50.657558:
------------------------------------------------------------------------
REGISTER sip:IP.IP.IP.IP;transport=UDP SIP/2.0
Via: SIP/2.0/UDP IP.IP.IP.IP:3326;branch=z9hG4bK-524287-1---54d1696294b87f3b;rport
Max-Forwards: 70
Contact: <sip:85313...@IP.IP.IP.IP:3326;transport=UDP;rinstance=4ade1350a8a6a207>
To: <sip:85313...@IP.IP.IP.IP;transport=UDP>
From: <sip:85313...@IP.IP.IP.IP;transport=UDP>;tag=fc1d847f
Call-ID: 4IIxAm5Sahr66f-MzhCk0w..
CSeq: 13 REGISTER
Expires: 60
User-Agent: Zoiper rv2.8.30
Authorization: Digest username="8531320135",realm="IP.IP.IP.IP",nonce="76e76856-62e3-4ae2-80a6-dad160f0e8bb",uri="sip:IP.IP.IP.IP;transport=UDP",response="924f1c649394c692b0a3d93e43499ff8",cnonce="cc22c11bc4da7b84eb6ef7a0be9f3b0e",nc=00000004,qop=auth,algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
------------------------------------------------------------------------
recv 785 bytes from udp/[IP.IP.IP.IP]:3326 at 16:28:51.160541:
------------------------------------------------------------------------
REGISTER sip:IP.IP.IP.IP;transport=UDP SIP/2.0
Via: SIP/2.0/UDP IP.IP.IP.IP:3326;branch=z9hG4bK-524287-1---54d1696294b87f3b;rport
Max-Forwards: 70
Contact: <sip:85313...@IP.IP.IP.IP:3326;transport=UDP;rinstance=4ade1350a8a6a207>
To: <sip:85313...@IP.IP.IP.IP;transport=UDP>
From: <sip:85313...@IP.IP.IP.IP;transport=UDP>;tag=fc1d847f
Call-ID: 4IIxAm5Sahr66f-MzhCk0w..
CSeq: 13 REGISTER
Expires: 60
User-Agent: Zoiper rv2.8.30
Authorization: Digest username="8531320135",realm="IP.IP.IP.IP",nonce="76e76856-62e3-4ae2-80a6-dad160f0e8bb",uri="sip:IP.IP.IP.IP;transport=UDP",response="924f1c649394c692b0a3d93e43499ff8",cnonce="cc22c11bc4da7b84eb6ef7a0be9f3b0e",nc=00000004,qop=auth,algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
------------------------------------------------------------------------
recv 785 bytes from udp/[IP.IP.IP.IP]:3326 at 16:28:52.162759:
------------------------------------------------------------------------
REGISTER sip:IP.IP.IP.IP;transport=UDP SIP/2.0
Via: SIP/2.0/UDP IP.IP.IP.IP:3326;branch=z9hG4bK-524287-1---54d1696294b87f3b;rport
Max-Forwards: 70
Contact: <sip:85313...@IP.IP.IP.IP:3326;transport=UDP;rinstance=4ade1350a8a6a207>
To: <sip:85313...@IP.IP.IP.IP;transport=UDP>
From: <sip:85313...@IP.IP.IP.IP;transport=UDP>;tag=fc1d847f
Call-ID: 4IIxAm5Sahr66f-MzhCk0w..
CSeq: 13 REGISTER
Expires: 60
User-Agent: Zoiper rv2.8.30
Authorization: Digest username="8531320135",realm="IP.IP.IP.IP",nonce="76e76856-62e3-4ae2-80a6-dad160f0e8bb",uri="sip:IP.IP.IP.IP;transport=UDP",response="924f1c649394c692b0a3d93e43499ff8",cnonce="cc22c11bc4da7b84eb6ef7a0be9f3b0e",nc=00000004,qop=auth,algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
------------------------------------------------------------------------
recv 785 bytes from udp/[IP.IP.IP.IP]:3326 at 16:28:54.163268:
------------------------------------------------------------------------
REGISTER sip:IP.IP.IP.IP;transport=UDP SIP/2.0
Via: SIP/2.0/UDP IP.IP.IP.IP:3326;branch=z9hG4bK-524287-1---54d1696294b87f3b;rport
Max-Forwards: 70
Contact: <sip:85313...@IP.IP.IP.IP:3326;transport=UDP;rinstance=4ade1350a8a6a207>
To: <sip:85313...@IP.IP.IP.IP;transport=UDP>
From: <sip:85313...@IP.IP.IP.IP;transport=UDP>;tag=fc1d847f
Call-ID: 4IIxAm5Sahr66f-MzhCk0w..
CSeq: 13 REGISTER
Expires: 60
User-Agent: Zoiper rv2.8.30
Authorization: Digest username="8531320135",realm="IP.IP.IP.IP",nonce="76e76856-62e3-4ae2-80a6-dad160f0e8bb",uri="sip:IP.IP.IP.IP;transport=UDP",response="924f1c649394c692b0a3d93e43499ff8",cnonce="cc22c11bc4da7b84eb6ef7a0be9f3b0e",nc=00000004,qop=auth,algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
------------------------------------------------------------------------
send 601 bytes to udp/[IP.IP.IP.IP]:3326 at 16:28:54.245487:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP.IP.IP.IP:3326;branch=z9hG4bK-524287-1---54d1696294b87f3b;rport=3326
From: <sip:85313...@IP.IP.IP.IP;transport=UDP>;tag=fc1d847f
To: <sip:85313...@IP.IP.IP.IP;transport=UDP>;tag=jaZ1vStNS4mBB
Call-ID: 4IIxAm5Sahr66f-MzhCk0w..
CSeq: 13 REGISTER
Contact: <sip:85313...@IP.IP.IP.IP:3326;transport=UDP;rinstance=4ade1350a8a6a207>;expires=60
Date: Wed, 12 Apr 2017 14:28:54 GMT
User-Agent: ASTPP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Content-Length: 0
------------------------------------------------------------------------
Saiful Alam
Apr 12, 2017, 2:35:53 PM4/12/17
to ASTPP
Try disabling "Allow Call Recording" for both customer account and provider account. Reload or restart freeswitch and try again. From your log i can see that call is trying to be recorded. G729 call cannot be recorded without license.
ZeuS
Apr 12, 2017, 5:52:07 PM4/12/17
to ASTPP
Thank you so much Alam, ITS WORK!, it was Allow Recording "yes" in customer and provider account settings , now disabled and its work perfectly
Saiful Alam
Apr 12, 2017, 8:16:13 PM4/12/17
to ASTPP
Requesting developers to turn off "Allow call recording" by default when a new user is created.
Right now when you create a new account, "Allow call recording" is Yes by default.
It becomes one more work for the admin to disable it everytime creating a new account.
Gilbert Arias
Apr 12, 2017, 9:41:14 PM4/12/17
to as...@googlegroups.com
I recommend you to implement this version is Open free and easy to setup
Was unable to post it early
--
=====================================================================
Documentation : https://astppdoc.atlassian.net/
Please contact at sa...@inextrix.com for commercial support.
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ZeuS
Apr 13, 2017, 6:38:04 AM4/13/17
to ASTPP
Thank you Garias, and Alam right, this option should be disable by default
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