Calls drops when answered

Moderator: areski

Calls drops when answered

Postby caronte » Thu Aug 09, 2012 5:46 pm

First, congratulations for this great piece of code, it has a lot of potential and the best is that it is open source. As me, many people are saying thank you.

I installed newfies on a testing machine (following your procedure) and look like it runs ok, for outgoing calls I am using an account on an a2billing server.

I did created a campaing, newfies start it and the log says that it goes as expected, newfies make the call but before it can be answered it disconnect that call, so the remote contact notice the phone ring ones and drop.

I installed newfies on both recommended versions of Ubuntu. I have been trying to troubleshoot myself but now, look like I need some help.

The interesting thing is that looking at the sip a message, newfies is sending a normal disconnect, creating a “normal clearing”message, in other hand, newfies logs shows that the procedure continues as normal (playing the audio file).

If somebody can give an idea I will appreciate it.

This is what I have:

192.168.6.29 <--- newfies IP
a.b.c.d <--- privider IP
23456789 <--- contact to dial


freeswitch@internal> sofia status
Name Type Data State
=================================================================================================
external profile sip:mod_sofia@192.168.6.29:5080 RUNNING (0)
external::example.com gateway sip:joeuser@example.com NOREG
external::a.b.c.d gateway sip:<my_username>@a.b.c.d REGED
192.168.6.29 alias internal ALIASED
internal profile sip:mod_sofia@192.168.6.29:5060 RUNNING (0)
internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0)
=================================================================================================
3 profiles 1 alias


root@newfies:/etc/freeswitch/sip_profiles/external# cat a2billing.xml
<include>
<gateway name="a.b.c.d">
<!--/// account username *required* ///-->
<param name="username" value="my_username"/>
<!--/// auth realm: *optional* same as gateway name, if blank ///-->

<param name="realm" value=""/>
<!--/// username to use in from: *optional* same as username, if blank ///-->
<param name="from-user" value="my_username"/>
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
<!--param name="from-domain" value=""/-->
<!--/// account password *required* ///-->
<param name="password" value="my_password"/>
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
<!--<param name="extension" value="cluecon"/>-->
<!--/// proxy host: *optional* same as realm, if blank ///-->
<!--<param name="proxy" value="asterlink.com"/>-->
<!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
<!--<param name="register-proxy" value="mysbc.com"/>-->
<!--/// expire in seconds: *optional* 3600, if blank ///-->
<!--<param name="expire-seconds" value="60"/>-->
<!--/// do not register ///-->
<param name="register" value="true"/>
<!-- which transport to use for register -->
<!--<param name="register-transport" value="udp"/>-->
<!--How many seconds before a retry when a failure or timeout occurs -->
<!--<param name="retry-seconds" value="30"/>-->
<!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
<!--<param name="caller-id-in-from" value="false"/>-->
<!--extra sip params to send in the contact-->
<!--<param name="contact-params" value="tport=tcp"/>-->
<!--send an options ping every x seconds, failure will unregister and/or mark it down-->
<!--<param name="ping" value="25"/>-->
</gateway>
</include>

--------------------------------------------------------------------------------------------------------------------

Newfies Debug

freeswitch@internal>
2012-08-09 07:23:39.338705 [DEBUG] switch_ivr_originate.c:1947 Parsing global variables
2012-08-09 07:23:39.338705 [DEBUG] switch_event.c:1527 Parsing variable [plivo_request_uuid]=[6e812780-e225-11e1-af74-000c29168f72]
2012-08-09 07:23:39.338705 [DEBUG] switch_event.c:1527 Parsing variable [plivo_answer_url]=[http://192.168.6.29:8008/api/v1/answercall/]
2012-08-09 07:23:39.338705 [DEBUG] switch_event.c:1527 Parsing variable [plivo_hangup_url]=[http://192.168.6.29:8008/api/v1/hangupcall/]
2012-08-09 07:23:39.338705 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_number]=[55555555]
2012-08-09 07:23:39.338705 [DEBUG] switch_event.c:1527 Parsing variable [bridge_early_media]=[true]
2012-08-09 07:23:39.338705 [DEBUG] switch_event.c:1527 Parsing variable [hangup_after_bridge]=[true]
2012-08-09 07:23:39.338705 [DEBUG] switch_event.c:1527 Parsing variable [plivo_from]=[55555555]
2012-08-09 07:23:39.338705 [DEBUG] switch_event.c:1527 Parsing variable [plivo_to]=[23456789]
2012-08-09 07:23:39.338705 [DEBUG] switch_event.c:1527 Parsing variable [plivo_app]=[true]
2012-08-09 07:23:39.338705 [DEBUG] switch_event.c:1527 Parsing variable [absolute_codec_string]=[PCMA]
2012-08-09 07:23:39.338705 [DEBUG] switch_event.c:1527 Parsing variable [originate_timeout]=[1800]
2012-08-09 07:23:39.338705 [DEBUG] switch_event.c:1527 Parsing variable [ignore_early_media]=[true]
2012-08-09 07:23:39.338705 [NOTICE] switch_channel.c:926 New Channel sofia/external/23456789 [6e81faca-e225-11e1-97af-01e5f63e041b]
2012-08-09 07:23:39.338705 [DEBUG] mod_sofia.c:4709 (sofia/external/23456789) State Change CS_NEW -> CS_INIT
2012-08-09 07:23:39.338705 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.338705 [DEBUG] switch_core_state_machine.c:382 (sofia/external/23456789) Running State Change CS_INIT
2012-08-09 07:23:39.338705 [DEBUG] switch_core_state_machine.c:421 (sofia/external/23456789) State INIT
2012-08-09 07:23:39.338705 [DEBUG] mod_sofia.c:85 sofia/external/23456789 SOFIA INIT
2012-08-09 07:23:39.338705 [DEBUG] switch_core_session.c:919 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.338705 [DEBUG] mod_sofia.c:129 (sofia/external/23456789) State Change CS_INIT -> CS_ROUTING
2012-08-09 07:23:39.338705 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.338705 [DEBUG] switch_core_state_machine.c:421 (sofia/external/23456789) State INIT going to sleep
2012-08-09 07:23:39.338705 [DEBUG] switch_core_state_machine.c:382 (sofia/external/23456789) Running State Change CS_ROUTING
2012-08-09 07:23:39.338705 [DEBUG] switch_channel.c:1887 (sofia/external/23456789) Callstate Change DOWN -> RINGING
2012-08-09 07:23:39.338705 [DEBUG] switch_core_state_machine.c:430 (sofia/external/23456789) State ROUTING
2012-08-09 07:23:39.338705 [DEBUG] mod_sofia.c:152 sofia/external/23456789 SOFIA ROUTING
2012-08-09 07:23:39.338705 [DEBUG] switch_ivr_originate.c:67 (sofia/external/23456789) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2012-08-09 07:23:39.338705 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.338705 [DEBUG] switch_core_state_machine.c:430 (sofia/external/23456789) State ROUTING going to sleep
2012-08-09 07:23:39.338705 [DEBUG] switch_core_state_machine.c:382 (sofia/external/23456789) Running State Change CS_CONSUME_MEDIA
2012-08-09 07:23:39.338705 [DEBUG] switch_core_state_machine.c:449 (sofia/external/23456789) State CONSUME_MEDIA
2012-08-09 07:23:39.338705 [DEBUG] switch_core_session.c:919 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.338705 [DEBUG] switch_core_state_machine.c:449 (sofia/external/23456789) State CONSUME_MEDIA going to sleep
2012-08-09 07:23:39.358672 [DEBUG] sofia.c:5677 Channel sofia/external/23456789 entering state [calling][0]
2012-08-09 07:23:39.358672 [DEBUG] switch_core_session.c:919 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.358672 [DEBUG] sofia.c:5677 Channel sofia/external/23456789 entering state [calling][0]
2012-08-09 07:23:39.438687 [DEBUG] switch_core_session.c:919 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.438687 [DEBUG] switch_core_session.c:919 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.438687 [DEBUG] sofia.c:5677 Channel sofia/external/23456789 entering state [completing][200]
2012-08-09 07:23:39.438687 [DEBUG] sofia.c:5688 Remote SDP:
v=0
o=root 2023549897 2023549897 IN IP4 a.b.c.d
s=Asterisk PBX 1.8.11-cert1
c=IN IP4 a.b.c.d
t=0 0
m=audio 14536 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2012-08-09 07:23:39.438687 [DEBUG] switch_core_session.c:919 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.438687 [DEBUG] switch_core_session.c:919 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.438687 [DEBUG] sofia.c:5677 Channel sofia/external/23456789 entering state [ready][200]
2012-08-09 07:23:39.438687 [DEBUG] sofia_glue.c:4911 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2012-08-09 07:23:39.438687 [DEBUG] sofia_glue.c:2995 Set Codec sofia/external/23456789 PCMA/8000 20 ms 160 samples 64000 bits
2012-08-09 07:23:39.438687 [DEBUG] switch_core_codec.c:111 sofia/external/23456789 Original read codec set to PCMA:8
2012-08-09 07:23:39.438687 [DEBUG] sofia_glue.c:5025 Set 2833 dtmf send payload to 101
2012-08-09 07:23:39.438687 [DEBUG] sofia_glue.c:3244 AUDIO RTP [sofia/external/23456789] 192.168.6.29 port 32712 -> a.b.c.d port

14536 codec: 8 ms: 20
2012-08-09 07:23:39.438687 [DEBUG] switch_rtp.c:1676 Starting timer [soft] 160 bytes per 20ms
2012-08-09 07:23:39.438687 [DEBUG] sofia_glue.c:3508 Set 2833 dtmf send payload to 101
2012-08-09 07:23:39.438687 [DEBUG] sofia_glue.c:3514 Set 2833 dtmf receive payload to 101
2012-08-09 07:23:39.438687 [DEBUG] sofia_glue.c:3541 sofia/external/23456789 Set rtp dtmf delay to 40
2012-08-09 07:23:39.438687 [DEBUG] switch_channel.c:3245 (sofia/external/23456789) Callstate Change RINGING -> ACTIVE
2012-08-09 07:23:39.438687 [DEBUG] switch_ivr_originate.c:3330 Originate Resulted in Success: [sofia/external/23456789]
2012-08-09 07:23:39.438687 [INFO] switch_channel.c:2711 sofia/external/23456789 Flipping CID from "" <55555555> to "Outbound Call"

<23456789>
2012-08-09 07:23:39.438687 [DEBUG] mod_commands.c:3577 (sofia/external/23456789) State Change CS_CONSUME_MEDIA -> CS_EXECUTE
2012-08-09 07:23:39.438687 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.438687 [NOTICE] sofia.c:6413 Channel [sofia/external/23456789] has been answered
2012-08-09 07:23:39.438687 [DEBUG] switch_core_state_machine.c:382 (sofia/external/23456789) Running State Change CS_EXECUTE
2012-08-09 07:23:39.438687 [DEBUG] switch_core_state_machine.c:437 (sofia/external/23456789) State EXECUTE
2012-08-09 07:23:39.438687 [DEBUG] mod_sofia.c:245 sofia/external/23456789 SOFIA EXECUTE
2012-08-09 07:23:39.438687 [DEBUG] switch_core_state_machine.c:193 sofia/external/23456789 Standard EXECUTE
EXECUTE sofia/external/23456789 socket(127.0.0.1:8084 async full)
2012-08-09 07:23:39.459698 [DEBUG] switch_core_session.c:1056 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.459698 [DEBUG] switch_core_session.c:1056 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.459698 [DEBUG] switch_core_session.c:1056 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:39.499684 [DEBUG] switch_ivr.c:598 sofia/external/23456789 Command Execute set(plivo_app=true)
EXECUTE sofia/external/23456789 set(plivo_app=true)
2012-08-09 07:23:39.499684 [DEBUG] mod_dptools.c:1294 sofia/external/23456789 SET [plivo_app]=[true]
2012-08-09 07:23:39.499684 [DEBUG] switch_ivr.c:598 sofia/external/23456789 Command Execute set(hangup_after_bridge=false)
EXECUTE sofia/external/23456789 set(hangup_after_bridge=false)
2012-08-09 07:23:39.499684 [DEBUG] mod_dptools.c:1294 sofia/external/23456789 SET [hangup_after_bridge]=[false]
2012-08-09 07:23:39.538672 [DEBUG] switch_ivr.c:598 sofia/external/23456789 Command Execute speak(flite|slt|Hello World)
EXECUTE sofia/external/23456789 speak(flite|slt|Hello World)
2012-08-09 07:23:39.538672 [DEBUG] switch_ivr_play_say.c:2473 OPEN TTS flite
2012-08-09 07:23:39.538672 [DEBUG] switch_ivr_play_say.c:2482 Raw Codec Activated
2012-08-09 07:23:39.618665 [DEBUG] switch_ivr_play_say.c:2164 Speaking text: Hello World
2012-08-09 07:23:39.858673 [DEBUG] switch_rtp.c:3252 Correct ip/port confirmed.
2012-08-09 07:23:41.258674 [DEBUG] switch_ivr_play_say.c:2361 done speaking text
2012-08-09 07:23:41.278678 [DEBUG] switch_core_session.c:1056 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:41.278678 [DEBUG] mod_event_socket.c:2644 (sofia/external/23456789) State Change CS_EXECUTE -> CS_RESET
2012-08-09 07:23:41.278678 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:41.298679 [DEBUG] switch_ivr.c:598 sofia/external/23456789 Command Execute hangup()
EXECUTE sofia/external/23456789 hangup()
2012-08-09 07:23:41.298679 [DEBUG] switch_channel.c:2849 (sofia/external/23456789) Callstate Change ACTIVE -> HANGUP
2012-08-09 07:23:41.298679 [NOTICE] mod_dptools.c:1134 Hangup sofia/external/23456789 [CS_RESET] [NORMAL_CLEARING]
2012-08-09 07:23:41.298679 [DEBUG] switch_channel.c:2872 Send signal sofia/external/23456789 [KILL]
2012-08-09 07:23:41.298679 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:41.298679 [DEBUG] switch_core_session.c:2329 sofia/external/23456789 skip receive message [APPLICATION_EXEC_COMPLETE]

(channel is hungup already)
2012-08-09 07:23:41.298679 [DEBUG] switch_core_session.c:2329 sofia/external/23456789 skip receive message [APPLICATION_EXEC_COMPLETE]

(channel is hungup already)
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:437 (sofia/external/23456789) State EXECUTE going to sleep
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:382 (sofia/external/23456789) Running State Change CS_HANGUP
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:622 (sofia/external/23456789) State HANGUP
2012-08-09 07:23:41.298679 [DEBUG] mod_sofia.c:473 Channel sofia/external/23456789 hanging up, cause: NORMAL_CLEARING
2012-08-09 07:23:41.298679 [DEBUG] mod_sofia.c:521 Sending BYE to sofia/external/23456789
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:47 sofia/external/23456789 Standard HANGUP, cause: NORMAL_CLEARING
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:622 (sofia/external/23456789) State HANGUP going to sleep
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:413 (sofia/external/23456789) State Change CS_HANGUP -> CS_REPORTING
2012-08-09 07:23:41.298679 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:382 (sofia/external/23456789) Running State Change CS_REPORTING
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:682 (sofia/external/23456789) State REPORTING
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:79 sofia/external/23456789 Standard REPORTING, cause: NORMAL_CLEARING
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:682 (sofia/external/23456789) State REPORTING going to sleep
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:407 (sofia/external/23456789) State Change CS_REPORTING -> CS_DESTROY
2012-08-09 07:23:41.298679 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/23456789 [BREAK]
2012-08-09 07:23:41.298679 [DEBUG] switch_core_session.c:1424 Session 17 (sofia/external/23456789) Locked, Waiting on external entities
2012-08-09 07:23:41.298679 [NOTICE] switch_core_session.c:1442 Session 17 (sofia/external/23456789) Ended
2012-08-09 07:23:41.298679 [NOTICE] switch_core_session.c:1444 Close Channel sofia/external/23456789 [CS_DESTROY]
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:511 (sofia/external/23456789) Callstate Change HANGUP -> DOWN
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:514 (sofia/external/23456789) Running State Change CS_DESTROY
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:524 (sofia/external/23456789) State DESTROY
2012-08-09 07:23:41.298679 [DEBUG] mod_sofia.c:378 sofia/external/23456789 SOFIA DESTROY
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:86 sofia/external/23456789 Standard DESTROY
2012-08-09 07:23:41.298679 [DEBUG] switch_core_state_machine.c:524 (sofia/external/23456789) State DESTROY going to sleep
freeswitch@internal>


--------------------------------------------------------------------------------------------------------------------

SIP Messages at Source (newfies)


#
U 192.168.6.29:5080 -> a.b.c.d:5060
INVITE sip:23456789@a.b.c.d SIP/2.0.
Via: SIP/2.0/UDP 192.168.6.29:5080;rport;branch=z9hG4bKS9HZv7agQD81S.
Max-Forwards: 70.
From: "" <sip:my_username@a.b.c.d>;tag=je2HZ99F3jDUj.
To: <sip:23456789@a.b.c.d>.
Call-ID: 45e0e545-5cc8-1230-349a-000c29168f72.
CSeq: 31923245 INVITE.
Contact: <sip:gw+a.b.c.d@192.168.6.29:5080;transport=udp;gw=a.b.c.d>.
User-Agent: FreeSWITCH-mod_sofia/1.2.0-.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 201.
X-FS-Support: update_display,send_info.
Remote-Party-ID: <sip:55555555@a.b.c.d>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1344485907 1344485908 IN IP4 192.168.6.29.
s=FreeSWITCH.
c=IN IP4 192.168.6.29.
t=0 0.
m=audio 32712 RTP/AVP 8 101 13.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U a.b.c.d:5060 -> 192.168.6.29:5080
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 192.168.6.29:5080;branch=z9hG4bKS9HZv7agQD81S;received=192.168.6.29;rport=5080.
From: "" <sip:my_username@a.b.c.d>;tag=je2HZ99F3jDUj.
To: <sip:23456789@a.b.c.d>;tag=as744b785f.
Call-ID: 45e0e545-5cc8-1230-349a-000c29168f72.
CSeq: 31923245 INVITE.
Server: Asterisk PBX 1.8.11-cert1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6faac799".
Content-Length: 0.
.

#
U 192.168.6.29:5080 -> a.b.c.d:5060
ACK sip:23456789@a.b.c.d SIP/2.0.
Via: SIP/2.0/UDP 192.168.6.29:5080;rport;branch=z9hG4bKS9HZv7agQD81S.
Max-Forwards: 70.
From: "" <sip:my_username@a.b.c.d>;tag=je2HZ99F3jDUj.
To: <sip:23456789@a.b.c.d>;tag=as744b785f.
Call-ID: 45e0e545-5cc8-1230-349a-000c29168f72.
CSeq: 31923245 ACK.
Content-Length: 0.
.

#
U 192.168.6.29:5080 -> a.b.c.d:5060
INVITE sip:23456789@a.b.c.d SIP/2.0.
Via: SIP/2.0/UDP 192.168.6.29:5080;rport;branch=z9hG4bKtjBry2UKmpymN.
Max-Forwards: 70.
From: "" <sip:my_username@a.b.c.d>;tag=je2HZ99F3jDUj.
To: <sip:23456789@a.b.c.d>.
Call-ID: 45e0e545-5cc8-1230-349a-000c29168f72.
CSeq: 31923246 INVITE.
Contact: <sip:gw+a.b.c.d@192.168.6.29:5080;transport=udp;gw=a.b.c.d>.
User-Agent: FreeSWITCH-mod_sofia/1.2.0-.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, refer.
Authorization: Digest username="my_username", realm="asterisk", nonce="6faac799", algorithm=MD5, uri="sip:23456789@a.b.c.d",

response="a4fdcd5ac0e467d10ea74793e2419ba3".
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 201.
X-FS-Support: update_display,send_info.
Remote-Party-ID: <sip:55555555@a.b.c.d>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1344485907 1344485908 IN IP4 192.168.6.29.
s=FreeSWITCH.
c=IN IP4 192.168.6.29.
t=0 0.
m=audio 32712 RTP/AVP 8 101 13.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U a.b.c.d:5060 -> 192.168.6.29:5080
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.6.29:5080;branch=z9hG4bKtjBry2UKmpymN;received=192.168.6.29;rport=5080.
From: "" <sip:my_username@a.b.c.d>;tag=je2HZ99F3jDUj.
To: <sip:23456789@a.b.c.d>.
Call-ID: 45e0e545-5cc8-1230-349a-000c29168f72.
CSeq: 31923246 INVITE.
Server: Asterisk PBX 1.8.11-cert1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:23456789@a.b.c.d:5060>.
Content-Length: 0.
.

#
U a.b.c.d:5060 -> 192.168.6.29:5080
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.6.29:5080;branch=z9hG4bKtjBry2UKmpymN;received=192.168.6.29;rport=5080.
From: "" <sip:my_username@a.b.c.d>;tag=je2HZ99F3jDUj.
To: <sip:23456789@a.b.c.d>;tag=as2897d1d3.
Call-ID: 45e0e545-5cc8-1230-349a-000c29168f72.
CSeq: 31923246 INVITE.
Server: Asterisk PBX 1.8.11-cert1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:23456789@a.b.c.d:5060>.
Content-Type: application/sdp.
Content-Length: 243.
.
v=0.
o=root 2023549897 2023549897 IN IP4 a.b.c.d.
s=Asterisk PBX 1.8.11-cert1.
c=IN IP4 a.b.c.d.
t=0 0.
m=audio 14536 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.

#
U 192.168.6.29:5080 -> a.b.c.d:5060
ACK sip:23456789@a.b.c.d:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.6.29:5080;rport;branch=z9hG4bKUU4g0XcQHZm7g.
Max-Forwards: 70.
From: "" <sip:my_username@a.b.c.d>;tag=je2HZ99F3jDUj.
To: <sip:23456789@a.b.c.d>;tag=as2897d1d3.
Call-ID: 45e0e545-5cc8-1230-349a-000c29168f72.
CSeq: 31923246 ACK.
Contact: <sip:gw+a.b.c.d@192.168.6.29:5080;transport=udp;gw=a.b.c.d>.
Authorization: Digest username="my_username", realm="asterisk", nonce="6faac799", algorithm=MD5, uri="sip:23456789@a.b.c.d",

response="a4fdcd5ac0e467d10ea74793e2419ba3".
Content-Length: 0.
.

#
U 192.168.6.29:5080 -> a.b.c.d:5060
BYE sip:23456789@a.b.c.d:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.6.29:5080;rport;branch=z9hG4bKv4X91rXte8atc.
Max-Forwards: 70.
From: "" <sip:my_username@a.b.c.d>;tag=je2HZ99F3jDUj.
To: <sip:23456789@a.b.c.d>;tag=as2897d1d3.
Call-ID: 45e0e545-5cc8-1230-349a-000c29168f72.
CSeq: 31923247 BYE.
Contact: <sip:gw+a.b.c.d@192.168.6.29:5080;transport=udp;gw=a.b.c.d>.
User-Agent: FreeSWITCH-mod_sofia/1.2.0-.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Authorization: Digest username="my_username", realm="asterisk", nonce="6faac799", algorithm=MD5, uri="sip:23456789@a.b.c.d:5060",

response="78e559f1b0387370d5d0a8f9cdc53db8".
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
.

#
U a.b.c.d:5060 -> 192.168.6.29:5080
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.6.29:5080;branch=z9hG4bKv4X91rXte8atc;received=192.168.6.29;rport=5080.
From: "" <sip:my_username@a.b.c.d>;tag=je2HZ99F3jDUj.
To: <sip:23456789@a.b.c.d>;tag=as2897d1d3.
Call-ID: 45e0e545-5cc8-1230-349a-000c29168f72.
CSeq: 31923247 BYE.
Server: Asterisk PBX 1.8.11-cert1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Length: 0.
.
caronte
 
Posts: 4
Joined: Wed Aug 08, 2012 8:28 pm

Re: Calls drops when answered

Postby areski » Fri Aug 10, 2012 9:39 am

I would advice you to look at the plivo logs.
http://www.newfies-dialer.org/docs/en/l ... oting.html
areski
Site Admin
 
Posts: 302
Joined: Tue Oct 18, 2011 5:28 pm

Re: Calls drops when answered

Postby caronte » Sat Aug 11, 2012 12:01 am

Following your advice, I did review the logs but couldn’t find anything.

Do you thing that it can be something with freeswitch?

I am attaching the log.

Thanks in advanced.
Attachments
fs_cli-and-plivo-logs.rar
logs
(7.36 KiB) Downloaded 746 times
caronte
 
Posts: 4
Joined: Wed Aug 08, 2012 8:28 pm

Re: Calls drops when answered

Postby areski » Sat Aug 11, 2012 1:17 am

From your Freeswitch logs we can see there is a message being played :

2012-08-10 17:02:29.361105 [DEBUG] switch_ivr.c:598 sofia/external/40505060 Command Execute speak(flite|slt|Hello World)
EXECUTE sofia/external/40505060 speak(flite|slt|Hello World)
2012-08-10 17:02:29.361105 [DEBUG] switch_ivr_play_say.c:2473 OPEN TTS flite
2012-08-10 17:02:29.361105 [DEBUG] switch_ivr_play_say.c:2482 Raw Codec Activated
2012-08-10 17:02:29.541099 [DEBUG] switch_ivr_play_say.c:2164 Speaking text: Hello World
2012-08-10 17:02:29.821105 [DEBUG] switch_rtp.c:3252 Correct ip/port confirmed.
2012-08-10 17:02:31.201100 [DEBUG] switch_ivr_play_say.c:2361 done speaking text
areski
Site Admin
 
Posts: 302
Joined: Tue Oct 18, 2011 5:28 pm

Re: Calls drops when answered

Postby caronte » Mon Aug 13, 2012 12:53 pm

That’s exactly the weird thing, freeswitch says that the message is been played but the server just send a disconnection to the provider and the call drops.

U 192.168.6.29:5080 -> a.b.c.d:5060
BYE sip:23456789@a.b.c.d:5060 SIP/2.0.

I am testing this in a virtual machine of vmware, is there a restriction for that?
caronte
 
Posts: 4
Joined: Wed Aug 08, 2012 8:28 pm

Re: Calls drops when answered

Postby areski » Mon Aug 13, 2012 2:20 pm

Personally I don't use VMware but I m sure with the right setup it should work correctly.

https://www.google.com/search?q=freeswitch+VMware

Yours,
/Areski
areski
Site Admin
 
Posts: 302
Joined: Tue Oct 18, 2011 5:28 pm

Re: Calls drops when answered

Postby caronte » Thu Sep 13, 2012 2:22 pm

The problem must be with the plivo's speak resource, when I upload a file for playing it works just fine.

Thanks
caronte
 
Posts: 4
Joined: Wed Aug 08, 2012 8:28 pm


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