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mikael at bjerkeland.com Guest
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Posted: Wed Apr 15, 2009 11:08 am Post subject: [Freeswitch-users] leg_delay_start not working and hangup_af |
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Hi,
I have two scenarios I'm having trouble figuring out and I'd be happy if someone could tell me what I'm doing wrong.
1. leg_delay_start=N not working
I am trying to delay the origination of the second leg in a forked dial with the following:
<action application="bridge" data="user/mikael-nokia@voip.domain.com (mikael-nokia@voip.domain.com),[leg_delay_start=10]openzap/1/a/99355151"/>
However the second leg is called at exactly the same time as the first one. I am away from my testing environment right now, so I'm sorry for not posting my logs. It appears to me that leg_delay_start is broken on at least rev 13013.
2. I'd like to stop processing the dialplan after a bridge, but not on specific hangup causes. If I get a MEDIA_TIMEOUT hangup cause in the call I'd like to continue in the dialplan. Currently I have the following:
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="bridge" data="user/mikael-nokia@voip.domain.com (mikael-nokia@voip.domain.com)"/>
       <!-- I will only get here if the first bridge is rejected or TODO: I get a MEDIA_TIMEOUT on it -->
       <action application="bridge" data="openzap/1/a/99355151"/>
Any ideas on how to accomplish this?
Thanks,
Mikael |
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mikael at bjerkeland.com Guest
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Posted: Thu Apr 16, 2009 3:21 am Post subject: [Freeswitch-users] leg_delay_start not working and hangup_af |
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El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland escribió:
Quote: | Hi,
I have two scenarios I'm having trouble figuring out and I'd be happy
if someone could tell me what I'm doing wrong.
1. leg_delay_start=N not working
I am trying to delay the origination of the second leg in a forked
dial with the following:
<action application="bridge"
data="user/mikael-nokia@voip.domain.com,[leg_delay_start=10]openzap/1/a/99355151"/>
However the second leg is called at exactly the same time as the first
one. I am away from my testing environment right now, so I'm sorry for
not posting my logs. It appears to me that leg_delay_start is broken
on at least rev 13013.
2. I'd like to stop processing the dialplan after a bridge, but not on
specific hangup causes. If I get a MEDIA_TIMEOUT hangup cause in the
call I'd like to continue in the dialplan. Currently I have the
following:
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="bridge"
data="user/mikael-nokia@voip.domain.com"/>
<!-- I will only get here if the first bridge is rejected or
TODO: I get a MEDIA_TIMEOUT on it -->
<action application="bridge" data="openzap/1/a/99355151"/>
Any ideas on how to accomplish this?
|
I started testing this with the following dialplan:
<extension name="mikael-nokia+fallback">
<condition field="destination_number" expression="^503$">
<action application="set" data="hangup_after_bridge=false"/>
<action application="set" data="continue_on_fail=true"/>
<action application="bridge"
data="user/mikael-nokia@fs.voip.domain.com"/>
<action application="info"/>
<action application="set" data="followme_extension=99355151"/>
<action application="execute_extension"
data="post_call_followme_check"/>
<action application="hangup"/>
</condition>
</extension>
<extension name="post_call_followme_check">
<condition field="destination_number"
expression="^post_call_followme_check$"/>
<condition field="${originate_disposition}"
expression="^MEDIA_TIMEOUT|$${continue_on_fail_causes}$"
break="on-true">
<action application="log" data="1 Follow me transferring call
because of orig disposition: ${originate_disposition}"/>
<action application="transfer" data="${followme_extension}"/>
</condition>
<condition>
<action application="log" data="1 Follow me call ended normally
with orig disposition: ${originate_disposition}."/>
<action application="hangup"/>
</condition>
</extension>
${originate_disposition} never has the value of MEDIA_TIMEOUT since the
call was answered, which is absolutely correct, so what I am searching
for now is how to get the actual hangup cause. The info app doesn't show
MEDIA_TIMEOUT anywhere, but my logs show this:
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:377
audio_bridge_thread() sofia/internal/sip:mikael-nokia@10.247.3.253
ending bridge by request from read function
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/sip:mikael-nokia@10.247.3.253 [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:452
audio_bridge_thread() BRIDGE THREAD DONE
[sofia/internal/sip:mikael-nokia@10.247.3.253]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/mikael-ekiga@fs.voip.domain.com [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:508
switch_core_session_run() (sofia/internal/sip:mikael-nokia@10.247.3.253)
State EXCHANGE_MEDIA going to sleep
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:397
switch_core_session_run() (sofia/internal/sip:mikael-nokia@10.247.3.253)
Running State Change CS_HANGUP
EXECUTE sofia/internal/mikael-ekiga@fs.voip.domain.com info()
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run() (sofia/internal/sip:mikael-nokia@10.247.3.253)
State HANGUP
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup() Channel
sofia/internal/sip:mikael-nokia@10.247.3.253 hanging up, cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:370 sofia_on_hangup() Sending
BYE to sofia/internal/sip:mikael-nokia@10.247.3.253
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:46
switch_core_standard_on_hangup()
sofia/internal/sip:mikael-nokia@10.247.3.253 Standard HANGUP, cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run() (sofia/internal/sip:mikael-nokia@10.247.3.253)
State HANGUP going to sleep
2009-04-16 10:02:34 [INFO] mod_dptools.c:946 info_function()
CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/mikael-ekiga@fs.voip.domain.com]
Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [answered]
Channel-Read-Codec-Name: [G722]
Channel-Read-Codec-Rate: [16000]
Channel-Write-Codec-Name: [G722]
Channel-Write-Codec-Rate: [16000]
Caller-Username: [mikael-ekiga]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [Mikael Bjerkeland]
Caller-Caller-ID-Number: [mikael-ekiga]
Caller-Network-Addr: [10.0.255.251]
Caller-Destination-Number: [503]
Caller-Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
Caller-Source: [mod_sofia]
Caller-Context: [customers]
Caller-Channel-Name: [sofia/internal/mikael-ekiga@fs.voip.domain.com]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1239868906687578]
Caller-Channel-Created-Time: [1239868906687578]
Caller-Channel-Answered-Time: [1239868911327578]
Caller-Channel-Progress-Time: [1239868907307602]
Caller-Channel-Progress-Media-Time: [1239868911327578]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
Other-Leg-Username: [mikael-ekiga]
Other-Leg-Dialplan: [XML]
Other-Leg-Caller-ID-Name: [Mikael Bjerkeland]
Other-Leg-Caller-ID-Number: [21651012]
Other-Leg-Network-Addr: [10.247.3.253]
Other-Leg-Destination-Number: [sip:mikael-nokia@10.247.3.253]
Other-Leg-Unique-ID: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
Other-Leg-Source: [mod_sofia]
Other-Leg-Context: [customers]
Other-Leg-Channel-Name: [sofia/internal/sip:mikael-nokia@10.247.3.253]
Other-Leg-Screen-Bit: [true]
Other-Leg-Privacy-Hide-Name: [false]
Other-Leg-Privacy-Hide-Number: [false]
variable_sip_received_ip: [10.0.255.251]
variable_sip_received_port: [5065]
variable_sip_via_protocol: [udp]
variable_sip_authorized: [true]
variable_sip_mailbox: [4723695000]
variable_sip_auth_username: [mikael-ekiga]
variable_sip_auth_realm: [fs.voip.domain.com]
variable_mailbox: [4723695000]
variable_user_name: [mikael-ekiga]
variable_domain_name: [fs.voip.domain.com]
variable_effective_caller_id_number: [21651012]
variable_effective_caller_id_name: [Mikael Bjerkeland]
variable_caller_id_number: [21651012]
variable_caller_id_name: [Mikael Bjerkeland]
variable_line_open_for_external_calls: [true]
variable_room_number: [800]
variable_user_context: [customers]
variable_sip_from_user: [mikael-ekiga]
variable_sip_from_uri: [mikael-ekiga@fs.voip.domain.com]
variable_sip_from_host: [fs.voip.domain.com]
variable_sip_from_user_stripped: [mikael-ekiga]
variable_sip_from_tag: [942742a2-ca28-de11-854f-0015c583ee77]
variable_sofia_profile_name: [internal]
variable_sip_req_user: [503]
variable_sip_req_uri: [503@fs.voip.domain.com]
variable_sip_req_host: [fs.voip.domain.com]
variable_sip_to_user: [503]
variable_sip_to_uri: [503@fs.voip.domain.com]
variable_sip_to_host: [fs.voip.domain.com]
variable_sip_contact_user: [mikael-ekiga]
variable_sip_contact_port: [5065]
variable_sip_contact_uri: [mikael-ekiga@10.0.255.251:5065]
variable_sip_contact_host: [10.0.255.251]
variable_channel_name: [sofia/internal/mikael-ekiga@fs.voip.domain.com]
variable_sip_call_id:
[e82d42a2-ca28-de11-854f-0015c583ee77@mikael-xpsm1530]
variable_sip_user_agent: [Ekiga/3.2.0]
variable_sip_via_host: [10.0.255.251]
variable_sip_via_port: [5065]
variable_sip_via_rport: [5065]
variable_max_forwards: [70]
variable_presence_id: [mikael-ekiga@fs.voip.domain.com]
variable_switch_r_sdp: [v=0
o=- 1239868973 1239868973 IN IP4 10.0.255.251
s=Opal SIP Session
c=IN IP4 10.0.255.251
t=0 0
m=audio 5090 RTP/AVP 9 8 117 0 116 101 120
a=rtpmap:9 G722/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:117 Speex/16000/1
a=fmtp:117 sr=16000,mode=any
a=rtpmap:0 PCMU/8000/1
a=rtpmap:116 Speex/8000/1
a=fmtp:116 sr=8000,mode=any
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:120 NSE/8000
a=fmtp:120 192-193
m=video 5092 RTP/AVP 119 31
a=rtpmap:119 theora/90000
a=fmtp:119
delivery-method="in_band";height=576;sampling="YCbCr-4:2:0";width=704
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
]
variable_ep_codec_string:
[G722@8000h@0i,PCMA@8000h@0i,SPEEX@16000h@0i,SPEEX@16000h@0i,SPEEX@16000h@0i,PCMU@8000h@0i,H261@90000h@0i]
variable_hangup_after_bridge: [false]
variable_continue_on_fail: [true]
variable_dialed_user: [mikael-nokia]
variable_dialed_domain: [fs.voip.domain.com]
variable_switch_m_sdp: [v=0
o=Nokia-SIPUA 603233522614072812 292890395656351010 IN IP4 10.247.3.253
s=FreeSWITCH
c=IN IP4 10.247.3.253
t=0 0
m=audio 49152 RTP/AVP 8 101 13
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=rtpmap:13 CN/8000/1
a=ptime:20
a=maxptime:200
m=video 0 RTP/AVP 99
a=rtpmap:99 H264/90000
]
variable_remote_media_ip: [10.0.255.251]
variable_remote_media_port: [5090]
variable_read_codec: [G722]
variable_read_rate: [16000]
variable_write_codec: [G722]
variable_write_rate: [16000]
variable_video_possible: [true]
variable_remote_video_ip: [10.0.255.251]
variable_remote_video_port: [5092]
variable_sip_video_fmtp: [CIF=1;QCIF=1]
variable_sip_video_pt: [31]
variable_local_media_ip: [10.100.4.192]
variable_local_media_port: [56008]
variable_local_video_ip: [10.100.4.192]
variable_local_video_port: [59022]
variable_video_read_codec: [H261]
variable_video_read_rate: [90000]
variable_video_write_codec: [H261]
variable_video_write_rate: [90000]
variable_endpoint_disposition: [ANSWER]
variable_originate_disposition: [SUCCESS]
variable_bridge_channel: [sofia/internal/sip:mikael-nokia@10.247.3.253]
variable_bridge_uuid: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
variable_signal_bond: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
variable_current_application: [info]
How do I get the "raw" hangup cause first mentioned below?
"
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup() Channel
sofia/internal/sip:mikael-nokia@10.247.3.253 hanging up, cause:
MEDIA_TIMEOUT
"
As mentioned earlier the origination was in fact a success, but since I
moved out of wi-fi coverage area I got a MEDIA_TIMEOUT which should
trigger a transfer to my cell phone number. :-)
_______________________________________________
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anthony.minessale at g... Guest
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Posted: Thu Apr 16, 2009 7:51 am Post subject: [Freeswitch-users] leg_delay_start not working and hangup_af |
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turn on the debug option in mod_cdr_csv and you will get something similar to the info app only at the end of the call
On Thu, Apr 16, 2009 at 3:19 AM, Mikael Aleksander Bjerkeland <mikael@bjerkeland.com (mikael@bjerkeland.com)> wrote:
Quote: | El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland escribió:
Quote: | Hi,
I have two scenarios I'm having trouble figuring out and I'd be happy
if someone could tell me what I'm doing wrong.
1. leg_delay_start=N not working
I am trying to delay the origination of the second leg in a forked
dial with the following:
<action application="bridge"
data="user/mikael-nokia@voip.domain.com (mikael-nokia@voip.domain.com),[leg_delay_start=10]openzap/1/a/99355151"/>
However the second leg is called at exactly the same time as the first
one. I am away from my testing environment right now, so I'm sorry for
not posting my logs. It appears to me that leg_delay_start is broken
on at least rev 13013.
2. I'd like to stop processing the dialplan after a bridge, but not on
specific hangup causes. If I get a MEDIA_TIMEOUT hangup cause in the
call I'd like to continue in the dialplan. Currently I have the
following:
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="bridge"
data="user/mikael-nokia@voip.domain.com (mikael-nokia@voip.domain.com)"/>
<!-- I will only get here if the first bridge is rejected or
TODO: I get a MEDIA_TIMEOUT on it -->
<action application="bridge" data="openzap/1/a/99355151"/>
Any ideas on how to accomplish this?
|
I started testing this with the following dialplan:
<extension name="mikael-nokia+fallback">
<condition field="destination_number" expression="^503$">
<action application="set" data="hangup_after_bridge=false"/>
<action application="set" data="continue_on_fail=true"/>
<action application="bridge"
data="user/mikael-nokia@fs.voip.domain.com (mikael-nokia@fs.voip.domain.com)"/>
<action application="info"/>
<action application="set" data="followme_extension=99355151"/>
<action application="execute_extension"
data="post_call_followme_check"/>
<action application="hangup"/>
</condition>
</extension>
<extension name="post_call_followme_check">
<condition field="destination_number"
expression="^post_call_followme_check$"/>
<condition field="${originate_disposition}"
expression="^MEDIA_TIMEOUT|$${continue_on_fail_causes}$"
break="on-true">
<action application="log" data="1 Follow me transferring call
because of orig disposition: ${originate_disposition}"/>
<action application="transfer" data="${followme_extension}"/>
</condition>
<condition>
<action application="log" data="1 Follow me call ended normally
with orig disposition: ${originate_disposition}."/>
<action application="hangup"/>
</condition>
</extension>
${originate_disposition} never has the value of MEDIA_TIMEOUT since the
call was answered, which is absolutely correct, so what I am searching
for now is how to get the actual hangup cause. The info app doesn't show
MEDIA_TIMEOUT anywhere, but my logs show this:
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:377
audio_bridge_thread() sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email])
ending bridge by request from read function
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email]) [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:452
audio_bridge_thread() BRIDGE THREAD DONE
[sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email])]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/mikael-ekiga@fs.voip.domain.com (mikael-ekiga@fs.voip.domain.com) [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:508
switch_core_session_run() (sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email]))
State EXCHANGE_MEDIA going to sleep
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:397
switch_core_session_run() (sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email]))
Running State Change CS_HANGUP
EXECUTE sofia/internal/mikael-ekiga@fs.voip.domain.com (mikael-ekiga@fs.voip.domain.com) info()
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run() (sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email]))
State HANGUP
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup() Channel
sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email]) hanging up, cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:370 sofia_on_hangup() Sending
BYE to sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email])
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:46
switch_core_standard_on_hangup()
sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email]) Standard HANGUP, cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run() (sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email]))
State HANGUP going to sleep
2009-04-16 10:02:34 [INFO] mod_dptools.c:946 info_function()
CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/mikael-ekiga@fs.voip.domain.com (mikael-ekiga@fs.voip.domain.com)]
Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [answered]
Channel-Read-Codec-Name: [G722]
Channel-Read-Codec-Rate: [16000]
Channel-Write-Codec-Name: [G722]
Channel-Write-Codec-Rate: [16000]
Caller-Username: [mikael-ekiga]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [Mikael Bjerkeland]
Caller-Caller-ID-Number: [mikael-ekiga]
Caller-Network-Addr: [10.0.255.251]
Caller-Destination-Number: [503]
Caller-Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
Caller-Source: [mod_sofia]
Caller-Context: [customers]
Caller-Channel-Name: [sofia/internal/mikael-ekiga@fs.voip.domain.com (mikael-ekiga@fs.voip.domain.com)]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1239868906687578]
Caller-Channel-Created-Time: [1239868906687578]
Caller-Channel-Answered-Time: [1239868911327578]
Caller-Channel-Progress-Time: [1239868907307602]
Caller-Channel-Progress-Media-Time: [1239868911327578]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
Other-Leg-Username: [mikael-ekiga]
Other-Leg-Dialplan: [XML]
Other-Leg-Caller-ID-Name: [Mikael Bjerkeland]
Other-Leg-Caller-ID-Number: [21651012]
Other-Leg-Network-Addr: [10.247.3.253]
Other-Leg-Destination-Number: [sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email])]
Other-Leg-Unique-ID: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
Other-Leg-Source: [mod_sofia]
Other-Leg-Context: [customers]
Other-Leg-Channel-Name: [sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email])]
Other-Leg-Screen-Bit: [true]
Other-Leg-Privacy-Hide-Name: [false]
Other-Leg-Privacy-Hide-Number: [false]
variable_sip_received_ip: [10.0.255.251]
variable_sip_received_port: [5065]
variable_sip_via_protocol: [udp]
variable_sip_authorized: [true]
variable_sip_mailbox: [4723695000]
variable_sip_auth_username: [mikael-ekiga]
variable_sip_auth_realm: [fs.voip.domain.com]
variable_mailbox: [4723695000]
variable_user_name: [mikael-ekiga]
variable_domain_name: [fs.voip.domain.com]
variable_effective_caller_id_number: [21651012]
variable_effective_caller_id_name: [Mikael Bjerkeland]
variable_caller_id_number: [21651012]
variable_caller_id_name: [Mikael Bjerkeland]
variable_line_open_for_external_calls: [true]
variable_room_number: [800]
variable_user_context: [customers]
variable_sip_from_user: [mikael-ekiga]
variable_sip_from_uri: [mikael-ekiga@fs.voip.domain.com (mikael-ekiga@fs.voip.domain.com)]
variable_sip_from_host: [fs.voip.domain.com]
variable_sip_from_user_stripped: [mikael-ekiga]
variable_sip_from_tag: [942742a2-ca28-de11-854f-0015c583ee77]
variable_sofia_profile_name: [internal]
variable_sip_req_user: [503]
variable_sip_req_uri: [503@fs.voip.domain.com (503@fs.voip.domain.com)]
variable_sip_req_host: [fs.voip.domain.com]
variable_sip_to_user: [503]
variable_sip_to_uri: [503@fs.voip.domain.com (503@fs.voip.domain.com)]
variable_sip_to_host: [fs.voip.domain.com]
variable_sip_contact_user: [mikael-ekiga]
variable_sip_contact_port: [5065]
variable_sip_contact_uri: [mikael-ekiga@10.0.255.251:5065]
variable_sip_contact_host: [10.0.255.251]
variable_channel_name: [sofia/internal/mikael-ekiga@fs.voip.domain.com (mikael-ekiga@fs.voip.domain.com)]
variable_sip_call_id:
[e82d42a2-ca28-de11-854f-0015c583ee77@mikael-xpsm1530]
variable_sip_user_agent: [Ekiga/3.2.0]
variable_sip_via_host: [10.0.255.251]
variable_sip_via_port: [5065]
variable_sip_via_rport: [5065]
variable_max_forwards: [70]
variable_presence_id: [mikael-ekiga@fs.voip.domain.com (mikael-ekiga@fs.voip.domain.com)]
variable_switch_r_sdp: [v=0
o=- 1239868973 1239868973 IN IP4 10.0.255.251
s=Opal SIP Session
c=IN IP4 10.0.255.251
t=0 0
m=audio 5090 RTP/AVP 9 8 117 0 116 101 120
a=rtpmap:9 G722/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:117 Speex/16000/1
a=fmtp:117 sr=16000,mode=any
a=rtpmap:0 PCMU/8000/1
a=rtpmap:116 Speex/8000/1
a=fmtp:116 sr=8000,mode=any
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:120 NSE/8000
a=fmtp:120 192-193
m=video 5092 RTP/AVP 119 31
a=rtpmap:119 theora/90000
a=fmtp:119
delivery-method="in_band";height=576;sampling="YCbCr-4:2:0";width=704
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
]
variable_ep_codec_string:
[G722@8000h@0i,PCMA@8000h@0i,SPEEX@16000h@0i,SPEEX@16000h@0i,SPEEX@16000h@0i,PCMU@8000h@0i,H261@90000h@0i]
variable_hangup_after_bridge: [false]
variable_continue_on_fail: [true]
variable_dialed_user: [mikael-nokia]
variable_dialed_domain: [fs.voip.domain.com]
variable_switch_m_sdp: [v=0
o=Nokia-SIPUA 603233522614072812 292890395656351010 IN IP4 10.247.3.253
s=FreeSWITCH
c=IN IP4 10.247.3.253
t=0 0
m=audio 49152 RTP/AVP 8 101 13
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=rtpmap:13 CN/8000/1
a=ptime:20
a=maxptime:200
m=video 0 RTP/AVP 99
a=rtpmap:99 H264/90000
]
variable_remote_media_ip: [10.0.255.251]
variable_remote_media_port: [5090]
variable_read_codec: [G722]
variable_read_rate: [16000]
variable_write_codec: [G722]
variable_write_rate: [16000]
variable_video_possible: [true]
variable_remote_video_ip: [10.0.255.251]
variable_remote_video_port: [5092]
variable_sip_video_fmtp: [CIF=1;QCIF=1]
variable_sip_video_pt: [31]
variable_local_media_ip: [10.100.4.192]
variable_local_media_port: [56008]
variable_local_video_ip: [10.100.4.192]
variable_local_video_port: [59022]
variable_video_read_codec: [H261]
variable_video_read_rate: [90000]
variable_video_write_codec: [H261]
variable_video_write_rate: [90000]
variable_endpoint_disposition: [ANSWER]
variable_originate_disposition: [SUCCESS]
variable_bridge_channel: [sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email])]
variable_bridge_uuid: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
variable_signal_bond: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
variable_current_application: [info]
How do I get the "raw" hangup cause first mentioned below?
"
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup() Channel
sofia/internal/sip:mikael-nokia@10.247.3.253 ([email]sip%3Amikael-nokia@10.247.3.253[/email]) hanging up, cause:
MEDIA_TIMEOUT
"
As mentioned earlier the origination was in fact a success, but since I
moved out of wi-fi coverage area I got a MEDIA_TIMEOUT which should
trigger a transfer to my cell phone number.
_______________________________________________
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
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mikael at bjerkeland.com Guest
|
Posted: Thu Apr 16, 2009 9:22 am Post subject: [Freeswitch-users] leg_delay_start not working and hangup_af |
|
|
Thanks. I just tested and got some more data but it didn't contain any
variable containing MEDIA_TIMEOUT. Perhaps it's not really set anywhere?
variable_hangup_cause and variable_originate_disposition contain
NORMAL_CLEARING and SUCCESS respectively. I need a var which contains
the real reason for the hangup of the bridge, which in this case is
MEDIA_TIMEOUT as you can see from the logs.
El jue, 16-04-2009 a las 07:37 -0500, Anthony Minessale escribió:
Quote: | turn on the debug option in mod_cdr_csv and you will get something
similar to the info app only at the end of the call
On Thu, Apr 16, 2009 at 3:19 AM, Mikael Aleksander Bjerkeland
<mikael@bjerkeland.com> wrote:
El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland
escribió:
Quote: | Hi,
I have two scenarios I'm having trouble figuring out and I'd
| be happy
Quote: | if someone could tell me what I'm doing wrong.
1. leg_delay_start=N not working
I am trying to delay the origination of the second leg in a
| forked
Quote: | dial with the following:
<action application="bridge"
| data="user/mikael-nokia@voip.domain.com,[leg_delay_start=10]openzap/1/a/99355151"/>
Quote: |
However the second leg is called at exactly the same time as
| the first
Quote: | one. I am away from my testing environment right now, so I'm
| sorry for
Quote: | not posting my logs. It appears to me that leg_delay_start
| is broken
Quote: | on at least rev 13013.
2. I'd like to stop processing the dialplan after a bridge,
| but not on
Quote: | specific hangup causes. If I get a MEDIA_TIMEOUT hangup
| cause in the
Quote: | call I'd like to continue in the dialplan. Currently I have
| the
Quote: | following:
<action application="set"
| data="hangup_after_bridge=true"/>
Quote: | <action application="set"
| data="continue_on_fail=true"/>
Quote: | <action application="bridge"
data="user/mikael-nokia@voip.domain.com"/>
<!-- I will only get here if the first bridge is
| rejected or
Quote: | TODO: I get a MEDIA_TIMEOUT on it -->
<action application="bridge"
| data="openzap/1/a/99355151"/>
Quote: |
Any ideas on how to accomplish this?
|
I started testing this with the following dialplan:
<extension name="mikael-nokia+fallback">
<condition field="destination_number" expression="^503$">
<action application="set"
data="hangup_after_bridge=false"/>
<action application="set"
data="continue_on_fail=true"/>
<action application="bridge"
data="user/mikael-nokia@fs.voip.domain.com"/>
<action application="info"/>
<action application="set"
data="followme_extension=99355151"/>
<action application="execute_extension"
data="post_call_followme_check"/>
<action application="hangup"/>
</condition>
</extension>
<extension name="post_call_followme_check">
<condition field="destination_number"
expression="^post_call_followme_check$"/>
<condition field="${originate_disposition}"
expression="^MEDIA_TIMEOUT|$${continue_on_fail_causes}$"
break="on-true">
<action application="log" data="1 Follow me transferring
call
because of orig disposition: ${originate_disposition}"/>
<action application="transfer"
data="${followme_extension}"/>
</condition>
<condition>
<action application="log" data="1 Follow me call ended
normally
with orig disposition: ${originate_disposition}."/>
<action application="hangup"/>
</condition>
</extension>
${originate_disposition} never has the value of MEDIA_TIMEOUT
since the
call was answered, which is absolutely correct, so what I am
searching
for now is how to get the actual hangup cause. The info app
doesn't show
MEDIA_TIMEOUT anywhere, but my logs show this:
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:377
audio_bridge_thread()
sofia/internal/sip:mikael-nokia@10.247.3.253
ending bridge by request from read function
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/sip:mikael-nokia@10.247.3.253 [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:452
audio_bridge_thread() BRIDGE THREAD DONE
[sofia/internal/sip:mikael-nokia@10.247.3.253]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/mikael-ekiga@fs.voip.domain.com [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:508
switch_core_session_run()
(sofia/internal/sip:mikael-nokia@10.247.3.253)
State EXCHANGE_MEDIA going to sleep
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:397
switch_core_session_run()
(sofia/internal/sip:mikael-nokia@10.247.3.253)
Running State Change CS_HANGUP
EXECUTE sofia/internal/mikael-ekiga@fs.voip.domain.com info()
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run()
(sofia/internal/sip:mikael-nokia@10.247.3.253)
State HANGUP
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup()
Channel
sofia/internal/sip:mikael-nokia@10.247.3.253 hanging up,
cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:370 sofia_on_hangup()
Sending
BYE to sofia/internal/sip:mikael-nokia@10.247.3.253
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:46
switch_core_standard_on_hangup()
sofia/internal/sip:mikael-nokia@10.247.3.253 Standard HANGUP,
cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run()
(sofia/internal/sip:mikael-nokia@10.247.3.253)
State HANGUP going to sleep
2009-04-16 10:02:34 [INFO] mod_dptools.c:946 info_function()
CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/mikael-ekiga@fs.voip.domain.com]
Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [answered]
Channel-Read-Codec-Name: [G722]
Channel-Read-Codec-Rate: [16000]
Channel-Write-Codec-Name: [G722]
Channel-Write-Codec-Rate: [16000]
Caller-Username: [mikael-ekiga]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [Mikael Bjerkeland]
Caller-Caller-ID-Number: [mikael-ekiga]
Caller-Network-Addr: [10.0.255.251]
Caller-Destination-Number: [503]
Caller-Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
Caller-Source: [mod_sofia]
Caller-Context: [customers]
Caller-Channel-Name:
[sofia/internal/mikael-ekiga@fs.voip.domain.com]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1239868906687578]
Caller-Channel-Created-Time: [1239868906687578]
Caller-Channel-Answered-Time: [1239868911327578]
Caller-Channel-Progress-Time: [1239868907307602]
Caller-Channel-Progress-Media-Time: [1239868911327578]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
Other-Leg-Username: [mikael-ekiga]
Other-Leg-Dialplan: [XML]
Other-Leg-Caller-ID-Name: [Mikael Bjerkeland]
Other-Leg-Caller-ID-Number: [21651012]
Other-Leg-Network-Addr: [10.247.3.253]
Other-Leg-Destination-Number: [sip:mikael-nokia@10.247.3.253]
Other-Leg-Unique-ID: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
Other-Leg-Source: [mod_sofia]
Other-Leg-Context: [customers]
Other-Leg-Channel-Name:
[sofia/internal/sip:mikael-nokia@10.247.3.253]
Other-Leg-Screen-Bit: [true]
Other-Leg-Privacy-Hide-Name: [false]
Other-Leg-Privacy-Hide-Number: [false]
variable_sip_received_ip: [10.0.255.251]
variable_sip_received_port: [5065]
variable_sip_via_protocol: [udp]
variable_sip_authorized: [true]
variable_sip_mailbox: [4723695000]
variable_sip_auth_username: [mikael-ekiga]
variable_sip_auth_realm: [fs.voip.domain.com]
variable_mailbox: [4723695000]
variable_user_name: [mikael-ekiga]
variable_domain_name: [fs.voip.domain.com]
variable_effective_caller_id_number: [21651012]
variable_effective_caller_id_name: [Mikael Bjerkeland]
variable_caller_id_number: [21651012]
variable_caller_id_name: [Mikael Bjerkeland]
variable_line_open_for_external_calls: [true]
variable_room_number: [800]
variable_user_context: [customers]
variable_sip_from_user: [mikael-ekiga]
variable_sip_from_uri: [mikael-ekiga@fs.voip.domain.com]
variable_sip_from_host: [fs.voip.domain.com]
variable_sip_from_user_stripped: [mikael-ekiga]
variable_sip_from_tag: [942742a2-ca28-de11-854f-0015c583ee77]
variable_sofia_profile_name: [internal]
variable_sip_req_user: [503]
variable_sip_req_uri: [503@fs.voip.domain.com]
variable_sip_req_host: [fs.voip.domain.com]
variable_sip_to_user: [503]
variable_sip_to_uri: [503@fs.voip.domain.com]
variable_sip_to_host: [fs.voip.domain.com]
variable_sip_contact_user: [mikael-ekiga]
variable_sip_contact_port: [5065]
variable_sip_contact_uri: [mikael-ekiga@10.0.255.251:5065]
variable_sip_contact_host: [10.0.255.251]
variable_channel_name:
[sofia/internal/mikael-ekiga@fs.voip.domain.com]
variable_sip_call_id:
[e82d42a2-ca28-de11-854f-0015c583ee77@mikael-xpsm1530]
variable_sip_user_agent: [Ekiga/3.2.0]
variable_sip_via_host: [10.0.255.251]
variable_sip_via_port: [5065]
variable_sip_via_rport: [5065]
variable_max_forwards: [70]
variable_presence_id: [mikael-ekiga@fs.voip.domain.com]
variable_switch_r_sdp: [v=0
o=- 1239868973 1239868973 IN IP4 10.0.255.251
s=Opal SIP Session
c=IN IP4 10.0.255.251
t=0 0
m=audio 5090 RTP/AVP 9 8 117 0 116 101 120
a=rtpmap:9 G722/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:117 Speex/16000/1
a=fmtp:117 sr=16000,mode=any
a=rtpmap:0 PCMU/8000/1
a=rtpmap:116 Speex/8000/1
a=fmtp:116 sr=8000,mode=any
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:120 NSE/8000
a=fmtp:120 192-193
m=video 5092 RTP/AVP 119 31
a=rtpmap:119 theora/90000
a=fmtp:119
delivery-method="in_band";height=576;sampling="YCbCr-4:2:0";width=704
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
]
variable_ep_codec_string:
[G722@8000h@0i,PCMA@8000h@0i,SPEEX@16000h@0i,SPEEX@16000h@0i,SPEEX@16000h@0i,PCMU@8000h@0i,H261@90000h@0i]
variable_hangup_after_bridge: [false]
variable_continue_on_fail: [true]
variable_dialed_user: [mikael-nokia]
variable_dialed_domain: [fs.voip.domain.com]
variable_switch_m_sdp: [v=0
o=Nokia-SIPUA 603233522614072812 292890395656351010 IN IP4
10.247.3.253
s=FreeSWITCH
c=IN IP4 10.247.3.253
t=0 0
m=audio 49152 RTP/AVP 8 101 13
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=rtpmap:13 CN/8000/1
a=ptime:20
a=maxptime:200
m=video 0 RTP/AVP 99
a=rtpmap:99 H264/90000
]
variable_remote_media_ip: [10.0.255.251]
variable_remote_media_port: [5090]
variable_read_codec: [G722]
variable_read_rate: [16000]
variable_write_codec: [G722]
variable_write_rate: [16000]
variable_video_possible: [true]
variable_remote_video_ip: [10.0.255.251]
variable_remote_video_port: [5092]
variable_sip_video_fmtp: [CIF=1;QCIF=1]
variable_sip_video_pt: [31]
variable_local_media_ip: [10.100.4.192]
variable_local_media_port: [56008]
variable_local_video_ip: [10.100.4.192]
variable_local_video_port: [59022]
variable_video_read_codec: [H261]
variable_video_read_rate: [90000]
variable_video_write_codec: [H261]
variable_video_write_rate: [90000]
variable_endpoint_disposition: [ANSWER]
variable_originate_disposition: [SUCCESS]
variable_bridge_channel:
[sofia/internal/sip:mikael-nokia@10.247.3.253]
variable_bridge_uuid: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
variable_signal_bond: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
variable_current_application: [info]
How do I get the "raw" hangup cause first mentioned below?
"
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup()
Channel
sofia/internal/sip:mikael-nokia@10.247.3.253 hanging up,
cause:
MEDIA_TIMEOUT
"
As mentioned earlier the origination was in fact a success,
but since I
moved out of wi-fi coverage area I got a MEDIA_TIMEOUT which
should
trigger a transfer to my cell phone number. :-)
_______________________________________________
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/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
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mikael at bjerkeland.com Guest
|
Posted: Thu Apr 16, 2009 9:56 am Post subject: [Freeswitch-users] leg_delay_start not working and hangup_af |
|
|
I think I know a bit more about the problem now. The MEDIA_TIMEOUT
hangup cause is probably coming from the B leg of the call and thus not
visible when I do info or debug on mod_cdr_csv.
I then tried the following after bridge to get it:
<action application="set" data="other_leg_hangup_cause=
${uuid_getvar(${bridge_uuid} hangup_cause)}"/>
However, since that bridge of the call is already hung up I got the
following in reply:
variable_other_leg_hangup_cause: [-ERR No Such Channel!
]
Is there a way to get it from the B leg of the call - assuming that's
where the hangup cause comes from?
Thanks!
El jue, 16-04-2009 a las 16:07 +0200, Mikael Aleksander Bjerkeland
escribió:
Quote: | Thanks. I just tested and got some more data but it didn't contain any
variable containing MEDIA_TIMEOUT. Perhaps it's not really set anywhere?
variable_hangup_cause and variable_originate_disposition contain
NORMAL_CLEARING and SUCCESS respectively. I need a var which contains
the real reason for the hangup of the bridge, which in this case is
MEDIA_TIMEOUT as you can see from the logs.
El jue, 16-04-2009 a las 07:37 -0500, Anthony Minessale escribió:
Quote: | turn on the debug option in mod_cdr_csv and you will get something
similar to the info app only at the end of the call
On Thu, Apr 16, 2009 at 3:19 AM, Mikael Aleksander Bjerkeland
<mikael@bjerkeland.com> wrote:
El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland
escribió:
Quote: | Hi,
I have two scenarios I'm having trouble figuring out and I'd
| be happy
Quote: | if someone could tell me what I'm doing wrong.
1. leg_delay_start=N not working
I am trying to delay the origination of the second leg in a
| forked
Quote: | dial with the following:
<action application="bridge"
| data="user/mikael-nokia@voip.domain.com,[leg_delay_start=10]openzap/1/a/99355151"/>
Quote: |
However the second leg is called at exactly the same time as
| the first
Quote: | one. I am away from my testing environment right now, so I'm
| sorry for
Quote: | not posting my logs. It appears to me that leg_delay_start
| is broken
Quote: | on at least rev 13013.
2. I'd like to stop processing the dialplan after a bridge,
| but not on
Quote: | specific hangup causes. If I get a MEDIA_TIMEOUT hangup
| cause in the
Quote: | call I'd like to continue in the dialplan. Currently I have
| the
Quote: | following:
<action application="set"
| data="hangup_after_bridge=true"/>
Quote: | <action application="set"
| data="continue_on_fail=true"/>
Quote: | <action application="bridge"
data="user/mikael-nokia@voip.domain.com"/>
<!-- I will only get here if the first bridge is
| rejected or
Quote: | TODO: I get a MEDIA_TIMEOUT on it -->
<action application="bridge"
| data="openzap/1/a/99355151"/>
Quote: |
Any ideas on how to accomplish this?
|
I started testing this with the following dialplan:
<extension name="mikael-nokia+fallback">
<condition field="destination_number" expression="^503$">
<action application="set"
data="hangup_after_bridge=false"/>
<action application="set"
data="continue_on_fail=true"/>
<action application="bridge"
data="user/mikael-nokia@fs.voip.domain.com"/>
<action application="info"/>
<action application="set"
data="followme_extension=99355151"/>
<action application="execute_extension"
data="post_call_followme_check"/>
<action application="hangup"/>
</condition>
</extension>
<extension name="post_call_followme_check">
<condition field="destination_number"
expression="^post_call_followme_check$"/>
<condition field="${originate_disposition}"
expression="^MEDIA_TIMEOUT|$${continue_on_fail_causes}$"
break="on-true">
<action application="log" data="1 Follow me transferring
call
because of orig disposition: ${originate_disposition}"/>
<action application="transfer"
data="${followme_extension}"/>
</condition>
<condition>
<action application="log" data="1 Follow me call ended
normally
with orig disposition: ${originate_disposition}."/>
<action application="hangup"/>
</condition>
</extension>
${originate_disposition} never has the value of MEDIA_TIMEOUT
since the
call was answered, which is absolutely correct, so what I am
searching
for now is how to get the actual hangup cause. The info app
doesn't show
MEDIA_TIMEOUT anywhere, but my logs show this:
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:377
audio_bridge_thread()
sofia/internal/sip:mikael-nokia@10.247.3.253
ending bridge by request from read function
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/sip:mikael-nokia@10.247.3.253 [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:452
audio_bridge_thread() BRIDGE THREAD DONE
[sofia/internal/sip:mikael-nokia@10.247.3.253]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/mikael-ekiga@fs.voip.domain.com [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:508
switch_core_session_run()
(sofia/internal/sip:mikael-nokia@10.247.3.253)
State EXCHANGE_MEDIA going to sleep
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:397
switch_core_session_run()
(sofia/internal/sip:mikael-nokia@10.247.3.253)
Running State Change CS_HANGUP
EXECUTE sofia/internal/mikael-ekiga@fs.voip.domain.com info()
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run()
(sofia/internal/sip:mikael-nokia@10.247.3.253)
State HANGUP
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup()
Channel
sofia/internal/sip:mikael-nokia@10.247.3.253 hanging up,
cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:370 sofia_on_hangup()
Sending
BYE to sofia/internal/sip:mikael-nokia@10.247.3.253
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:46
switch_core_standard_on_hangup()
sofia/internal/sip:mikael-nokia@10.247.3.253 Standard HANGUP,
cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run()
(sofia/internal/sip:mikael-nokia@10.247.3.253)
State HANGUP going to sleep
2009-04-16 10:02:34 [INFO] mod_dptools.c:946 info_function()
CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/mikael-ekiga@fs.voip.domain.com]
Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [answered]
Channel-Read-Codec-Name: [G722]
Channel-Read-Codec-Rate: [16000]
Channel-Write-Codec-Name: [G722]
Channel-Write-Codec-Rate: [16000]
Caller-Username: [mikael-ekiga]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [Mikael Bjerkeland]
Caller-Caller-ID-Number: [mikael-ekiga]
Caller-Network-Addr: [10.0.255.251]
Caller-Destination-Number: [503]
Caller-Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
Caller-Source: [mod_sofia]
Caller-Context: [customers]
Caller-Channel-Name:
[sofia/internal/mikael-ekiga@fs.voip.domain.com]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1239868906687578]
Caller-Channel-Created-Time: [1239868906687578]
Caller-Channel-Answered-Time: [1239868911327578]
Caller-Channel-Progress-Time: [1239868907307602]
Caller-Channel-Progress-Media-Time: [1239868911327578]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
Other-Leg-Username: [mikael-ekiga]
Other-Leg-Dialplan: [XML]
Other-Leg-Caller-ID-Name: [Mikael Bjerkeland]
Other-Leg-Caller-ID-Number: [21651012]
Other-Leg-Network-Addr: [10.247.3.253]
Other-Leg-Destination-Number: [sip:mikael-nokia@10.247.3.253]
Other-Leg-Unique-ID: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
Other-Leg-Source: [mod_sofia]
Other-Leg-Context: [customers]
Other-Leg-Channel-Name:
[sofia/internal/sip:mikael-nokia@10.247.3.253]
Other-Leg-Screen-Bit: [true]
Other-Leg-Privacy-Hide-Name: [false]
Other-Leg-Privacy-Hide-Number: [false]
variable_sip_received_ip: [10.0.255.251]
variable_sip_received_port: [5065]
variable_sip_via_protocol: [udp]
variable_sip_authorized: [true]
variable_sip_mailbox: [4723695000]
variable_sip_auth_username: [mikael-ekiga]
variable_sip_auth_realm: [fs.voip.domain.com]
variable_mailbox: [4723695000]
variable_user_name: [mikael-ekiga]
variable_domain_name: [fs.voip.domain.com]
variable_effective_caller_id_number: [21651012]
variable_effective_caller_id_name: [Mikael Bjerkeland]
variable_caller_id_number: [21651012]
variable_caller_id_name: [Mikael Bjerkeland]
variable_line_open_for_external_calls: [true]
variable_room_number: [800]
variable_user_context: [customers]
variable_sip_from_user: [mikael-ekiga]
variable_sip_from_uri: [mikael-ekiga@fs.voip.domain.com]
variable_sip_from_host: [fs.voip.domain.com]
variable_sip_from_user_stripped: [mikael-ekiga]
variable_sip_from_tag: [942742a2-ca28-de11-854f-0015c583ee77]
variable_sofia_profile_name: [internal]
variable_sip_req_user: [503]
variable_sip_req_uri: [503@fs.voip.domain.com]
variable_sip_req_host: [fs.voip.domain.com]
variable_sip_to_user: [503]
variable_sip_to_uri: [503@fs.voip.domain.com]
variable_sip_to_host: [fs.voip.domain.com]
variable_sip_contact_user: [mikael-ekiga]
variable_sip_contact_port: [5065]
variable_sip_contact_uri: [mikael-ekiga@10.0.255.251:5065]
variable_sip_contact_host: [10.0.255.251]
variable_channel_name:
[sofia/internal/mikael-ekiga@fs.voip.domain.com]
variable_sip_call_id:
[e82d42a2-ca28-de11-854f-0015c583ee77@mikael-xpsm1530]
variable_sip_user_agent: [Ekiga/3.2.0]
variable_sip_via_host: [10.0.255.251]
variable_sip_via_port: [5065]
variable_sip_via_rport: [5065]
variable_max_forwards: [70]
variable_presence_id: [mikael-ekiga@fs.voip.domain.com]
variable_switch_r_sdp: [v=0
o=- 1239868973 1239868973 IN IP4 10.0.255.251
s=Opal SIP Session
c=IN IP4 10.0.255.251
t=0 0
m=audio 5090 RTP/AVP 9 8 117 0 116 101 120
a=rtpmap:9 G722/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:117 Speex/16000/1
a=fmtp:117 sr=16000,mode=any
a=rtpmap:0 PCMU/8000/1
a=rtpmap:116 Speex/8000/1
a=fmtp:116 sr=8000,mode=any
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:120 NSE/8000
a=fmtp:120 192-193
m=video 5092 RTP/AVP 119 31
a=rtpmap:119 theora/90000
a=fmtp:119
delivery-method="in_band";height=576;sampling="YCbCr-4:2:0";width=704
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
]
variable_ep_codec_string:
[G722@8000h@0i,PCMA@8000h@0i,SPEEX@16000h@0i,SPEEX@16000h@0i,SPEEX@16000h@0i,PCMU@8000h@0i,H261@90000h@0i]
variable_hangup_after_bridge: [false]
variable_continue_on_fail: [true]
variable_dialed_user: [mikael-nokia]
variable_dialed_domain: [fs.voip.domain.com]
variable_switch_m_sdp: [v=0
o=Nokia-SIPUA 603233522614072812 292890395656351010 IN IP4
10.247.3.253
s=FreeSWITCH
c=IN IP4 10.247.3.253
t=0 0
m=audio 49152 RTP/AVP 8 101 13
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=rtpmap:13 CN/8000/1
a=ptime:20
a=maxptime:200
m=video 0 RTP/AVP 99
a=rtpmap:99 H264/90000
]
variable_remote_media_ip: [10.0.255.251]
variable_remote_media_port: [5090]
variable_read_codec: [G722]
variable_read_rate: [16000]
variable_write_codec: [G722]
variable_write_rate: [16000]
variable_video_possible: [true]
variable_remote_video_ip: [10.0.255.251]
variable_remote_video_port: [5092]
variable_sip_video_fmtp: [CIF=1;QCIF=1]
variable_sip_video_pt: [31]
variable_local_media_ip: [10.100.4.192]
variable_local_media_port: [56008]
variable_local_video_ip: [10.100.4.192]
variable_local_video_port: [59022]
variable_video_read_codec: [H261]
variable_video_read_rate: [90000]
variable_video_write_codec: [H261]
variable_video_write_rate: [90000]
variable_endpoint_disposition: [ANSWER]
variable_originate_disposition: [SUCCESS]
variable_bridge_channel:
[sofia/internal/sip:mikael-nokia@10.247.3.253]
variable_bridge_uuid: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
variable_signal_bond: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
variable_current_application: [info]
How do I get the "raw" hangup cause first mentioned below?
"
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup()
Channel
sofia/internal/sip:mikael-nokia@10.247.3.253 hanging up,
cause:
MEDIA_TIMEOUT
"
As mentioned earlier the origination was in fact a success,
but since I
moved out of wi-fi coverage area I got a MEDIA_TIMEOUT which
should
trigger a transfer to my cell phone number. :-)
_______________________________________________
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
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googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
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|
mikael at bjerkeland.com Guest
|
Posted: Fri Apr 17, 2009 2:40 am Post subject: [Freeswitch-users] leg_delay_start not working and hangup_af |
|
|
Anthony implemented ${bridge_hangup_cause} in r13065 which has the value
of the last B leg bridge attempt. Works as expected!
The leg_delay_start problem still remains though. It would be great if
someone could do a test to see if leg_delay_start on a leg isn't
honored.
El jue, 16-04-2009 a las 16:36 +0200, Mikael Aleksander Bjerkeland
escribió:
Quote: | I think I know a bit more about the problem now. The MEDIA_TIMEOUT
hangup cause is probably coming from the B leg of the call and thus not
visible when I do info or debug on mod_cdr_csv.
I then tried the following after bridge to get it:
<action application="set" data="other_leg_hangup_cause=
${uuid_getvar(${bridge_uuid} hangup_cause)}"/>
However, since that bridge of the call is already hung up I got the
following in reply:
variable_other_leg_hangup_cause: [-ERR No Such Channel!
]
Is there a way to get it from the B leg of the call - assuming that's
where the hangup cause comes from?
Thanks!
El jue, 16-04-2009 a las 16:07 +0200, Mikael Aleksander Bjerkeland
escribió:
Quote: | Thanks. I just tested and got some more data but it didn't contain any
variable containing MEDIA_TIMEOUT. Perhaps it's not really set anywhere?
variable_hangup_cause and variable_originate_disposition contain
NORMAL_CLEARING and SUCCESS respectively. I need a var which contains
the real reason for the hangup of the bridge, which in this case is
MEDIA_TIMEOUT as you can see from the logs.
El jue, 16-04-2009 a las 07:37 -0500, Anthony Minessale escribió:
Quote: | turn on the debug option in mod_cdr_csv and you will get something
similar to the info app only at the end of the call
On Thu, Apr 16, 2009 at 3:19 AM, Mikael Aleksander Bjerkeland
<mikael@bjerkeland.com> wrote:
El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland
escribió:
Quote: | Hi,
I have two scenarios I'm having trouble figuring out and I'd
| be happy
Quote: | if someone could tell me what I'm doing wrong.
1. leg_delay_start=N not working
I am trying to delay the origination of the second leg in a
| forked
Quote: | dial with the following:
<action application="bridge"
| data="user/mikael-nokia@voip.domain.com,[leg_delay_start=10]openzap/1/a/99355151"/>
Quote: |
However the second leg is called at exactly the same time as
| the first
Quote: | one. I am away from my testing environment right now, so I'm
| sorry for
Quote: | not posting my logs. It appears to me that leg_delay_start
| is broken
Quote: | on at least rev 13013.
2. I'd like to stop processing the dialplan after a bridge,
| but not on
Quote: | specific hangup causes. If I get a MEDIA_TIMEOUT hangup
| cause in the
Quote: | call I'd like to continue in the dialplan. Currently I have
| the
Quote: | following:
<action application="set"
| data="hangup_after_bridge=true"/>
Quote: | <action application="set"
| data="continue_on_fail=true"/>
Quote: | <action application="bridge"
data="user/mikael-nokia@voip.domain.com"/>
<!-- I will only get here if the first bridge is
| rejected or
Quote: | TODO: I get a MEDIA_TIMEOUT on it -->
<action application="bridge"
| data="openzap/1/a/99355151"/>
Quote: |
Any ideas on how to accomplish this?
|
I started testing this with the following dialplan:
<extension name="mikael-nokia+fallback">
<condition field="destination_number" expression="^503$">
<action application="set"
data="hangup_after_bridge=false"/>
<action application="set"
data="continue_on_fail=true"/>
<action application="bridge"
data="user/mikael-nokia@fs.voip.domain.com"/>
<action application="info"/>
<action application="set"
data="followme_extension=99355151"/>
<action application="execute_extension"
data="post_call_followme_check"/>
<action application="hangup"/>
</condition>
</extension>
<extension name="post_call_followme_check">
<condition field="destination_number"
expression="^post_call_followme_check$"/>
<condition field="${originate_disposition}"
expression="^MEDIA_TIMEOUT|$${continue_on_fail_causes}$"
break="on-true">
<action application="log" data="1 Follow me transferring
call
because of orig disposition: ${originate_disposition}"/>
<action application="transfer"
data="${followme_extension}"/>
</condition>
<condition>
<action application="log" data="1 Follow me call ended
normally
with orig disposition: ${originate_disposition}."/>
<action application="hangup"/>
</condition>
</extension>
${originate_disposition} never has the value of MEDIA_TIMEOUT
since the
call was answered, which is absolutely correct, so what I am
searching
for now is how to get the actual hangup cause. The info app
doesn't show
MEDIA_TIMEOUT anywhere, but my logs show this:
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:377
audio_bridge_thread()
sofia/internal/sip:mikael-nokia@10.247.3.253
ending bridge by request from read function
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/sip:mikael-nokia@10.247.3.253 [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:452
audio_bridge_thread() BRIDGE THREAD DONE
[sofia/internal/sip:mikael-nokia@10.247.3.253]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/mikael-ekiga@fs.voip.domain.com [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:508
switch_core_session_run()
(sofia/internal/sip:mikael-nokia@10.247.3.253)
State EXCHANGE_MEDIA going to sleep
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:397
switch_core_session_run()
(sofia/internal/sip:mikael-nokia@10.247.3.253)
Running State Change CS_HANGUP
EXECUTE sofia/internal/mikael-ekiga@fs.voip.domain.com info()
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run()
(sofia/internal/sip:mikael-nokia@10.247.3.253)
State HANGUP
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup()
Channel
sofia/internal/sip:mikael-nokia@10.247.3.253 hanging up,
cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:370 sofia_on_hangup()
Sending
BYE to sofia/internal/sip:mikael-nokia@10.247.3.253
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:46
switch_core_standard_on_hangup()
sofia/internal/sip:mikael-nokia@10.247.3.253 Standard HANGUP,
cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run()
(sofia/internal/sip:mikael-nokia@10.247.3.253)
State HANGUP going to sleep
2009-04-16 10:02:34 [INFO] mod_dptools.c:946 info_function()
CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/mikael-ekiga@fs.voip.domain.com]
Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [answered]
Channel-Read-Codec-Name: [G722]
Channel-Read-Codec-Rate: [16000]
Channel-Write-Codec-Name: [G722]
Channel-Write-Codec-Rate: [16000]
Caller-Username: [mikael-ekiga]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [Mikael Bjerkeland]
Caller-Caller-ID-Number: [mikael-ekiga]
Caller-Network-Addr: [10.0.255.251]
Caller-Destination-Number: [503]
Caller-Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
Caller-Source: [mod_sofia]
Caller-Context: [customers]
Caller-Channel-Name:
[sofia/internal/mikael-ekiga@fs.voip.domain.com]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1239868906687578]
Caller-Channel-Created-Time: [1239868906687578]
Caller-Channel-Answered-Time: [1239868911327578]
Caller-Channel-Progress-Time: [1239868907307602]
Caller-Channel-Progress-Media-Time: [1239868911327578]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
Other-Leg-Username: [mikael-ekiga]
Other-Leg-Dialplan: [XML]
Other-Leg-Caller-ID-Name: [Mikael Bjerkeland]
Other-Leg-Caller-ID-Number: [21651012]
Other-Leg-Network-Addr: [10.247.3.253]
Other-Leg-Destination-Number: [sip:mikael-nokia@10.247.3.253]
Other-Leg-Unique-ID: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
Other-Leg-Source: [mod_sofia]
Other-Leg-Context: [customers]
Other-Leg-Channel-Name:
[sofia/internal/sip:mikael-nokia@10.247.3.253]
Other-Leg-Screen-Bit: [true]
Other-Leg-Privacy-Hide-Name: [false]
Other-Leg-Privacy-Hide-Number: [false]
variable_sip_received_ip: [10.0.255.251]
variable_sip_received_port: [5065]
variable_sip_via_protocol: [udp]
variable_sip_authorized: [true]
variable_sip_mailbox: [4723695000]
variable_sip_auth_username: [mikael-ekiga]
variable_sip_auth_realm: [fs.voip.domain.com]
variable_mailbox: [4723695000]
variable_user_name: [mikael-ekiga]
variable_domain_name: [fs.voip.domain.com]
variable_effective_caller_id_number: [21651012]
variable_effective_caller_id_name: [Mikael Bjerkeland]
variable_caller_id_number: [21651012]
variable_caller_id_name: [Mikael Bjerkeland]
variable_line_open_for_external_calls: [true]
variable_room_number: [800]
variable_user_context: [customers]
variable_sip_from_user: [mikael-ekiga]
variable_sip_from_uri: [mikael-ekiga@fs.voip.domain.com]
variable_sip_from_host: [fs.voip.domain.com]
variable_sip_from_user_stripped: [mikael-ekiga]
variable_sip_from_tag: [942742a2-ca28-de11-854f-0015c583ee77]
variable_sofia_profile_name: [internal]
variable_sip_req_user: [503]
variable_sip_req_uri: [503@fs.voip.domain.com]
variable_sip_req_host: [fs.voip.domain.com]
variable_sip_to_user: [503]
variable_sip_to_uri: [503@fs.voip.domain.com]
variable_sip_to_host: [fs.voip.domain.com]
variable_sip_contact_user: [mikael-ekiga]
variable_sip_contact_port: [5065]
variable_sip_contact_uri: [mikael-ekiga@10.0.255.251:5065]
variable_sip_contact_host: [10.0.255.251]
variable_channel_name:
[sofia/internal/mikael-ekiga@fs.voip.domain.com]
variable_sip_call_id:
[e82d42a2-ca28-de11-854f-0015c583ee77@mikael-xpsm1530]
variable_sip_user_agent: [Ekiga/3.2.0]
variable_sip_via_host: [10.0.255.251]
variable_sip_via_port: [5065]
variable_sip_via_rport: [5065]
variable_max_forwards: [70]
variable_presence_id: [mikael-ekiga@fs.voip.domain.com]
variable_switch_r_sdp: [v=0
o=- 1239868973 1239868973 IN IP4 10.0.255.251
s=Opal SIP Session
c=IN IP4 10.0.255.251
t=0 0
m=audio 5090 RTP/AVP 9 8 117 0 116 101 120
a=rtpmap:9 G722/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:117 Speex/16000/1
a=fmtp:117 sr=16000,mode=any
a=rtpmap:0 PCMU/8000/1
a=rtpmap:116 Speex/8000/1
a=fmtp:116 sr=8000,mode=any
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:120 NSE/8000
a=fmtp:120 192-193
m=video 5092 RTP/AVP 119 31
a=rtpmap:119 theora/90000
a=fmtp:119
delivery-method="in_band";height=576;sampling="YCbCr-4:2:0";width=704
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
]
variable_ep_codec_string:
[G722@8000h@0i,PCMA@8000h@0i,SPEEX@16000h@0i,SPEEX@16000h@0i,SPEEX@16000h@0i,PCMU@8000h@0i,H261@90000h@0i]
variable_hangup_after_bridge: [false]
variable_continue_on_fail: [true]
variable_dialed_user: [mikael-nokia]
variable_dialed_domain: [fs.voip.domain.com]
variable_switch_m_sdp: [v=0
o=Nokia-SIPUA 603233522614072812 292890395656351010 IN IP4
10.247.3.253
s=FreeSWITCH
c=IN IP4 10.247.3.253
t=0 0
m=audio 49152 RTP/AVP 8 101 13
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=rtpmap:13 CN/8000/1
a=ptime:20
a=maxptime:200
m=video 0 RTP/AVP 99
a=rtpmap:99 H264/90000
]
variable_remote_media_ip: [10.0.255.251]
variable_remote_media_port: [5090]
variable_read_codec: [G722]
variable_read_rate: [16000]
variable_write_codec: [G722]
variable_write_rate: [16000]
variable_video_possible: [true]
variable_remote_video_ip: [10.0.255.251]
variable_remote_video_port: [5092]
variable_sip_video_fmtp: [CIF=1;QCIF=1]
variable_sip_video_pt: [31]
variable_local_media_ip: [10.100.4.192]
variable_local_media_port: [56008]
variable_local_video_ip: [10.100.4.192]
variable_local_video_port: [59022]
variable_video_read_codec: [H261]
variable_video_read_rate: [90000]
variable_video_write_codec: [H261]
variable_video_write_rate: [90000]
variable_endpoint_disposition: [ANSWER]
variable_originate_disposition: [SUCCESS]
variable_bridge_channel:
[sofia/internal/sip:mikael-nokia@10.247.3.253]
variable_bridge_uuid: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
variable_signal_bond: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
variable_current_application: [info]
How do I get the "raw" hangup cause first mentioned below?
"
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup()
Channel
sofia/internal/sip:mikael-nokia@10.247.3.253 hanging up,
cause:
MEDIA_TIMEOUT
"
As mentioned earlier the origination was in fact a success,
but since I
moved out of wi-fi coverage area I got a MEDIA_TIMEOUT which
should
trigger a transfer to my cell phone number. :-)
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
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