[asterisk-dev] jingle audio
Clod Patry
cpatry at gmail.com
Thu Jul 27 19:36:48 MST 2006
im trying to see why audio isn't passing thru chan_jingle when i call from a
google talk client.
on my dialplan, im just making a Dial(SIP/10), where 10 is my polycom phone.
this is what i get:
JABBER: asterisk INCOMING: <iq to="cpatry at gmail.com/asterisk79A5E541"
type="set" id="33" from="juliedube01 at gmail.com/Talk.v921D8E248B"><session
type="initiate" id="471827166" initiator="
juliedube01 at gmail.com/Talk.v921D8E248B"
xmlns="http://www.google.com/session"><description
xmlns="http://www.google.com/session/phone"><payload-type id="103"
name="ISAC"/><payload-type id="97" name="IPCMWB"/><payload-type id="4"
name="G723"/><payload-type id="100" name="EG711U"/><payload-type id="101"
name="EG711A"/><payload-type id="0" name="PCMU"/><payload-type id="8"
name="PCMA"/><payload-type id="13" name="CN"/><payload-type id="102"
name="iLBC"/><payload-type id="117" name="red"/><payload-type id="106"
name="audio/telephone-event"/></description></session></iq>
Jul 27 22:19:32 DEBUG[5288]: chan_jingle.c:647 jingle_alloc: The client is
guest for alloc
caster*CLI>
JABBER: asterisk OUTGOING: <iq type='result' from='
cpatry at gmail.com/asterisk79A5E541' to='
juliedube01 at gmail.com/Talk.v921D8E248B' id='33'/>
caster*CLI>
JABBER: asterisk OUTGOING: <iq from='cpatry at gmail.com/asterisk79A5E541' to='
juliedube01 at gmail.com/Talk.v921D8E248B' type='set' id='aaaae'><session
type='candidates' id='471827166' initiator='
juliedube01 at gmail.com/Talk.v921D8E248B' xmlns='
http://www.google.com/session'><candidate name='rtp' address='127.0.1.1'
port='17744' username='034cba614b4c73e8' password='337eb98b143b5bc0'
preference='1.00' protocol='udp' type='local' network='0'
generation='0'/></session></iq>
-- Executing [s at polycom:1] Answer("Jingle/juliedube01-c12b", "") in new
stack
Jul 27 22:19:32 DEBUG[5499]: chan_jingle.c:398 jingle_answer: Answer!
caster*CLI>
JABBER: asterisk OUTGOING: <iq type='set' to='
juliedube01 at gmail.com/Talk.v921D8E248B' id='aaaaf'><session xmlns='
http://www.google.com/session' type='accept' initiator='
juliedube01 at gmail.com/Talk.v921D8E248B' id='471827166'><description xmlns='
http://www.google.com/session/phone'><payload-type id='0' name='PCMU'
xmlns='http://www.google.com/session/phone'/><payload-type id='100'
name='EG711U' xmlns='http://www.google.com/session/phone'/><payload-type
id='117' name='red' xmlns='http://www.google.com/session/phone'/><payload-type
id='106' name='audio/telephone-event' xmlns='
http://www.google.com/session/phone'/><payload-type id='13' name='CN'
xmlns='http://www.google.com/session/phone'/></description></session></iq>
Jul 27 22:19:32 DEBUG[5284]: channel.c:873 channel_find_locked: Avoiding
initial deadlock for channel '0x81fd810'
-- Executing [s at polycom:2] Dial("Jingle/juliedube01-c12b", "SIP/10") in
new stack
Jul 27 22:19:32 DEBUG[5499]: acl.c:213 ast_apply_ha: ##### Testing
192.168.1.10 with 192.168.0.0
-- Called 10
caster*CLI>
JABBER: asterisk INCOMING: <iq to="cpatry at gmail.com/asterisk79A5E541"
type="set" id="35" from="juliedube01 at gmail.com/Talk.v921D8E248B"><session
type="candidates" id="471827166" initiator="
juliedube01 at gmail.com/Talk.v921D8E248B"
xmlns="http://www.google.com/session"><candidate
name="rtp" address="192.168.1.126" port="2023" username="vk1kh7qZI10Y3ll4"
password="1LVHvwyHpSY26BiT" preference="1" protocol="udp" type="local"
network="0" generation="0"/></session></iq>
Jul 27 22:19:32 DEBUG[5288]: chan_jingle.c:1331 jingle_parser: About to add
candidate!
JABBER: asterisk OUTGOING: <iq type='result' from='
cpatry at gmail.com/asterisk79A5E541' to='
juliedube01 at gmail.com/Talk.v921D8E248B' id='35'/>
Jul 27 22:19:32 DEBUG[5288]: chan_jingle.c:1333 jingle_parser: Candidate
Added!
caster*CLI>
JABBER: asterisk INCOMING: <iq to="cpatry at gmail.com/asterisk79A5E541"
type="set" id="36" from="juliedube01 at gmail.com/Talk.v921D8E248B"><session
type="candidates" id="471827166" initiator="
juliedube01 at gmail.com/Talk.v921D8E248B"
xmlns="http://www.google.com/session"><candidate
name="rtp" address="70.81.175.205" port="2024" username="5J/dzRIefDQoTt9f"
password="3oX373f/FxiftjOd" preference="0.9" protocol="udp" type="stun"
network="0" generation="0"/></session></iq>
Jul 27 22:19:32 DEBUG[5288]: chan_jingle.c:1331 jingle_parser: About to add
candidate!
JABBER: asterisk OUTGOING: <iq type='result' from='
cpatry at gmail.com/asterisk79A5E541' to='
juliedube01 at gmail.com/Talk.v921D8E248B' id='36'/>
Jul 27 22:19:32 DEBUG[5288]: chan_jingle.c:1333 jingle_parser: Candidate
Added!
-- SIP/10-08203f58 is ringing
Jul 27 22:19:32 NOTICE[5499]: chan_jingle.c:1129 jingle_indicate: Don't know
how to indicate condition '3'
Jul 27 22:19:32 DEBUG[5499]: channel.c:2187 ast_indicate_data: Driver for
channel 'Jingle/juliedube01-c12b' does not support indication 3, emulating
it
caster*CLI>
JABBER: asterisk INCOMING: <iq type="result" to="
cpatry at gmail.com/asterisk79A5E541" id="aaaae" from="
juliedube01 at gmail.com/Talk.v921D8E248B"/>
caster*CLI>
JABBER: asterisk INCOMING: <iq type="result" to="
cpatry at gmail.com/asterisk79A5E541" id="aaaaf" from="
juliedube01 at gmail.com/Talk.v921D8E248B"/>
[i pressed answer here, on the polycom]
Jul 27 22:19:35 DEBUG[5308]: chan_sip.c:1995 __sip_ack: Acked pending invite
102
-- SIP/10-08203f58 answered Jingle/juliedube01-c12b
Jul 27 22:19:35 NOTICE[5499]: chan_jingle.c:1129 jingle_indicate: Don't know
how to indicate condition '-1'
Jul 27 22:19:35 WARNING[5499]: rtp.c:2520 ast_rtp_bridge: Can't find native
functions for channel 'Jingle/juliedube01-c12b'
-- Native bridging Jingle/juliedube01-c12b and SIP/10-08203f58 ended
Jul 27 22:19:40 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
192.168.1.3 with 192.168.0.0
Jul 27 22:19:40 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
24.37.198.131 with 192.168.0.0
Jul 27 22:19:40 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
192.168.1.3 with 192.168.0.0
[i had no audio, just making a show channel to see if its bridged correctly]
caster*CLI> show channels
Channel Location State Application(Data)
SIP/10-08203f58 (None) Up Bridged
Call(Jingle/juliedube0
Jingle/juliedube01-c s at polycom:2 Up Dial(SIP/10)
2 active channels
1 active call
caster*CLI>
JABBER: asterisk INCOMING: <iq to="cpatry at gmail.com/asterisk79A5E541"
type="set" id="37" from="juliedube01 at gmail.com/Talk.v921D8E248B"><session
type="candidates" id="471827166" initiator="
juliedube01 at gmail.com/Talk.v921D8E248B"
xmlns="http://www.google.com/session"><candidate
name="rtp" address="192.168.1.126" port="2026" username="EwzANAI7zI+pzjcK"
password="hlvEbGin8DP1RBVi" preference="1" protocol="udp" type="local"
network="0" generation="1"/></session></iq>
Jul 27 22:19:52 DEBUG[5288]: chan_jingle.c:1331 jingle_parser: About to add
candidate!
JABBER: asterisk OUTGOING: <iq type='result' from='
cpatry at gmail.com/asterisk79A5E541' to='
juliedube01 at gmail.com/Talk.v921D8E248B' id='37'/>
Jul 27 22:19:52 DEBUG[5288]: chan_jingle.c:1333 jingle_parser: Candidate
Added!
caster*CLI>
JABBER: asterisk INCOMING: <iq to="cpatry at gmail.com/asterisk79A5E541"
type="set" id="38" from="juliedube01 at gmail.com/Talk.v921D8E248B"><session
type="candidates" id="471827166" initiator="
juliedube01 at gmail.com/Talk.v921D8E248B"
xmlns="http://www.google.com/session"><candidate
name="rtp" address="70.81.175.205" port="2027" username="RzrYI/z+VcT6V/A0"
password="f2bdyUBxrOd58lBT" preference="0.9" protocol="udp" type="stun"
network="0" generation="1"/></session></iq>
Jul 27 22:19:52 DEBUG[5288]: chan_jingle.c:1331 jingle_parser: About to add
candidate!
JABBER: asterisk OUTGOING: <iq type='result' from='
cpatry at gmail.com/asterisk79A5E541' to='
juliedube01 at gmail.com/Talk.v921D8E248B' id='38'/>
Jul 27 22:19:52 DEBUG[5288]: chan_jingle.c:1333 jingle_parser: Candidate
Added!
Jul 27 22:19:54 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
192.168.1.3 with 192.168.0.0
caster*CLI>
caster*CLI>
[i pressed hangup here on the polycom]
Jul 27 22:20:06 DEBUG[5499]: channel.c:3433 ast_generic_bridge: Didn't get a
frame from channel: SIP/10-08203f58
Jul 27 22:20:06 DEBUG[5499]: channel.c:3720 ast_channel_bridge: Bridge stops
bridging channels Jingle/juliedube01-c12b and SIP/10-08203f58
== Spawn extension (polycom, s, 2) exited non-zero on
'Jingle/juliedube01-c12b'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is ''
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is ''
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is 's'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is 'polycom'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is
'Jingle/juliedube01-c12b'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is 'SIP/10-08203f58'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is 'Dial'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is 'SIP/10'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is '2006-07-27
22:19:32'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is '2006-07-27
22:19:32'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is '2006-07-27
22:20:06'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is '34'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is '34'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is 'ANSWERED'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is ''
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is '1154053172.4'
Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
pbx_substitute_variables_helper_full: Function result is ''
caster*CLI>
JABBER: asterisk OUTGOING: <iq type='set' from='
cpatry at gmail.com/asterisk79A5E541' to='
juliedube01 at gmail.com/Talk.v921D8E248B' id='aaaag'><session type='terminate'
id='471827166' initiator='juliedube01 at gmail.com/Talk.v921D8E248B' xmlns='
http://www.google.com/session'/></iq>
caster*CLI>
JABBER: asterisk INCOMING: <iq type="result" to="
cpatry at gmail.com/asterisk79A5E541" id="aaaag" from="
juliedube01 at gmail.com/Talk.v921D8E248B"/>
caster*CLI>
JABBER: asterisk INCOMING: <iq to="cpatry at gmail.com/asterisk79A5E541"
type="set" id="39" from="juliedube01 at gmail.com/Talk.v921D8E248B"><session
type="terminate" id="471827166" initiator="
juliedube01 at gmail.com/Talk.v921D8E248B" xmlns="http://www.google.com/session
"/></iq>
Jul 27 22:20:06 DEBUG[5288]: chan_jingle.c:501 jingle_hangup_farend: The
client is guest
Jul 27 22:20:06 NOTICE[5288]: chan_jingle.c:512 jingle_hangup_farend: Whoa,
didn't find call!
JABBER: asterisk OUTGOING: <iq type='result' from='
cpatry at gmail.com/asterisk79A5E541' to='
juliedube01 at gmail.com/Talk.v921D8E248B' id='39'/>
Jul 27 22:20:08 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
192.168.1.3 with 192.168.0.0
Jul 27 22:20:08 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
24.37.198.131 with 192.168.0.0
Jul 27 22:20:08 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
192.168.1.3 with 192.168.0.0
Jul 27 22:20:10 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
24.37.198.131 with 192.168.0.0
Jul 27 22:20:10 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
192.168.1.3 with 192.168.0.0
Jul 27 22:20:10 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
192.168.1.10 with 192.168.0.0
caster*CLI>
But even if im just doing a SayDigits(1241), im not hearing it from the
google talk client.
Any idea on how to solve this issue? Is a problem with RTP and chan_jingle
or anything i forgot from my side?
--
Clod Patry
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20060727/40a14288/attachment.htm
More information about the asterisk-dev
mailing list