NO Audio Issue when calling SIP to SIP

pacman*CLI> core show version
Asterisk 1.4.20.1 built by root @ pacman on a i386 running FreeBSD on 2008-07-10 18:40:08 UTC

what it does is something like, receiving a call, splitting on an extension, ask for a 5sec message (Record), then dial to the correct extension with a Macro that will ask this extension if they want to answer to this person… press something for yes, hang up for no…

if yes, talk and then… bye bye

if no, to voicemail… and then bye bye

Everything worked just fine while I was testing throught gtalk calling my extension. you can still hear the HOLD while the sip phone answers…

but then if i try to call from a sip phone (internal)… when I answer to my extension i hear the HOLD music stopping… if I decide to answer… everything is ok… but if not… the call after a few hangs… with no other sound from the caller SIP phone, I should hear the voicemail… but nothing. and I see from the logs…

-- Executing [answer@unique_dep:1] Playback("SIP/200-08744000", "tt-weasels") in new stack
-- <SIP/200-08744000> Playing 'tt-weasels' (language 'en')
-- Executing [answer@unique_dep:2] Record("SIP/200-08744000", "reason-call%d.ulaw|3|5") in new stack
-- <SIP/200-08744000> Playing 'beep' (language 'en')
-- Executing [answer@unique_dep:3] Playback("SIP/200-08744000", "transfer") in new stack
-- <SIP/200-08744000> Playing 'transfer' (language 'en')
-- Executing [answer@unique_dep:4] Set("SIP/200-08744000", "y=0") in new stack
-- Executing [answer@unique_dep:5] GotoIf("SIP/200-08744000", "1?6:15") in new stack
-- Goto (unique_dep,answer,6)
-- Executing [answer@unique_dep:6] ChanIsAvail("SIP/200-08744000", "SIP/206") in new stack
-- Executing [answer@unique_dep:8] Dial("SIP/200-08744000", "SIP/206|300|gmM(callback|1217364726.0|reason-call42)") in new stack
-- Called 206
-- Started music on hold, class 'default', on SIP/200-08744000
-- SIP/206-08771000 is ringing
-- SIP/206-08771000 answered SIP/200-08744000
-- Executing [s@macro-callback:1] Set("SIP/206-08771000", "fromWho=1217364726.0") in new stack
-- Executing [s@macro-callback:2] Set("SIP/206-08771000", "daReason=reason-call42") in new stack
-- Executing [s@macro-callback:3] SetGlobalVar("SIP/206-08771000", "RISPOSTA=NO") in new stack

== Setting global variable ‘RISPOSTA’ to ‘NO’
– Executing [s@macro-callback:4] Playback(“SIP/206-08771000”, “tt-monkeysintro”) in new stack
– <SIP/206-08771000> Playing ‘tt-monkeysintro’ (language ‘en’)
– Executing [s@macro-callback:5] Playback(“SIP/206-08771000”, “reason-call42”) in new stack
– <SIP/206-08771000> Playing ‘reason-call42’ (language ‘en’)
– Executing [s@macro-callback:6] Read(“SIP/206-08771000”, “RISPOSTAA|tt-somethingwrong|1||5|60”) in new stack
– Accepting a maximum of 1 digits.
– <SIP/206-08771000> Playing ‘tt-somethingwrong’ (language ‘en’)
– User disconnected
== Spawn extension (macro-callback, s, 6) exited non-zero on ‘SIP/206-08771000’ in macro ‘callback’
– Stopped music on hold on SIP/200-08744000
– Executing [answer@unique_dep:10] Set(“SIP/200-08744000”, “EXTEN=answer”) in new stack
– Executing [answer@unique_dep:11] Goto(“SIP/200-08744000”, “sw-4-ANSWER|10”) in new stack
– Goto (unique_dep,sw-4-ANSWER,10)
– Executing [sw-4-ANSWER@unique_dep:10] Set(“SIP/200-08744000”, “EXTEN=sw-4-ANSWER”) in new stack
– Executing [sw-4-ANSWER@unique_dep:11] Goto(“SIP/200-08744000”, “sw-5-NO|10”) in new stack
– Goto (unique_dep,sw-5-NO,10)
– Executing [sw-5-NO@unique_dep:10] Goto(“SIP/200-08744000”, “unique_dep|noanswer|1”) in new stack
– Goto (unique_dep,noanswer,1)
– Executing [noanswer@unique_dep:1] Set(“SIP/200-08744000”, “call_status=vm”) in new stack
– Executing [noanswer@unique_dep:2] Wait(“SIP/200-08744000”, “1”) in new stack
– Executing [noanswer@unique_dep:3] VoiceMail(“SIP/200-08744000”, “206@tngbox”) in new stack
– <SIP/200-08744000> Playing ‘vm-intro’ (language ‘en’)
– <SIP/200-08744000> Playing ‘beep’ (language ‘en’)
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/tngbox/206/tmp/mg8M0j format: wav49, 0x873d600
[2008-07-29 13:52:45] WARNING[17919]: app.c:611 __ast_play_and_record: No audio available on SIP/200-08744000??
– User hung up
– Recording was 0 seconds long but needs to be at least 3 - abandoning
== Spawn extension (unique_dep, noanswer, 3) exited non-zero on ‘SIP/200-08744000’
– Executing [h@unique_dep:1] GotoIf(“SIP/200-08744000”, “1?2:6”) in new stack
– Goto (unique_dep,h,2)
– Executing [h@unique_dep:2] Set(“SIP/200-08744000”, “EXTEN=h”) in new stack
– Executing [h@unique_dep:3] Goto(“SIP/200-08744000”, “sw-7-FAILED|10”) in new stack

do you have any idea?? its driving me nut

sip.conf


[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
dtmfmode=rfc2833
relaxdtmf=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
maxexpirey=30
defaultexpirey=180
canreinvite=no
nat=0
UserAgent=Asterisk


[200]
type=friend
callerid="XXXX XXXXX" <200>
username=200
host=dynamic
secret=xxxxxxxxx
regcontext=sip_outgoing
regexten=200
insecure=invite
canreinvite=yes
nat=yes
qualify=yes
context=sip_outgoing
pickupgroup=1
callgroup=1
mailbox=200@xxxxx

[206]
type=friend
callerid="XXXX XXXXX" <206>
username=206
host=dynamic
secret=xxxxxxxxx
regcontext=sip_outgoing
regexten=206
insecure=invite
canreinvite=yes
nat=yes
qualify=yes
context=sip_outgoing
pickupgroup=1
callgroup=1
mailbox=206@xxxxx

thank you very much in advance…

i solved the problem


[200]
type=friend
callerid="XXXX XXXX" <200>
username=200
host=dynamic
secret=xxxxxxxxxxxx
;regcontext=sip_outgoing
regexten=200
insecure=no
canreinvite=no
nat=no
qualify=yes
context=sip_outgoing
mailbox=200@tngbox

[206]
type=friend
callerid="XXXX XXXX" <206>
username=206
host=dynamic
secret=xxxxxxxxxx
;regcontext=sip_outgoing
regexten=206
insecure=no
canreinvite=no
nat=no
qualify=yes
context=sip_outgoing
mailbox=206@tngbox