[help] sip calls not picking up

Hello.

When I try to dial out from my * using BroadVoice, the call is connected and I can talk to the other party, but only for 30 seconds or so Asterisk when determines the call has not been answered and hangs up. I’m assuming this means the audio I can hear is what * thinks is ringing.

Incoming calls from BV rings my Zap channel correctly, and * registers it off-hook when I answer, but the caller hears nothing but ringing.

I’m behind NAT, but port 5060, as well as 10000-20000 (listed on BV’s website) are forwarded to the * box. My main internal extension is on a Zaptel FXS card, which is working flawlessly.

With verbosity set to 9, here is what I see in the * console:

    -- Starting simple switch on 'Zap/1-1'
    -- Executing SetCallerID("Zap/1-1", "2063579435") in new stack
    -- Executing Dial("Zap/1-1", "SIP/sip.broadvoice.com/12063548295|30") in new stack
    -- Called sip.broadvoice.com/12063548295
    -- SIP/sip.broadvoice.com-d1ac is making progress passing it to Zap/1-1
  == No one is available to answer at this time (1:0/0/0)
    -- Executing Congestion("Zap/1-1", "") in new stack
  == Spawn extension (internal, 12063548295, 3) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'

Which is all well and good, except the call was answered before it got to [quote]== No one is available to answer at this time[/quote]

Here are the relevant portions of my config files:

sip.conf: (basically taken from BV’s example. I have experimented with various NAT settings, to no avail)

[general]
pedantic=no
externip={HIDDEN}
localnet=10.0.0.0/255.0.0.0
nat=yes 
disallow=all
allow=ulaw
allow=alaw
allow=gsm

context=initial-screen

register => 2063579435:{SECRET}@sip.broadvoice.com



[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=2063579435
secret={SECRET}
username=2063579435
insecure=very
context=initial-screen
authname=2063579435
dtmfmode=inband
dtmf=inband
canreinvite=no
qualify=yes

::snip:: ... internal sip extensions here...

And now Extensions.conf:

[initial-screen]

exten => ${BVNUM}, 1, Wait(3) ;maybe we're too fast
exten => ${BVNUM}, 2, NoOp()
::snip:: ... callerid matching stuff was here. it works correctly... 
exten => ${BVNUM},3,Goto(ring,s,1)

exten => s, 1, Goto(${BVNUM},1)
exten => i, 1, Goto(${BVNUM},1)

exten => t, 1, Goto(ring,s,1)


::snip:: ... per callerid specific stuff

[ring] 
include => ring-weekday|08:00-17:00|Mon-Fri|*|*

include => ring-home|17:00-21:00|Mon-Fri|*|*
include => ring-home|09:00-21:00|Sat-Sun|*|*

include => ring-night|21:00-23:59|Mon-Fri|*|*
include => ring-night|00:00-08:00|Mon-Fri|*|*
include => ring-night|21:00-23:59|Sat-Sun|*|*
include => ring-night|00:00-09:00|Sat-Sun|*|*


[ring-weekday]
exten => s, 1, Answer()
exten => s, 2, Playback(nbdy-avail-to-take-call)
exten => s, 3, Voicemail(102)

[ring-night]
exten => s, 1, Answer()
exten => s, 2, Ringing()
exten => s, 3, Wait(1)
exten => s, 4, Playback(weasels-eaten-phonesys)
exten => s, 5, Wait(5)
;exten => s, 5, Voicemail(102)
exten => s, 6, Hangup()

[ring-home]
exten => s, 1, Dial(SIP/100&SIP/101&ZAP/1,20)
exten => s, 2, Voicemail(102)


[internal]

exten => _1NXXNXXXXXX, 1, SetCallerID(${BVNUM}) 
exten => _1NXXNXXXXXX, 2, dial(SIP/sip.broadvoice.com/${EXTEN},30) 
exten => _1NXXNXXXXXX, 3, congestion() 
exten => _1NXXNXXXXXX, 102, busy()


::snip:: internal extensions

What am I missing to convince everything that the calls are, in fact, being answered?

Any and all help is greatly appreciated.

either remove nat=yes from the sip general config or add nat=no to the broadvoice sip config.

Forgot to mention that I’ve tried all possible combinations of nat= without success. For the sake of things finally working, I just tried again, and the same thing occurrs. Outgoing calls connect and sound just fine until * determines that no one has answered, and incoming calls hear ringing even after I pick up.

I have noticed that the call will be terminated if either party hangs up, both on incoming and outgoing calls.

I also noticed an error message when looking at a sip debug, but it seemed more like a symtom rather than the problem its self:

[code]Retransmitting #6 (no NAT) to 147.135.32.128:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.32.128:5060;received=147.135.32.128
From: "Seattle WA"sip:%number%@147.135.32.128;user=phone;tag=qtuw
To: "ReverendRyan"sip:s@67.183.234.90;user=phone;tag=as00df1e04
Call-ID: 20345-2@147.135.32.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:%BVnumber%@67.183.234.90
Content-Type: application/sdp
Content-Length: 184

v=0
o=root 4910 4910 IN IP4 67.183.234.90
s=session
c=IN IP4 67.183.234.90
t=0 0
m=audio 12342 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -


Aug 7 20:08:46 WARNING[4921]: chan_sip.c:1210 retrans_pkt: Maximum retries exceeded on transmission 20345-2@147.135.32.128 for seqno 1 (Non-critical Response)
[/code]

Thanks again for any help.