asterisk users - Jan 2006 - ZAP <--> sip(polycom301) can not hear each other

please help!!!
  
  I am dialing into our asterisk server(TDM400p) from the psnt. I hear  our
voicemail message and I press the extention 1000. The Polycom ip  phone in the
office rings. I pickup but neither side can hear one  another. What have I done
wrong?
  
  thanks
  
  sip.conf:
  [general]
  context=local-access                  ; Default context for incoming calls
  bindport=5060                    ; UDP Port to bind to (SIP standard port is
5060)
  bindaddr=0.0.0.0                 ; IP address to bind to (0.0.0.0 binds to
all)
  srvlookup=yes                    ; Enable DNS SRV lookups on outbound calls
  musicclass=default
  
  [authentication]
  
  [1000]
  username=1000
  regexten=1000
  mailbox=1000@voicemail
  callerid="jon Smith" <1000>
  context=local-access
  nat=yes
  secret=password
  type=friend
  host=dynamic
  canreinvite=yes
  disallow=all
  allow=all
  
  [1001]
  username=1001
  regexten=1001
  mailbox=1001@voicemail
  callerid="jane doe" <1001>
  context=local-access
  nat=yes
  secret=password
  type=friend
  host=dynamic
  canreinvite=yes
  disallow=all
  allow=all
  
  extensions.conf:
  [general]
  static=yes
  writeprotect=no
  autofallthrough=yes
  clearglobalvars=no
  priorityjumping=no
  
  [globals]
  ATTENDANT=1001
  OUTBOUNDTRUNK=ZAP/g1
  
  [extentions]
  exten => _10XX,1,Ringing
  exten => _10XX,2,Dial(SIP/${EXTEN},20)
  exten => _10XX,3,Answer
  exten => _10XX,4,VoiceMail(u${EXTEN}@voicemail)
  exten => _10XX,5,Hangup
  
  [voicemail]
  exten => _910XX,1,Wait(1)
  exten => _910XX,2,VoiceMailMain(${EXTEN:1}@voicemail)
  
  [local]
  include => extentions
  include => voicemail
  
  [incoming]
  exten => s,1,Answer
  exten => s,2,Background(our-voicemail-sound)
  exten => t,1,Playback(vm-goodbye)
  exten => t,2,Hangup( )
  exten => 0,1,Dial(SIP/${ATTENDANT},20)
  exten => 1,1,Directory(voicemail,internal,f)
  exten => 2,1,Directory(voicemail,internal)
  include => extentions
  
  [local-trunks]
  exten => _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
  exten => _9XXXXXXXXXX,2,Congestion( )
  exten => _9XXXXXXXXXX,102,Congestion( )
  exten => 911,1,Dial(${OUTBOUNDTRUNK}/911)
  exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)
  
  [local-access]
  ignorepat => 9
  include => local
  include => local-trunks
  
  
  zapata.conf:
  
  [trunkgroups]
  [channels]
  context=default
  switchtype=national
  signalling=fxo_ls
  rxwink=300              ; Atlas seems to use long (250ms) winks
  usecallerid=yes
  hidecallerid=no
  callwaiting=yes
  usecallingpres=yes
  callwaitingcallerid=yes
  threewaycalling=yes
  transfer=yes
  canpark=yes
  cancallforward=yes
  callreturn=yes
  echocancel=yes
  echocancelwhenbridged=yes
  rxgain=0.0
  txgain=0.0
  group=1
  callgroup=1
  pickupgroup=1
  immediate=no
  group=1
  echocancel=yes
  switchtype=national
  signalling=fxs_ks
  context=incoming
  echocancelwhenbridged=yes
  channel => 1-4
  
  
  /etc/zaptel.conf:
  fxsks=1,2,3,4
  loadzone = us
  defaultzone=us
  
  log:
  Asterisk Ready.
      -- Starting simple switch on 'Zap/1-1'
  Jan 31 15:55:28 NOTICE[2525]: chan_zap.c:6040 ss_thread: Got event 18 (Ring
Begin)...
  Jan 31 15:55:29 ERROR[2525]: callerid.c:276 callerid_feed: fsk_serie made
mylen < 0 (-155)
  Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6070 ss_thread: CallerID feed
failed: Success
  Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6114 ss_thread: CallerID returned
with error on channel 'Zap/1-1'
      -- Executing Answer("Zap/1-1", "") in new stack
      -- Executing BackGround("Zap/1-1",
"our-voicemail-sound") in new stack
      -- Playing 'our-voicemail-sound' (language 'en')
    == CDR updated on Zap/1-1
      -- Executing Ringing("Zap/1-1", "") in new stack
      -- Executing Dial("Zap/1-1", "SIP/1000|20") in new
stack
      -- Called 1000
      -- SIP/1000-54e4 is ringing
      -- SIP/1000-54e4 answered Zap/1-1
    == Spawn extension (incoming, 1000, 2) exited non-zero on 'Zap/1-1'
      -- Hungup 'Zap/1-1'
  
  

		
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>From your description, it sounds as if the SIP phones are local to the
Asterisk box.  If this is so, having "nat=yes" might be a problem.

-Ken

sdgesa gaeharth wrote:
> please help!!!
>
> I am dialing into our asterisk server(TDM400p) from the psnt. I hear
> our voicemail message and I press the extention 1000. The Polycom ip
> phone in the office rings. I pickup but neither side can hear one
> another. What have I done wrong?
>
> thanks
>
> sip.conf:
> [general]
> context=local-access                 ; Default context for incoming calls
> bindport=5060                   ; UDP Port to bind to (SIP standard
> port is 5060)
> bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds
> to all)
> srvlookup=yes                   ; Enable DNS SRV lookup s on outbound
> calls
> musicclass=default
>
> [authentication]
>
> [1000]
> username=1000
> regexten=1000
> mailbox=1000@voicemail
> callerid="jon Smith" <1000>
> context=local-access
> nat=yes
> secret=password
> type=friend
> host=dynamic
> canreinvite=yes
> disallow=all
> allow=all
>
> [1001]
> username=1001
> regexten=1001
> mailbox=1001@voicemail
> callerid="jane doe" <1001>
> context=local-access
> nat=yes
> secret=password
> type=friend
> host=dynamic
> canreinvite=yes
> disallow=all
> allow=all
>
> extensions.conf:
> [general]
> static=yes
> writeprotect=no
> autofallthrough=yes
> clearglobalvars=no
> priorityjumping=no
>
> [globals]
> ATTENDANT=1001
> OUTBOUNDTRUNK=ZAP/g1
>
> [extentions]
> exten => _10XX,1,Ringing
> exten => _10XX,2,Dial(SIP/${EXTEN},20)
> exten => _10XX,3,Answer
> exten => _10XX,4,VoiceMail(u${EXTEN}@voicemail)
> exten => _10XX,5,Hangup
>
> [voicemail]
> exten => _910XX,1,Wait(1)
> exten => _910XX,2,VoiceMailMain(${EXTEN:1}@voicemail)
>
> [local]
> include => extentions
> include => voicemail
>
> [incoming]
> exten => s,1,Answer
> exten => s,2,Background(our-voicemail-sound)
> exten => t,1,Playback(vm-goodbye)
> exten => t,2,Hangup( )
> exten => 0,1,Dial(SIP/${ATTENDANT},20)
> exten => 1,1,Directory(voicemail,internal,f)
> exten => 2,1,Directory(voicemail,internal)
> include => extentions
>
> [local-trunks]
> exten => _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> exten => _9XXXXXXXXXX,2,Congestion( )
> exten => _9XXXXXXXXXX,102,Congestion( )
> exten => 911,1,Dial(${OUTBOUNDTRUNK}/911)
> exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)
>
> [local-access]
> ignorepat => 9
> include => local
> include => local-trunks
>
>
> zapata.conf:
>
> [trunkgroups]
> [channels]
> context=default
> switchtype=national
> signalling=fxo_ls
> rxwink=300              ; Atlas seems to use long (250ms) winks
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
> group=1
> echocancel=yes
> switchtype=national
> signalling=fxs_ks
> context=incoming
> echocancelwhenbridged=yes
> channel => 1-4
>
>
> /etc/zaptel.conf:
> fxsks=1,2,3,4
> loadzone = us
> defaultzone=us
>
> log:
> Asterisk Ready.
>     -- Star ting simple switch on 'Zap/1-1'
> Jan 31 15:55:28 NOTICE[2525]: chan_zap.c:6040 ss_thread: Got event 18
> (Ring Begin)...
> Jan 31 15:55:29 ERROR[2525]: callerid.c:276 callerid_feed: fsk_serie
> made mylen < 0 (-155)
> Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6070 ss_thread: CallerID
> feed failed: Success
> Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6114 ss_thread: CallerID
> returned with error on channel 'Zap/1-1'
>     -- Executing Answer("Zap/1-1", "") in new stack
>     -- Executing BackGround("Zap/1-1",
"our-voicemail-sound") in new stack
>     -- Playing 'our-voicemail-sound' (language 'en')
>   == CDR updated on Zap/1-1
>     -- Executing Ringing("Zap/1-1", "") in new stack
>     -- Executing Dial("Zap/1-1", "SIP/1000|20") in new
stack
>     -- Called 1000
>     -- SIP/1000-54e4 is ringing
>     -- SIP/1000-54e4 an swered Zap/1-1
>   == Spawn extension (incoming, 1000, 2) exited non-zero on
'Zap/1-1'
>     -- Hungup 'Zap/1-1'
>
> ------------------------------------------------------------------------
> Bring words and photos together (easily) with
> PhotoMail
>
<http://us.rd.yahoo.com/mail_us/taglines/PMHM3/*http://photomail.mail.yahoo.com>
> - it's free and works with Yahoo! Mail.
>
>------------------------------------------------------------------------
>
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>

Anyone????? This has been killing  me for days!!!!
  
  
  thanks

sdgesa gaeharth <pollux1234567890@yahoo.com> wrote:  That is correct, The
SIP phones are all on our  LAN. I changed the nat's to say no, but I still
get the same problem.  Another thing, when I call out to the pstn from our local
sip phones.  The same problem happens.  The outid line rings, the person picks 
p but no sounds.
  
   Any suggestions????
  
  thanks

Ken D'Ambrosio <ken@jots.org> wrote:  >From your description, it
sounds as if the SIP phones are local to the
Asterisk box.  If this is so, having "nat=yes" might be a problem.

-Ken

sdgesa gaeharth wrote:
> please help!!!
>
> I am dialing into our asterisk server(TDM400p) from the psnt. I hear
> our voicemail message and I press the extention 1000. The Polycom ip
> phone in the office rings. I pickup but neither side can hear one
>   another. What have I done wrong?
>
> thanks
>
> sip.conf:
> [general]
> context=local-access                 ; Default context for incoming calls
> bindport=5060                   ; UDP Port to bind to (SIP standard
> port is 5060)
> bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds
> to all)
> srvlookup=yes                   ; Enable DNS SRV lookup s on outbound
> calls
> musicclass=default
>
> [authentication]
>
> [1000]
> username=1000
> regexten=1000
> mailbox=1000@voicemail
> callerid="jon Smith" <1000>
> context=local-access
> nat=yes
> secret=password
> type=friend
> host=dynamic
> canreinvite=yes
> disallow=all
> allow=all
>
> [1001]
> username=1001
> regexten=1001
> mailbox=1001@voicemail
> callerid="jane doe" <1001>
>   context=local-access
> nat=yes
> secret=password
> type=friend
> host=dynamic
> canreinvite=yes
> disallow=all
> allow=all
>
> extensions.conf:
> [general]
> static=yes
> writeprotect=no
> autofallthrough=yes
> clearglobalvars=no
> priorityjumping=no
>
> [globals]
> ATTENDANT=1001
> OUTBOUNDTRUNK=ZAP/g1
>
> [extentions]
> exten => _10XX,1,Ringing
> exten => _10XX,2,Dial(SIP/${EXTEN},20)
> exten => _10XX,3,Answer
> exten => _10XX,4,VoiceMail(u${EXTEN}@voicemail)
> exten => _10XX,5,Hangup
>
> [voicemail]
> exten => _910XX,1,Wait(1)
> exten => _910XX,2,VoiceMailMain(${EXTEN:1}@voicemail)
>
> [local]
> include => extentions
> include => voicemail
>
> [incoming]
> exten => s,1,Answer
> exten =>   s,2,Background(our-voicemail-sound)
> exten => t,1,Playback(vm-goodbye)
> exten => t,2,Hangup( )
> exten => 0,1,Dial(SIP/${ATTENDANT},20)
> exten => 1,1,Directory(voicemail,internal,f)
> exten => 2,1,Directory(voicemail,internal)
> include => extentions
>
> [local-trunks]
> exten => _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> exten => _9XXXXXXXXXX,2,Congestion( )
> exten => _9XXXXXXXXXX,102,Congestion( )
> exten => 911,1,Dial(${OUTBOUNDTRUNK}/911)
> exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)
>
> [local-access]
> ignorepat => 9
> include => local
> include => local-trunks
>
>
> zapata.conf:
>
> [trunkgroups]
> [channels]
> context=default
> switchtype=national
> signalling=fxo_ls
> rxwink=300              ; Atlas seems to use long (250ms) winks
>   usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
> group=1
> echocancel=yes
> switchtype=national
> signalling=fxs_ks
> context=incoming
> echocancelwhenbridged=yes
> channel => 1-4
>
>
> /etc/zaptel.conf:
> fxsks=1,2,3,4
> loadzone = us
> defaultzone=us
>
> log:
> Asterisk Ready.
>     -- Star ting simple switch on 'Zap/1-1'
> Jan 31 15:55:28 NOTICE[2525]: chan_zap.c:6040 ss_thread: Got event 18
> (Ring Begin)...
> Jan 31 15:55:29 ERROR[2525]: callerid.c:276 callerid_feed: fsk_serie
> made mylen   < 0 (-155)
> Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6070 ss_thread: CallerID
> feed failed: Success
> Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6114 ss_thread: CallerID
> returned with error on channel 'Zap/1-1'
>     -- Executing Answer("Zap/1-1", "") in new stack
>     -- Executing BackGround("Zap/1-1",
"our-voicemail-sound") in new stack
>     -- Playing 'our-voicemail-sound' (language 'en')
>   == CDR updated on Zap/1-1
>     -- Executing Ringing("Zap/1-1", "") in new stack
>     -- Executing Dial("Zap/1-1", "SIP/1000|20") in new
stack
>     -- Called 1000
>     -- SIP/1000-54e4 is ringing
>     -- SIP/1000-54e4 an swered Zap/1-1
>   == Spawn extension (incoming, 1000, 2) exited non-zero on
'Zap/1-1'
>     -- Hungup 'Zap/1-1'
>
> ------------------------------------------------------------------------
> Bring words and photos together (easily) with
>   PhotoMail
> 
> - it's free and works with Yahoo! Mail.
>
>------------------------------------------------------------------------
>
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>
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Hi - 
> That is correct, The SIP phones are all on our  LAN. I changed the
nat's to
> say no, but I still get the same problem.  Another thing, when I call out
to
> the pstn from our local sip phones.  The same problem happens.  The outid
line
> rings, the person picks  p but no sounds.
>   
>    Any suggestions????
I've been away from the list and just saw your thread now.  Although, I
don't see the normal CLI errors, you problem is probably this:
>> disallow=all
>> allow=all
You are trying to use all possible codecs that asterisk knows.  The polycom
phones only know ulaw, alaw and g729.  Try this instead:

disallow=all
allow=ulaw


- Noah