[Asterisk-Users] ZAP <--> sip(polycom301) can not hear each other

Wed Feb 1 07:57:46 MST 2006

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

  thanks

sdgesa gaeharth <pollux1234567890 at 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 at 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 at 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 at 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'
>
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