[asterisk-users] Re: Polycom 601 loop
Nathan Bell
nathanb at actarg.com
Mon Mar 26 14:43:52 MST 2007
No loop now, but instead I get this:
Mar 26 15:42:18 NOTICE[1854] app_dial.c: Unable to create channel of
type 'SIP' (cause 3 - No route to destination)
Mar 26 15:42:18 VERBOSE[1854] logger.c: == Everyone is busy/congested
at this time (1:0/0/1)
Mar 26 15:42:18 DEBUG[1854] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.
dave cantera wrote:
> nathan,
> try dial() directly to the extension
>
> [to-sip]
> exten => _X.,1,Dial(SIP/${EXTEN}@192.168.2.13,120)
>
> try
> exten => _X.,1,Dial(SIP/${EXTEN},20)
>
> where ${EXTEN} = 201
> and
> [201] in /etc/sip.conf is
>
> [201]
> type=friend ; Friends place calls and receive calls
> context=from-sip ; Context for incoming calls from this
> user
>
>
> in extensions.conf
> [from-sip]
> exten => 201,1,Wait(1)
> exten => 201,n,Answer()
> exten => 201,n,Dial(SIP/201,15)
> exten => 201,n,VoiceMailMain
> exten => 201,n,Hangup()
>
>
> Nathan Bell wrote:
>
>> Sorry, forgot to attach the sip.conf and extensions.conf files.
>> Attached now.
>> ------------------------------------------------------------------------
>>
>>
>> [general]
>> context=from-sip ; Default context for incoming calls
>> ; if asterisk was compiled with OSP support.
>> realm=actarg.com ; Realm for digest authentication
>> ; defaults to "asterisk"
>> ; Realms MUST be globally unique according to RFC 3261
>> ; Set this to your host name or domain name
>> 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
>> ; Note: Asterisk only uses the first host
>> ; in SRV records
>> ; Disabling DNS SRV lookups disables the
>> ; ability to place SIP calls based on domain
>> ; names to some other SIP users on the Internet
>> autodomain=yes ; Turn this on to have Asterisk add local host
>> ; name and local IP to domain list.
>> ; and multiline formatted headers for strict
>> qualify=yes
>> disallow=all
>> allow=ulaw ; dtmfmode=inband only works with ulaw
>> or alaw!
>> progressinband=no ; Polycom phones don't work properly
>> with "never"
>> dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
>> nat=no ; there is not NAT between phone and
>> Asterisk
>> canreinvite=no ; disallow RTP voice traffic to bypass
>> Asterisk
>>
>> [201]
>> type=friend ; Friends place calls and receive calls
>> context=from-sip ; Context for incoming calls from this
>> user
>> secret=asteriskpassword
>> host=dynamic ; This peer register with us
>> callerid=John Doe <201>
>>
>> [202]
>> type=friend ; Friends place calls and receive calls
>> context=from-sip ; Context for incoming calls from this
>> user
>> secret=asteriskpassword
>> host=dynamic ; This peer register with us
>> callerid=Jane Doe <202>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> ; from outside T1
>> [from-ptsn]
>> exten => s,1,Answer()
>> include => cac-ext
>> include => sip-ext
>> include => intertel-ext
>> exten => t,1,Playback(vm-goodbye)
>> exten => t,2,Hangup()
>>
>> ; from sip lines
>> [from-sip]
>> include => internal
>>
>> ; generic interal route
>> [internal]
>> exten => s,1,Answer()
>> include => cac-ext
>> include => sip-ext
>> include => intertel-ext
>> include => to-ptsn
>>
>> ; check if extension is to sip
>> [sip-ext]
>> exten => _20X,1,Goto(to-sip,${EXTEN},1)
>>
>> ; send call to sip
>> [to-sip]
>> exten => _X.,1,Dial(SIP/${EXTEN}@192.168.2.13,120)
>> exten => _X.,2,Playback(vm-nobodyavail)
>> exten => _X.,3,Hangup()
>> exten => _X.,102,Playback(tt-allbusy)
>> exten => _X.,103,Hangup()
>>
>>
>> ------------------------------------------------------------------------
>>
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>> ------------------------------------------------------------------------
>>
>> No virus found in this incoming message.
>> Checked by AVG Free Edition.
>> Version: 7.5.446 / Virus Database: 268.18.15/728 - Release Date:
>> 03/20/2007 08:07 AM
>>
>
>
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