[asterisk-users] no progress indication

[Satish Patel]

Try to use Answer() in your dial plan. Not sure though but it had been  
resoved my issue years ago.

--
Sent from my iPhone

On Feb 18, 2011, at 3:59 PM, Cassius Smith <cassius at cassius.org> wrote:

> I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with  
> VOIP
> only trunks, and this server only has soft phones.
> When I dial an extension and the phone is not registered, I don't  
> get any
> ring or progress indications, and eventually the Dial() times out and
> drops into voicemail (as expected).
>
> CLI output:
>    -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",
> "SIP/RickEndpoint&SIP/xlite-RickEndpoint,20") in new stack
>  == Using SIP RTP CoS mark 5
> [Feb 18 20:43:04] WARNING[6160]: acl.c:698 ast_ouraddrfor: Cannot  
> connect
> [Feb 18 20:43:04] WARNING[6160]: chan_sip.c:3115 __sip_xmit:  
> sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
> argument
>    -- Called RickEndpoint
> [Feb 18 20:43:04] WARNING[6160]: app_dial.c:2039 dial_exec_full:  
> Unable to
> create channel of type 'SIP' (cause 20 - Unknown)
> [Feb 18 20:43:04] WARNING[3550]: chan_sip.c:3115 __sip_xmit:  
> sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
> argument
> [Feb 18 20:43:05] WARNING[3550]: chan_sip.c:3115 __sip_xmit:  
> sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
> argument
> [Feb 18 20:43:07] WARNING[3550]: chan_sip.c:3115 __sip_xmit:  
> sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
> argument
>  == Spawn extension (macro-StdExten, s, 6) exited non-zero on
> 'IAX2/barneveld-2036' in macro 'StdExten'
>  == Spawn extension (no911, RickEndpoint, 1) exited non-zero on
> 'IAX2/barneveld-2036'
>    -- Hungup 'IAX2/barneveld-2036'
> [Feb 18 20:43:11] WARNING[3550]: chan_sip.c:3115 __sip_xmit:  
> sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
> argument
> [Feb 18 20:43:19] WARNING[3550]: chan_sip.c:3115 __sip_xmit:  
> sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
> argument
> [Feb 18 20:43:35] WARNING[3550]: chan_sip.c:3115 __sip_xmit:  
> sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
> argument
> [Feb 18 20:43:36] WARNING[3550]: chan_sip.c:3386 retrans_pkt:
> Retransmission timeout reached on transmission
> 367fd44f3a944b134765a4dc4c95b88d at 127.0.0.1:5060 for seqno 102  
> (Critical
> Request) -- See doc/sip-retransmit.txt.
>
>
>
> Here is my StdExten macro:
>
> [macro-StdExten]
> exten => s,1,Verbose(2,>>>>>>>>>>>>>>>Processing StdExten call for
> ${MACRO_EXTEN}<<<<<<<<<<<<<<<<)
> exten => s,n,Verbose(2,CallerID => ${CALLERID(all)})
> exten => s,n,Set(Device=${ARG1})
> exten => s,n,Set(UserID=${MACRO_EXTEN})
> exten => s,n,Dial(${ARG1},${ARG2})
> exten => s,n,Verbose(2,==> Voicemail ${MACRO_EXTEN} -- unavail)
> exten => s,n,Voicemail(${MACRO_EXTEN}@default,u)
> exten => s,n,Hangup()
>
>
> I was expecting the system to indicate that ringing was ?
> I know I can check channel availability to bypass this behavior; just
> curious why it's happening or whether it's expected.
>
> Cassius
>
> -- 
>
>
>
>
>
> --
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