[asterisk-users] no progress indication

[Cassius Smith]

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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