[asterisk-users] no progress indication

[Paul Belanger]

On 11-02-18 03:59 PM, Cassius Smith 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> core show application Progress()

> 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.
> 
There is something going wrong here, netsock2 is not parsing the IP
address correctly.  Are you using realtime?  It would be good to see a
full debug[1] log of your call.

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

-- 
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org