[asterisk-bugs] [Asterisk 0014652]: [patch] early-dial SIP 484 "incomplete address"

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=14652 
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Reported By:                vieri
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14652
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Target Version:             1.4.28
Asterisk Version:           1.4.23 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-03-12 04:40 CDT
Last Modified:              2009-07-21 06:35 CDT
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Summary:                    [patch] early-dial SIP 484 "incomplete address"
Description: 
Calls in which at least one Grandstream GXP2000 or GXP280 is involved, get
dropped after about 20 seconds. This happens only if the Grandstreams are
configured to use "early dial" and Asterisk has pedantic=yes (with
pedantic=no the calls are not dropped).

I really need pedantic=yes to encode the # digits.

Would it be possible to use pedantic=no but make the # digits "work"?
(because I realize this may be a Grandstream firmware bug which I don't
think will be fixed any time soon)


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 (0108012) vieri (reporter) - 2009-07-21 06:35
 https://issues.asterisk.org/view.php?id=14652#c108012 
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If the full.earlydial log I attached does not help, I'm afraid I have no
idea why calls are dropped with pedantic=yes.

I see messages such as:

[Mar 12 09:19:28] WARNING[6501] chan_sip.c: Maximum retries exceeded on
transmission e09a6863f663a2aa at 10.215.146.162 for seqno 55056 (Critical
Response) -- See doc/sip-retransmit.txt.
[Mar 12 09:19:28] WARNING[6501] chan_sip.c: Hanging up call
e09a6863f663a2aa at 10.215.146.162 - no reply to our critical packet (see
doc/sip-retransmit.txt).

and I've also reviewed the chan_sip implementation but I'm not an expert
and just proposed a simple patch which "works for me". I'm wondering if
other Grandstream "early dial" users have experienced the same issue with
Asterisk.
Of course, I too would prefer to fix the root problem instead of adding
yet another sip option. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-07-21 06:35 vieri          Note Added: 0108012                          
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