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Summary:ASTERISK-07798: [patch] chan_sip unable to terminate call from servers using multipart/mixed Content-Types
Reporter:Dinesh Nair (alphaque)Labels:
Date Opened:2006-09-22 01:10:49Date Closed:2006-11-10 12:58:49.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) chan_sip.c.patch-1.2
( 1) chan_sip.c.patch-svn
Description:When a call is made to Asterisk from a server with multipart/mixed content types, the parsing of the SDP portion to pull out the application/sdp block ends prematurely in a loop in find_sdp(), leading to chan_sip being unable to get the SDP parameters from the block.

****** ADDITIONAL INFORMATION ******

This bug exists in 1.2.12.1 (and possible all previous versions since bug ASTERISK-6939) as well as Asterisk trunk (1.4.x). Use chan_sip.c.patch-svn for trunk and chan_sip.c.patch-1.2 for 1.2.x.
Comments:By: Dinesh Nair (alphaque) 2006-09-22 01:28:50

I dont have delete access to delete the wrongly named han_sip.c.patch-1.2. would someone delete it as it's redundant given that i've uploaded a file with the correct name ?

By: Serge Vecher (serge-v) 2006-09-22 10:47:09

switching this to 1.2.12.1, as it needs to be fixed the 1.2 branch, then 1.4, then trunk.

By: Dinesh Nair (alphaque) 2006-10-04 08:09:32

Can this be fixed in both 1.2.12.1 and trunk, as there's been no objections as of yet ?

By: jmls (jmls) 2006-11-02 12:10:22.000-0600

ping. housekeeping

By: Serge Vecher (serge-v) 2006-11-09 08:17:11.000-0600

alphaque: let's see if we can show that your patch actually works. Please enable sip debug on 1.2.13 plain and with patch applied and attach both outputs here. Thanks.

By: Olle Johansson (oej) 2006-11-10 12:51:14.000-0600

Committed to 1.2, 1.4 and trunk. Thanks!

By: Olle Johansson (oej) 2006-11-10 12:58:48.000-0600

I modified your patch a tiny little bit. Please make sure it works :-) Thank you for your contribution to Asterisk. /Olle