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Summary:ASTERISK-03479: [stable only] CallerID broken for me in latest chan_sip.c
Reporter:tog (tog)Labels:
Date Opened:2005-02-10 20:08:32.000-0600Date Closed:2008-01-15 15:25:23.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
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Description:CallerID to my SIP clients broke between Asterisk 1.0.3 and 1.0.5. I have not tried this in CVS HEAD, I only run stable.

I updated, compiled and installed 1.0.5 today and my incoming callerid via incoming SIP calls became "asterisk" in the name field and UNAVAILABLE in the number field no matter what.

I overwrote my chan_sip.so with the one from around 1.0.3 and callerid worked properly again.

The incoming callerid information is coming from zaptel channels (a US PRI.) Only incoming ANI exists to provide callerid information, CNAME delivery is not enabled with the telco.
Comments:By: Mark Spencer (markster) 2005-02-11 00:59:19.000-0600

Is this an issue with CVS head?

By: tog (tog) 2005-02-11 02:54:58.000-0600

I do not have CVS head running anywhere, this bug report only applies to the latest CVS stable. I do not want to run CVS head in production. If you are only interested in bug reports pertaining to CVS head and not to CVS stable then I'm afraid I can't be of much help to you. Since I hadn't yet seen a similar report I thought it would be a good idea to report this problem in 1.0.5. I am assuming the callerid changes made in 1.0.4 and 1.0.5 caused a problem.

It is very easy for me to reproduce, it happens with any SIP client I use if I just compile and run 1.0.5.

By: Olle Johansson (oej) 2005-02-11 03:06:48.000-0600

Tog: We are VERY interested in bug reports for CVS stable, don't hesitate about it. A stable release needs to be bug free.

I think markster just wanted to check if it is something we already fixed in CVS head, but forgot to fix in stable. If the behaviour is the same in the two versions of the code, we have to hunt the bug from scratch. If CVS head behaves differently, we can start with comparing the code to see what we are doing differently.

By: tog (tog) 2005-02-11 06:19:48.000-0600

I just installed Asterisk CVS head and tested it. Callerid works as expected.

By: Mark Spencer (markster) 2005-02-11 08:21:17.000-0600

Can you attach a relevent SIP trace of the problem occuring in CVS stable, (i.e. the asterisk console output with -vvvgc and sip debug enabled)

By: tog (tog) 2005-02-11 08:34:43.000-0600

It would be great if somebody could install CVS stable real quick and see if they can reproduce this same problem in their own setup. If the bug appears to have something to do with my specific configuration, I will spend the time to provide the detail necessary to track this down. I'm hoping this one is quick and easy for everybody to reproduce so it's easy to find, though.

Given my lack of time for the next couple of days, please excuse me if it takes me a while (even a couple days) to respond with the necessary details if it turns out not to be as easy to reproduce as I think it is.

By: Philipp Kolmann (pkolmann) 2005-02-11 08:52:23.000-0600

I have the same. Found the reasone. Its bugfix 3519. I have just filed another bug which is a duplicate (# 3563). (should have checked before. Sorry)

By: tog (tog) 2005-02-11 08:55:34.000-0600

Here is the console output edited for privacy purposes (the SIP client is my home phone and my cell is placing the call)

The console output doesn't seem any different from before when it worked.

   -- Accepting call from '305923XXXX' to '305453XXXX' on channel 0/1, span 1
   -- Executing Goto("Zap/1-1", "1305453XXXX|1") in new stack
   -- Goto (inbound,1305453XXXX,1)
   -- Executing Goto("Zap/1-1", "305453XXXX|s|1") in new stack
   -- Goto (305453XXXX,s,1)
   -- Executing Dial("Zap/1-1", "SIP/tog/tog") in new stack
   -- Called tog/tog
   -- SIP/tog-2811 is ringing
   -- Channel 0/1, span 1 got hangup
 == Spawn extension (305453XXXX, s, 1) exited non-zero on 'Zap/1-1'
   -- Hungup 'Zap/1-1'

By: tog (tog) 2005-02-11 08:56:42.000-0600

...And I was just about to do the sip debug output but pkolmann just took care of it. Thanks!

By: Tony Mountifield (softins) 2005-02-14 05:56:55.000-0600

I've just found the same problem. It is due to a one-char typo in 1.510.2.49. (CVS HEAD does not have the typo).

It's not worth uploading a patch file; the fix is just this:

Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.510.2.49
diff -u -r1.510.2.49 chan_sip.c
--- chan_sip.c  7 Feb 2005 17:10:25 -0000       1.510.2.49
+++ chan_sip.c  14 Feb 2005 11:50:22 -0000
@@ -3670,7 +3670,7 @@
               ast_callerid_parse(cid, &n, &l);
               if (l)
                       ast_shrink_phone_number(l);
-               if (!l || (ast_isphonenumber(l) && default_callerid[0]))
+               if (!l || (!ast_isphonenumber(l) && default_callerid[0]))
                               l = default_callerid;
       }
       /* if user want's his callerid restricted */

By: paradise (paradise) 2005-02-14 08:45:01.000-0600

i had the same problem with stable
this change, fixed it!
thanks

By: Mark Spencer (markster) 2005-02-14 09:29:01.000-0600

Marking as resolved for Russell to look at!  Thanks!

By: Russell Bryant (russell) 2005-02-14 16:50:42.000-0600

The typo has now been fixed.

I committed the change that introduced the typo on my PDA.  But don't worry, I have managed to break it within a month and a half, so it shouldn't happen again.  :)

By: Digium Subversion (svnbot) 2008-01-15 15:25:23.000-0600

Repository: asterisk
Revision: 5030

U   branches/v1-0/channels/chan_sip.c

------------------------------------------------------------------------
r5030 | russell | 2008-01-15 15:25:23 -0600 (Tue, 15 Jan 2008) | 2 lines

typo (bug ASTERISK-3479)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=5030