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Summary:ASTERISK-17014: [patch] Missing P-Asserted-Identity
Reporter:George Konopacki (georgekonopacki)Labels:
Date Opened:2010-11-24 07:19:46.000-0600Date Closed:2011-05-06 11:23:16
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/CallCompletionSupplementaryServices
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) cc_configs.txt
( 1) issue18367_v1.8.patch
Description:Phone A is monitoring Phone B.

Phone B becomes available so the Asterisk server sends a NOTIFY(cc-ready) to Phone A.

Phone A calls Phone B (using the URI provided by the NOTIFY(cc-ready)).

Phone B receives P-Asserted-Identity in its INVITE message – GOOD

Phone A does NOT receive P-Asserted-Identity in any of its messages – BAD

This means the Phone A is displaying the 32 digit URI. Phone B displays the information provided by the P-Asserted-Identity.


SIP.CONF

sendrpid = yes
sendrpid = pai


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

Could this be a config issue?
Comments:By: Leif Madsen (lmadsen) 2010-12-07 10:42:55.000-0600

I'm not sure if this is a configuration issue right now.

I think it might be useful for you to attach a SIP trace as a text file, along with the console output with debug level logging and the appropriate parts of your dialplan and sip.conf files.
By: Mark Michelson (mmichelson) 2011-04-14 16:58:12

I've uploaded some configs that I tested that allowed for correct connected line information to be displayed.

To explain a bit, I initially did not have those CONNECTEDLINE(any) function calls in the dialplan. When I ran my test, correct P-Asserted-Identity information only came in the 200 OK. Then I added the CONNECTEDLINE(any) function call prior to the Dial, and correct P-Asserted-Identity information arrived in the 180 Ringing and the 200 OK.

I'm not 100% sure why info did not arrive in the 180 prior to the addition of the CONNECTEDLINE(any) functions, because I think that simply setting the callerid for the SIP peers and setting sendrpid=pai should have been enough. But what I have here worked for me, so give it a try.
By: George Konopacki (georgekonopacki) 2011-04-15 09:21:42

Hello Mark,

I have made your changes to our sip.conf and extensions.conf file. I can confirm it works.

I have heard from Jon that testing went well, except for a multiline issue. It has been very beneficial testing with you guys at SIPIT. I am happy you like how we have implemented the UI for call completion.

Regards,

George
By: Richard Mudgett (rmudgett) 2011-04-26 17:33:44

The issue18367_v1.8.patch file allows the connected line information to be updated as mmichelson mentioned in 18367#133790 without requiring the CONNECTEDLINE lines in the dialplan.
By: Digium Subversion (svnbot) 2011-05-06 11:19:20

Repository: asterisk
Revision: 317670

U   branches/1.8/channels/chan_sip.c

------------------------------------------------------------------------
r317670 | rmudgett | 2011-05-06 11:19:19 -0500 (Fri, 06 May 2011) | 22 lines

Fix SIP connected line updates.

This patch fixes a couple SIP connected line update problems:

1) The connected line needs to be updated when the initial INVITE is sent
if there is a peer callerid configured.  Previously, the connected line
information did not get reported until the call was connected so SIP could
not report connected line information in ringing or progress messages.

2) The connected line should not be updated on initial connect if there is
no connected line information.  Previously, all it did was wipe out any
default preset CONNECTEDLINE information set by the dialplan with empty
strings.

(closes issue ASTERISK-17014)
Reported by: GeorgeKonopacki
Patches:
     issue18367_v1.8.patch uploaded by rmudgett (license 664)
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1199/

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

http://svn.digium.com/view/asterisk?view=rev&revision=317670
By: Digium Subversion (svnbot) 2011-05-06 11:23:15

Repository: asterisk
Revision: 317671

_U  trunk/
U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r317671 | rmudgett | 2011-05-06 11:23:15 -0500 (Fri, 06 May 2011) | 29 lines

Merged revisions 317670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
 r317670 | rmudgett | 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) | 22 lines
 
 Fix SIP connected line updates.
 
 This patch fixes a couple SIP connected line update problems:
 
 1) The connected line needs to be updated when the initial INVITE is sent
 if there is a peer callerid configured.  Previously, the connected line
 information did not get reported until the call was connected so SIP could
 not report connected line information in ringing or progress messages.
 
 2) The connected line should not be updated on initial connect if there is
 no connected line information.  Previously, all it did was wipe out any
 default preset CONNECTEDLINE information set by the dialplan with empty
 strings.
 
 (closes issue ASTERISK-17014)
 Reported by: GeorgeKonopacki
 Patches:
       issue18367_v1.8.patch uploaded by rmudgett (license 664)
 Tested by: rmudgett
 
 Review: https://reviewboard.asterisk.org/r/1199/
........

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

http://svn.digium.com/view/asterisk?view=rev&revision=317671